Internet Engineering Task Force SIP WG
Internet Draft Jonathan Rosenberg
dynamicsoft
Henning Schulzrinne
Columbia U.
Gonzalo Camarillo
Ericsson
Alan Johnston
Worldcom
Jon Peterson
Neustar
Robert Sparks
dynamicsoft
Mark Handley
ACIRI
Eve Schooler
AT&T
draft-ietf-sip-rfc2543bis-05.txt
October 26, 2001
Expires: April 2002
SIP: Session Initiation Protocol
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
The Session Initiation Protocol (SIP) is an application-layer control
(signaling) protocol for creating, modifying and terminating sessions
with one or more participants. These sessions include Internet
telephone calls, multimedia distribution and multimedia conferences.
SIP invitations used to create sessions carry session descriptions
which allow participants to agree on a set of compatible media types.
SIP makes use of elements called proxy servers to help route requests
to the users current location, assist in firewall traversal, and
provide features to users. SIP also provides a registration function
that allows them to upload their current location for use by proxy
servers. SIP runs ontop of several different transport protocols.
1 Introduction
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There are many applications of the Internet that require the creation
and management of a session, where a session is considered an
exchange of data between an association of participants. The
implementation of these services is complicated by the practices of
participants; users may move between endpoints, they may be
addressable by multiple names, and they may communicate in several
different media - sometimes simultaneously. Numerous protocols have
been authored that carry various forms of real-time multimedia
session data such as voice, video, or text messages. SIP works in
concert with these protocols by enabling Internet endpoints (called
"user agents") to discover one another and to agree on a
characterization of a session they would like to share. For locating
prospective session participants, SIP relies on an infrastructure of
network hosts (called "proxy servers") to which user agents can send
registrations, invitations to sessions and other requests. SIP is an
agile, general-purpose tool for creating, modifying and terminating
sessions that works independently of underlying transport protocols
and without dependency on the type of session that is being
established.
2 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer control
protocol that can establish, modify and terminate multimedia sessions
(conferences) such as Internet telephony calls. SIP can also invite
participants to already existing sessions. A SIP entity issuing an
invitation for an already existing session does not necessarily have
to be a member of the session to which it is inviting. Media can be
added to (and removed from) an existing session. SIP transparently
supports name mapping and redirection services, which supports
personal mobility [1] - users can maintain a single externally
visible identifier (SIP URI) regardless of their network location.
SIP supports five facets of establishing and terminating multimedia
communications:
User location: determination of the end system to be used for
communication;
User availability: determination of the willingness of the
called party to engage in communications;
User capabilities: determination of the media and media
parameters to be used;
Session setup: "ringing", establishment of session parameters at
both called and calling party;
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Session handling: including transfer and termination of
sessions, modifying session parameters, and invoking
services.
SIP is not a vertically integrated communications system. SIP is
rather a component of the overall IETF multimedia data and control
architecture which incorporates protocols such as RSVP (RFC 2205 [2])
for reserving network resources, the real-time transport protocol
(RTP) (RFC 1889 [3]) for transporting real-time data and providing
QOS feedback, the real-time streaming protocol (RTSP) (RFC 2326 [4])
for controlling delivery of streaming media, the session announcement
protocol (SAP) [5] for advertising multimedia sessions via multicast
and the session description protocol (SDP) (RFC 2327 [6]) for
describing multimedia sessions. Therefore, SIP should be used in
conjunction with other protocols in order to provide complete
services to the users. However, the basic functionality and operation
of SIP does not depend on any of these protocols.
SIP does not provide services. SIP rather provides primitives that
can be used to implement different services. For example, SIP can
locate a user and deliver an opaque object to his current location.
If this primitive is used to deliver a session description written in
SDP, for instance, the parameters of a session can be agreed between
endpoints. If the same primitive is used to deliver a photo of the
caller as well as the session description, a "caller ID" service can
be easily implemented. As this example shows, a single primitive is
typically used to provide several different services. Consequently,
generality is more important than efficiency when designing SIP
primitives.
SIP does not offer conference control services such as floor control
or voting and does not prescribe how a conference is to be managed,
but SIP can be used to initiate a session that uses some other
conference control protocol. SIP does not allocate multicast
addresses and does not reserve network resources.
3 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALLNOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [7] and
indicate requirement levels for compliant SIP implementations.
4 Overview of Operation
This section will introduce the basic operations of the SIP protocol
using simple examples. Note that this section is tutorial in nature
and does not contain any normative statements.
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The first example will show the basic functions of SIP: location of
an end point, signaling a desire to communicate, negotiation of
session parameters to establish the session, and teardown of the
session once established.
Figure 1 shows a typical example of a SIP message exchange between
two users, Alice and Bob. (Each message is labeled with the letter
"F" and a number for reference by the text.) In this example, Alice
uses a SIP application on her PC (referred to as a softphone) to call
Bob on his SIP phone over the Internet. Also shown are two SIP proxy
servers which act on behalf of Alice and Bob to facilitate the
session establishment. This typical arrangement is often referred to
as the "SIP trapezoid" as shown by the geometric shape of the dashed
lines in Figure 1.
Alice "calls" Bob using his SIP identity, a type of Uniform Resource
Identifier (URI) called a SIP URI and defined in Section 21.1. It has
a similar form to an email address, typically containing a username
and a host name. In this case it is sip:bob@biloxi.com, where
biloxi.com is the domain of Bob's SIP service provider (which can be
an enterprise, retail provider, etc). Alice also has a SIP URI of
sip:alice@atlanta.com. Alice might have typed in Bob's URI or perhaps
clicked on a hyperlink or an entry in an address book.
SIP is based on an HTTP-like request/response transacton model. Each
transaction consists of a request that invokes a particular "Method",
or function, on the server, and at least one response. In this
example, the transaction begins with Alice's softphone sending an
INVITE request addressed to Bob's SIP URI. INVITE is an example of a
SIP method which specifies the action that the requestor (Alice)
wants the server (Bob) to take. The INVITE request contains a number
of header fields. Header fields are additional named attributes which
provide additional information about a message. The ones present in
an INVITE include a unique identifier for the call, the destination
address, Alice's address, and information about the type of session
that Alice wishes to establish with Bob. The INVITE (message F1 in
Figure 1) might look like this:
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:alice@10.1.3.3>
Content-Type: application/sdp
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atlanta.com . . . biloxi.com
. proxy proxy .
. .
Alice's . . . . . . . . . . . . . . . . . . . . Bob's
softphone SIP Phone
| | | |
| INVITE F1 | | |
|--------------->| INVITE F3 | |
| 100 Trying F2 |--------------->| INVITE F5 |
|<---------------| 100 Trying F4 |--------------->|
| |<-------------- | 180 Ringing F6 |
| | 180 Ringing F7 |<---------------|
| 180 Ringing F8 |<---------------| 200 OK F9 |
|<---------------| 200 OK F10 |<---------------|
| 200 OK F11 |<---------------| |
|<---------------| | |
| ACK F12 |
|------------------------------------------------->|
| Media Session |
|<================================================>|
| BYE F13 |
|<-------------------------------------------------|
| 200 OK F14 |
|------------------------------------------------->|
| |
Figure 1: SIP session setup example with SIP trapezoid
Contact-Length: 142
(Alice's SDP not shown)
The first line of the text-encoded message contains the method name
(INVITE). The lines which follow are a list of header fields. This
example contains a minimum required set. The headers are briefly
described below:
Via contains the IP address (10.1.3.3), port number (5060), and
transport protocol (UDP) on which Alice is expecting to receive
responses to this request.
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To contains a display name (Bob) and a SIP URI (sip:bob@biloxi.com)
that the request was originally directed towards.
From also contains a display name (Alice) and a SIP URI
(sip:alice@atlanta.com) that indicate the originator of the request.
This header field also has a tag parameter which contains a
pseudorandom string (1928301774) which was added to the URI by the
softphone. It is used for identification purposes.
Call-ID contains a globally unique identifier for this call,
generated by the combination of a pseudorandom string and the
softphone's IP address. The combination of the To, From, and Call-ID
completely define a peer-to-peer SIP relationship betwee Alice and
Bob, and is referred to as a "dialog".
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request, and is a traditional
sequence number.
Contact contains a SIP URI which represents a direct route to reach
or contact Alice, usually composed of a username at an IP address.
While the Via header field is used to tell other elements where to
send the response, the Contact header field tells other elements
where to send future requests for this dialog.
Content-Type contains a description of the message body (not shown).
Content-Length contains an octet (byte) count of the message body.
The complete set of SIP header fields is defined in Section 22.
The details of the session, type of media, codec, sampling rate, etc.
are not described using SIP. Rather, the body of a SIP message
contains a description of the session, encoded in some other protocol
format. One such format is Session Description Protocol (SDP) [6].
This SDP message (not shown in the example) is carried by the SIP
message in an analogous way that a document attachment is carried by
an email message, or a web page is carried in an HTTP message.
Since the softphone has no knowledge of Bob's exact location, or how
to locate the SIP server in the biloxi.com domain, the softphone
sends the INVITE to the SIP server that serves Alice's domain,
atlanta.com. The IP address of the atlanta.com SIP server could have
been configured in Alice's softphone, or it could have been
discovered by DHCP, for example.
The atlanta.com SIP server is a type of SIP server known as a proxy
server. A proxy server receives SIP requests and forwards them on
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behalf of the requestor. In this example, the proxy server receives
the INVITE request and generates a 100 Trying response which is sent
back to Alice's softphone. The 100 Trying response indicates that the
INVITE has been received and that the proxy is working on her behalf
to try to route the INVITE to the destination. Responses in SIP use
a numerical three digit code followed by a descriptive phrase. This
response contains the same To, From, Call-ID, and CSeq as the INVITE,
which allows Alice's softphone to correlate this response to the sent
INVITE. The atlanta.com proxy server locates the proxy server at
biloxi.com, possibly by performing a DNS (Domain Name Service) lookup
to find the SIP server which serves the biloxi.com domain. This is
described in Section 24. As a result, it obtains the IP address of
the biloxi.com proxy server and forwards, or proxies, the INVITE
request there. Before forwarding the request, the atlanta.com proxy
server adds an additional Via header field which contains its own IP
address (the INVITE already contains Alice's IP address in the first
Via). The biloxi.com proxy server receives the INVITE and responds
with a 100 Trying response back to the Atlanta.com proxy server to
indicate that it has received the INVITE and is processing the
request. The proxy server consults a database, generically called a
location service, which contains the current IP address of Bob. (We
shall see in the next section how this database can be populated.)
The biloxi.com proxy server adds another Via header with its own IP
address to the INVITE and proxies it to Bob's SIP phone.
Bob's SIP phone receives the INVITE and begins to alert Bob to the
incoming call from Alice so that Bob can decide whether or not to
answer the call - i.e. Bob's phone rings. Bob's SIP phone sends an
indication of this in a 180 Ringing response. This response is routed
back thorough the two proxies in the reverse direction. Each proxy
uses the Via header to figure out where to send the response, and
removes its own address from the top. As a result, although DNS and
location service lookups were required to route the initial INVITE,
the 180 Ringing response can be returned to the caller without
lookups, or without state being maintained in the proxies. This also
has the desirable property that each proxy that sees the INVITE will
also see all responses to the INVITE.
When Alice's softphone receives the 180 Ringing response, it passes
this information to Alice, perhaps using an audio ringback tone, or
just by displaying or flashing a message on Alice's screen.
In this example, Bob decides to answer the call. When he picks up the
handset his SIP phone sends a 200 OK response to indicate that the
call has been answered. The 200 OK contains a message body containing
the SDP media description of the type of session that Bob is willing
to establish with Alice. As a result, there is a two-phase exchange
of SDP messages; Alice sent one to Bob, and Bob sent one back to
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Alice. This two-phase exchange provides basic negotiation
capabilities, and is based on a simple offer/answer model, If Bob did
not wish to answer the call, or was busy on another call, an error
response would have been sent instead of the 200 OK, which would have
resulted in no media session being established. The complete list of
SIP response codes is in Section 23. The 200 OK (message F9 in Figure
1) might look like this:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:bob@10.4.1.4>
Content-Type: application/sdp
Contact-Length: 131
(Bob's SDP not shown)
The first line of the response contains the response code (200) and
the reason phrase (OK). The remaining lines contain header fields.
The Via header fields, To, From, Call- ID, and CSeq are all copied
from the INVITE request. (Note that there are three Via headers -
one added by Alice's SIP phone, one added by the atlanta.com proxy,
and one added by the biloxi.com proxy.) Also note that Bob's SIP
phone has added a tag parameter to the To header field. This tag will
be incorporated by both User Agents into the dialog and will be
included in all future requests and responses in this call. The
Contact header field contains a URI at which Bob can be directly
reached at his SIP phone. The Content-Type and Content-Length refer
to the not shown message body which contains Bob's SDP media
information.
In additon to DNS and location service lookups shown in this example,
proxy servers can make arbitrarily complex "routing decisions" in
order to decide where to send a request. For example, if Bob's SIP
phone returned a 486 Busy Here response, the biloxi.com proxy server
could proxy the INVITE to Bob's voicemail server. A proxy server can
also send an INVITE to a number of locations at the same time. This
type of parallel search is known as "forking".
In this case, the 200 OK is routed back through the two proxies and
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is received by Alice's softphone which then stops the ringback tone
and indicates that the call has been answered. Finally, an
acknowledgement message, ACK, is sent by Alice to Bob to confirm the
reception of the final response (200 OK). Note that in this example,
the ACK is sent directly from Alice to Bob, bypassing the two
proxies. This is due to the fact that through the INVITE/200 OK
exchange, the two SIP user agents have learned each other's IP
address through the Contact header fields, something which was not
known when the initial INVITE was sent. The lookups performed by the
two proxies are no longer needed, so they drop put of the call flow.
This completes the INVITE/200/ACK three way handshake used to
establish SIP sessions, and is the end of the transaction. Full
details on session setup is in Section 13.
Alice and Bob's media session has now begun, and they begin sending
media packets using the format agreed to in the exchange of SDP. In
general, the end-to-end media packets will take a different path from
the SIP signaling messages.
During the session, either Alice or Bob may decide to change the
characteristics of the media session. This is accomplished by sending
a re-INVITE containing a new media description. If the change is
acceptable to the other party, a 200 OK is sent which is itself
responded to with an ACK. This re-INVITE will reference the existing
dialog so the other party knows that it is to modify an existing
session instead of establishing a new session. If the change is not
acceptable, an error response, such as a 406 Not Acceptable response
is sent, which also receives an ACK. However, the failure of the re-
INVITE does not cause the existing call to fail - the session
continues using the previously negotiated characteristics. Full
details on session modification is in Section 14.
At the end of the call, Bob disconnects (hangs up) first, and
generates a BYE message. This BYE is routed directly to Alice's
softphone, again bypassing the proxies. Alice confirms receipt of the
BYE with a 200 OK response, which terminates the session and the BYE
transaction. Note that no ACK is sent - an ACK is only sent in
response to a response to an INVITE request. The reasons for this
special handling for INVITE will be discussed later, but relate to
the reliability mechanisms in SIP, the length of time it can take for
a ringing phone to be answered, and forking. For this reason, request
handling in SIP is often classified as either INVITE or non- INVITE,
referring to all other methods besides INVITE. Full details on
session termination is in Section 15.
Full details of all the messages shown in the example of Figure 1 are
shown in Section 25.2.
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In some cases, it may be useful for proxies in the SIP signaling path
see all the messaging between the two endpoints for the duration of
the session. For example, if the biloxi.com proxy server wished to
remain in the SIP messaging path beyond the initial INVITE, it would
add to the INVITE a required routing header field known as Record-
Route containing a URI which resolves to the proxy. This information
would be received by both Bob's SIP phone and (due to the Record-
Route header field being passed back in the 200 OK) Alice's softphone
and stored for the duration of the dialog. This would then result in
the ACK, BYE, and 200 OK to the BYE being received and proxied by the
biloxi.com proxy server. Each proxy can independently decide to
receive subsequent messaging, and that messaging will go through all
proxies that elected to receive it. A common use of this capability
is in firewall traversal or mid-call feature implementation.
Registration is another common operation in SIP. Registration is one
way in which the biloxi.com server can learn the current location of
Bob. Upon initialization, and at periodic intervals, Bob's SIP phone
sends REGISTER messages a server in the biloxi.com domain known as a
SIP registrar. The REGISTER messages associate Bob's SIP URL
(sip:bob@biloxi.com) with the machine he is currently logged in at
(conveyed as a SIP URL in the Contact header). The registrar writes
this association, also called a binding, to a database, called the
location service , where it can be used by the proxy in the
biloxi.com domain. Often, a registrar server for a domain is co-
located with the proxy for that domain. It is an important concept
that the distinction between types of SIP servers are logical, not
physical.
Bob is not limited to registering from a single device. For example,
both his SIP phone at home and the one in the office could send in
registrations. This information is stored together in the location
service, and allows a proxy to perform various types of searches to
locate Bob. Similarly, more than one user can be registered on a
single device at the same time.
The location service is just an abstract concept. It generally
contains information that allows a proxy to input a URI and get back
a translated URI that tells the proxy where to send the request.
Registrations are one way to create this information, but not the
only way. Arbitrarily complex mapping functions can be programmed, at
the discretion of the administrator.
Finally, it is important to note that in SIP, registration is used
for routing incoming SIP requests and has no role in authorizing
outgoing requests. Authorization and authentication are handled in
SIP either on a request by request, challenge/response mechanism, or
using a lower layer scheme as discussed in Section 20.
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The complete set of SIP message details for this registration example
is in Section 25.2.
Additional operations in SIP include querying for the capabilities of
a SIP server or client using OPTIONS, and canceling a pending request
using CANCEL will be introduced in later sections.
5 Structure of the Protocol
The SIP protocol is structured as a layered protocol, which means
that its behavior is described in terms of a set of fairly
independent processing stages, with only a loose coupling between
each stage. The structuring of the protocols into layers is for the
purpose of presentation and conciseness; it allows the grouping of
functions common across elements into a single place. It does not
dictate an implementation in any way. When we say that an element
"contains" a layer, that means it is compliant to the set of rules
defined by that layer.
Not every element specified by the protocol contains every layer.
Furthermore, the elements specified by SIP are logical elements, not
physical ones. A physical realization can choose to act as different
logical elements, perhaps even on a transaction by transaction basis.
The lowest layer of the SIP protocol is its syntax and encoding. Its
encoding is specified using a BNF. The complete BNF is specified in
Section 26. However, a basic overview of the structure of a SIP
message can be found in Section 7. This section introduces enough of
an understanding of the format of a SIP message to facilitate
understanding the remainder of the protocol.
The next higher layer is the transport layer. This layer defines how
a client takes a request, and physically sends it over the network,
and how a response is sent by a server, and then received by a
client. All SIP elements contain a transport layer. The transport
layer is described in Section 19.
The next higher layer is the transaction layer. Transactions are a
fundamental component of SIP. A transaction is a request, sent by a
client transaction (using the transport layer), to a server
transaction, along with all responses to that request sent from the
server transaction back to the client. The transaction layer handles
retransmissions, matching of responses to requests, and timeouts. Any
task that a UAC wishes to accomplish takes place using a series of
transactions. Discussion of transactions can be found in Section 17.
User agents contain a transaction layer, as do stateful proxies.
Stateless proxies do not contain a transaction layer.
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The transaction layer has a client component (referred to as a client
transaction), and a server component (referred to as a server
transaction), each of which are represented by an FSM that is
constructed to process a particular request. The layer on top of the
transaction layer is called the transaction user (TU), of which there
are several types. When a TU wishes to send a request, it creates a
client transaction instance and passes it the request, along with the
destination IP address, port, and transport to send the request to.
SIP provides the ability for a transaction to be canceled by the
client which initiated it. When a client cancels a transaction, it
requests that the server give up on further processing, revert to the
state that existed before the transaction was initiated, and generate
a specific error response to that transaction. This is done with a
CANCEL request, which constitutes its own transaction, but references
the transaction to be cancelled. Cancellation is described in Section
9.
There are several different types of transaction users. A UAC
contains a UAC core, a UAS contains a UAS core, and a proxy contains
a proxy core. The behavior of the UAC and UAS cores depend largely on
the method. However, there are some common rules for all methods.
These rules are captured in Section 8. The primarily deal with
construction of a request, in the case of a UAC, and processing of
that request, and generation of a response, in the case of a UAS.
UAC and UAS core behavior for the REGISTER method is described in
Section 10. Registrations play an important role in SIP. In fact, a
UAS that handles a REGISTER is given a special name - a registrar -
and it is described in that section.
UAC and UAS core behavior for the OPTIONS method, used for
determining the capabilities of a UAC, are described in Section 11.
Certain other requests are sent within a dialog peer-to-peer SIP
relationship between a two user agents that persists for some time.
The dialog facilitates sequencing of messages between the user
agents, and proper routing of requests between both them. One way to
setup a dialog is with the INVITE method. When a UAC sends a request
that is within the context of a dialog, it follows the common UAC
rules as discussed in Section 8, but also the rules for mid-dialog
requests. Section 12 discusses dialogs, and presents the procedures
for their construction, and maintenance, in addition to construction
of requests within a dialog.
The most important method in SIP is the INVITE method, which is used
to establish a session between participants. A session is a
collection of participants, and streams of media between them, for
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the purposes of communication. Section 13 discusses how sessions are
initiated, resulting in one or more SIP dialogs. Section 14 discusses
how characteristics of that session are modified, through the use of
an INVITE request within a dialog. Finally, section 15 discusses how
a session is terminated.
The procedures of Sections 8, 10, 11, 12, 13, 14, and 15 deal
entirely with the UA core. Section 16 discusses the proxy element,
which facilitates routing of messages between user agents.
6 Definitions
This specification uses a number of terms to refer to the roles
played by participants in SIP communications. The definitions of
client, server and proxy are similar to those used by the Hypertext
Transport Protocol (HTTP) (RFC 2616 [8]). The terms and generic
syntax of URI and URL are defined in RFC 2396 [9]. The following
terms have special significance for SIP.
Back-to-Back user agent: A back-to-back user agent (B2BUA) is a
logical entity that receives a request, and processes it as
a UAS. In order to determine how the request should be
answered, it acts as a UAC and generates requests. Unlike a
proxy server, it maintains dialog state, and must
participate in all requests sent on the dialogs it has
established. Since it is a concatenation of a UAC and UAS,
no explicit definitions are needed for its behavior.
Call: A call is an informal term that refers to a dialog between
peers, generally set up for the purposes of a multimedia
conversation.
Call leg: Another name for a dialog.
Call stateful: A proxy is call stateful if it retains state for
a dialog from the initiating INVITE to the terminating BYE
request. A call stateful proxy is always stateful, but the
converse is not true.
Client: A client is any network element that sends SIP requests,
and receives SIP responses. Clients may or may not interact
directly with a human user. User agent clients and proxies
are clients.
Conference: A multimedia session (see below) that contains
multiple participants.
Dialog: A dialog is a peer-to-peer SIP relationship between a
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UAC and UAS that persists for some time. A dialog is
established by SIP messages, such as a 2xx response to an
INVITE request. A dialog is identified by a call
identifier, local address, and remote address. A dialog was
formerly known as a call leg in RFC 2543.
Downstream: A direction of message forwarding within a
transaction which refers to the direction that requests
flow from the user agent client to user agent server.
Final response: A response that terminates a SIP transaction, as
opposed to a provisional response that does not. All 2xx,
3xx, 4xx, 5xx and 6xx responses are final.
Informational Response: Same as a provisional response.
Initiator, calling party, caller: The party initiating a session
with an INVITE request. A caller retains this role from the
time it sends the INVITE until the termination of any
dialogs established by the INVITE.
Invitation: An INVITE request.
Invitee, invited user, called party, callee: The party that
receives an INVITE request for the purposes of establishing
a new session. A callee retains this role from the time it
receives the INVITE until the termination of the dialog
established by that INVITE.
Isomorphic request or response: Two requests are defined to be
isomorphic for the purposes of this document if they have
the same values for the Call-ID, To, From, CSeq, Request-
URI and the top-most Via header. Two responses are
isomorphic if they have the same values for the Call-ID,
To, From, CSeq and top Via header. A message which is
isomorphic to another is also known as a retransmission.
Location server: See location service.
Location service: A location service is used by a SIP redirect
or proxy server to obtain information about a callee's
possible location(s). It is an abstract database, sometimes
referred to as a location server. The contents of the
database can be populated in many ways, including being
written by registrars.
Loop: A request that arrives at a proxy, is forwarded, and later
arrives back at the same proxy. When it arrives the second
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time, its Request-URI is identical to the first time, and
other headers that affect proxy operation are unchanged, so
that the proxy would make the same processing decision on
the request it made the first time around. Looped requests
are errors, and the procedures for detecting them and
handling them are described by the protocol.
Method: The method is the primary function that a request is
meant to invoke on a server. The method is carried in the
request message itself. Example methods are INVITE and BYE.
Outbound proxy: A proxy that receives all requests from a
client, even though it is not the server resolved by the
Request-URI. The outbound proxy sends these requests, after
any local processing, to the address indicated in the
Request-URI, or to another outbound proxy.
Parallel search: In a parallel search, a proxy issues several
requests to possible user locations upon receiving an
incoming request. Rather than issuing one request and then
waiting for the final response before issuing the next
request as in a sequential search , a parallel search
issues requests without waiting for the result of previous
requests.
Provisional response: A response used by the server to indicate
progress, but that does not terminate a SIP transaction.
1xx responses are provisional, other responses are
considered final.
Proxy, proxy server: An intermediary entity that acts as both a
server and a client for the purpose of making requests on
behalf of other clients. A proxy server primarily plays to
role of routing, which means its job is to ensure that a
request is passed on to another entity that can further
process the request. Proxies are also useful for enforcing
policy and for firewall traversal. A proxy interprets, and,
if necessary, rewrites parts of a request message before
forwarding it.
Registrar: A registrar is a server that accepts REGISTER
requests, and places the information it receives in those
requests into the location service for the domain it
handles.
Regular Transaction: A regular transaction is any transaction
with a method other than INVITE, ACK, or CANCEL.
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Ringback: Ringback is the signaling tone produced by the calling
party's application indicating that a called party is being
alerted (ringing).
Server: A server is a network element that receives requests in
order to service them, and sends back responses to those
requests. Examples of servers are proxies, user agent
servers, redirect servers, and registrars.
Sequential search: In a sequential search, a proxy server
attempts each contact address in sequence, proceeding to
the next one only after the previous has generated a non-
2xx final response.
Session: From the SDP specification: "A multimedia session is a
set of multimedia senders and receivers and the data
streams flowing from senders to receivers. A multimedia
conference is an example of a multimedia session." (RFC
2327 [6]) (A session as defined for SDP can comprise one or
more RTP sessions.) As defined, a callee can be invited
several times, by different calls, to the same session. If
SDP is used, a session is defined by the concatenation of
the user name , session id , network type , address type
and address elements in the origin field.
(SIP) transaction: A SIP transaction occurs between a client and
a server and comprises all messages from the first request
sent from the client to the server up to a final (non-1xx)
response sent from the server to the client, and the ACK
for the response in the case the response was a 2xx. The
ACK for a 2xx response is a separate transaction.
Spiral: A spiral is a SIP request which is routed to a proxy,
forwarded onwards, and arrives once again at that proxy,
but this time, differs in a way which will result in a
different processing decision than the original request.
Typically, this means that it has a Request-URI that
differs from the previous arrival. A spiral is not an error
condition, unlike a loop.
Stateless proxy: A logical entity that does not maintain the
client or server transaction state machines defined in this
specification when it processes requests. A stateless proxy
forwards every request it receives downstream and every
response it receives upstream.
Stateful proxy: A logical entity that maintains the client and
server transaction state machines defined by this
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specification during the processing of a request. Also
known as a transaction stateful proxy. The behavior of a
stateful proxy is further defined in Section 16. A stateful
proxy is not the same as a call stateful proxy.
Transaction User (TU): The layer of protocol processing that
resides above the transaction layer. Transaction users
include the UAC core, UAS core, and proxy core.
Upstream: A direction of message forwarding within a transaction
which refers to the direction that responses flow from the
user agent server to user agent client.
URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [10].
User agent client (UAC): A user agent client is a logical entity
that creates a new request, and then uses the client
transaction state machinery to send it. The role of UAC
lasts only for the duration of that transaction. In other
words, if a piece of software initiates a request, it acts
as a UAC for the duration of that transaction. If it
receives a request later on, it takes on the role of a User
Agent Server for the processing of that transaction.
UAC Core: The set of processing functions required of a UAC that
reside above the transaction and transport layers.
User agent server (UAS): A user agent server is a logical entity
that generates a response to a SIP request. The response
accepts, rejects or redirects the request. This role lasts
only for the duration of that transaction. In other words,
if a piece of software responds to a request, it acts as a
UAS for the duration of that transaction. If it generates a
request later on, it takes on the role of a User agent
client for the processing of that transaction.
UAS Core: The set of processing functions required at a UAS that
reside above the transaction and transport layers.
User agent (UA): A logical entity which can act as both a user
agent client and user agent server for the duration of a
dialog.
The role of UAC and UAS as well as proxy and redirect servers are
defined on a transaction-by-transaction basis. For example, the user
agent initiating a call acts as a UAC when sending the initial INVITE
request and as a UAS when receiving a BYE request from the callee.
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Similarly, the same software can act as a proxy server for one
request and as a redirect server for the next request.
Proxy, location and registrar servers defined above are logical
entities; implementations MAY combine them into a single application
program.
7 SIP Messages
SIP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [11]).
A SIP message is either a request from a client to a server, or a
response from a server to a client.
Both Request (section 7.1) and Response (section 7.2) messages use
the generic-message format of RFC 822 [12]. Both types of messages
consist of a start-line, one or more header fields (also known as
"headers"), an empty line indicating the end of the header fields,
and an optional message-body.
generic-message = start-line
*message-header
CRLF
[ message-body ]
The start-line, each message-header line, and the empty line MUST be
terminated by a carriage-return line-feed sequence (CRLF). Note that
the empty line MUST be present even if the message-body is not.
Except for the above difference in character sets, much of SIP's
message and header field syntax is identical to HTTP/1.1. Rather than
repeating the syntax and semantics here we use [HX.Y] to refer to
Section X.Y of the current HTTP/1.1 specification (RFC 2616 [8]).
Note, however, that SIP is not an extension of HTTP.
7.1 Requests
SIP Requests are distinguished by having a Request-Line for a start-
line. A Request-Line begins with a method token, followed by the
Request-URI and the protocol version, and ending with CRLF. The
elements are separated by SP characters. No CR or LF are allowed
except in the end-of-line CRLF sequence. No LWS is allowed in any of
the elements.
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Method Request-URI SIP-Version
o Method
This specification defines six methods : REGISTER for
registering contact information, INVITE, ACK and CANCEL for
setting up sessions, BYE for terminating sessions and OPTIONS
for querying servers about their capabilities. SIP extensions
may define additional methods.
o Request-URI
The Request-URI is a SIP URL as described in Section 21.1 or a
general URI (RFC 2396 [9]). It indicates the user or service
to which this request is being addressed. The Request-URI
MUST NOT contain unescaped spaces or control characters and
MUST NOT be enclosed in "<>".
SIP servers MAY support Request-URIs with schemes other than
"sip", for example the "tel" URI scheme of RFC 2806 [13]. It
MAY translate non-SIP URIs using any mechanism at its
disposal, resulting in either a SIP URI or some other scheme.
o SIP Version
Both request and response messages include the version of SIP
in use, and follow [H3.1] (with HTTP replaced by SIP, and
HTTP/1.1 replaced by SIP/2.0) regarding version ordering,
compliance requirements, and upgrading of version numbers. To
be compliant with this specification, applications sending SIP
messages MUST include a SIP- Version of "SIP/2.0". The string
is case-insensitive, but implementations MUST send upper-case.
Unlike HTTP/1.1, SIP treats the version number as a
literal string. In practice, this should make no
difference.
7.2 Responses
SIP Responses are distinguished by having a Status-Line for a start-
line. A Status-Line, consists of the protocol version followed by a
numeric Status-Code and its associated textual phrase, with each
element separated by SP characters. No CR or LF is allowed except in
the final CRLF sequence.
SIP-version Status-Code Reason-Phrase
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The Status-Code is a 3-digit integer result code that indicates the
outcome of an attempt to understand and satisfy a request. The
Reason-Phrase is intended to give a short textual description of the
Status-Code. The Status-Code is intended for use by automata, whereas
the Reason-Phrase is intended for the human user. A client is not
required to examine or display the Reason-Phrase.
The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. For this
reason, any response with a status code between 100 and 199 is
referred to as a "1xx response", any response with a status code
between 200 and 299 as a "2xx response", and so on. SIP/2.0 allows 6
values for the first digit:
1xx: Informational -- request received, continuing to process
the request;
2xx: Success -- the action was successfully received,
understood, and accepted;
3xx: Redirection -- further action needs to be taken in order to
complete the request;
4xx: Client Error -- the request contains bad syntax or cannot
be fulfilled at this server;
5xx: Server Error -- the server failed to fulfill an apparently
valid request;
6xx: Global Failure -- the request cannot be fulfilled at any
server.
Full definitions of these classes and each registered code appear in
Section 23.
7.3 Header Fields
SIP header fields are similar to HTTP header fields in both syntax
and semantics. In particular, SIP header fields follow the [H4.2]
definitions of syntax for message-header, the rules for extending
header fields over multiple lines, the use of multiple message-header
fields with the same field-name, and the rules regarding ordering of
header fields.
7.3.1 Header Field Format
Header fields follow the same generic header format as that given in
Section 3.1 of RFC 822 [12]. Each header field consists of a field
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name followed by a colon (":") and the field value.
field-name: field-value
Note that the formal grammar for a message-header specified in
Section 26 allow for an arbitrary amount of whitespace on either side
of the colon. No space before the colon and a single space (SP)
between the colon and the field-value is preferred. That is,
Subject: lunch
Subject : lunch
Subject :lunch
Subject: lunch
are all valid, and equivalent, but the last is the preferred form.
Header fields can be extended over multiple lines by preceding each
extra line with at least one SP or horizontal tab (HT). The line
break and the whitespace at the beginning of the next line are
treated as a single SP character. Thus the following are equivalent:
Subject: I know you're there, pick up the phone and talk to me!
Subject: I know you're there,
pick up the phone
and talk to me!
The relative order of header fields with different field names is not
significant. The relative order of those with the same field name is
important. Multiple header fields with the same field-name may be
present in a message if and only if the entire field-value for that
header field is defined as a comma-separated list (i.e., #(values)).
It MUST be possible to combine the multiple header fields into one
"field-name: field-value" pair, without changing the semantics of the
message, by appending each subsequent field-value to the first, each
separated by a comma.
Implementations MUST be able to process multiple header fields with
the same name in any combination of the single-value-per-line or
comma-separated value forms.
The following blocks of headers are valid and equivalent:
Route: sip:alice@atlanta.com
Subject: Lunch
Route: sip:bob@biloxi.com
Route: sip:carol@chicago.com
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Route: sip:alice@atlanta.com, sip:bob@biloxi.com
Route: sip:carol@chicago.com
Subject: Lunch
Subject: Lunch
Route: sip:alice@atlanta.com, sip:bob@biloxi.com, sip:carol@chicago.com
Each of the following blocks is valid but not equivalent to the
others:
Route: sip:alice@atlanta.com
Route: sip:bob@biloxi.com
Route: sip:carol@chicago.com
Route: sip:bob@biloxi.com
Route: sip:alice@atlanta.com
Route: sip:carol@chicago.com
Route: sip:alice@atlanta.com,sip:carol@chicago.com,sip:bob@biloxi.com
The format of a header field-value is defined per header-name. It
will always be either an opaque sequence of TEXT-UTF8 octets, or a
combination of whitespace, tokens, separators, and quoted strings.
Many of them will adhere to the general form of a value followed by a
semi-colon separated sequence of parameter-name, parameter-value
pairs:
field-name: field-value *(;parameter-name=parameter-value)
When comparing headers, field names are always case-insensitive.
Unless otherwise stated in the definition of a particular header
field, field values, parameter names, and parameter values (tokens in
general) are case-insensitive. Unless specified otherwise, values
expressed as quoted strings are case-sensitive.
The following are equivalent:
Contact: <sip:alice@atlanta.com>;expires=3600
CONTACT: <sip:alice@atlanta.com>;ExPiReS=3600
Contact-Disposition: session;handling=optional
contact-disposition: Session;HANDLING=OPTIONAL
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The following are not equivalent;
Warning: 370 devnull "Choose a bigger pipe"
Warning: 370 devnull "CHOOSE A BIGGER PIPE"
7.3.2 Header Field Classification
Some header fields only make sense in requests or responses. These
are called Request Header Fields and Response Header fields
respectively. Those header fields that can appear in either a
request or response are called General Header Fields. If a header
appears in a message not matching its category (such as a request
header in a response), it MUST be ignored. Section 22 defines the
classification of each header.
7.3.3 Compact Form
SIP provides a mechanism to represent common header fields in an
abbreviated form. This may be useful when messages would otherwise
become to large to be carried on the transport available to it
(exceeding the MTU when using UDP for example). These compact forms
are defined in Section 22. A compact form MAY be substituted for the
longer form of a header name at any time without changing the
semantics of a the message. Multiple header fields in a message with
the same header name MAY appear with an arbitrary mix of its long and
short field name form. Implementations MUST accept both the long and
short forms of each header name.
7.4 Bodies
Requests, including new requests defined in extensions to this
specification, MAY contain message bodies unless otherwise noted.
For response messages, the request method and the response status
code determine the type and interpretation of any message body. All
responses MAY include a body.
7.4.1 Message Body Type
The Internet media type of the message body MUST be given by the
Content-Type header field. If the body has undergone any encoding
(such as compression) then this MUST be indicated by the Content-
Encoding header field, otherwise Content-Encoding MUST be omitted. If
applicable, the character set of the message body is indicated as
part of the Content-Type header-field value.
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The "multipart" MIME type defined in RFC 2046 [14] MAY be used within
the body of the message.
Implementations that send requests containing multipart message
bodies MUST be able to send a session description as a non-multipart
message body if the remote implementation requests this through an
Accept header field.
7.4.2 Message Body Length
The body length in bytes is provided by the Content-Length header
field. Section 22.14 describes the necessary contents of this header
in detail.
The "chunked" transfer encoding of HTTP/1.1 MUST NOT be used for SIP.
(Note: The chunked encoding modifies the body of a message in order
to transfer it as a series of chunks, each with its own size
indicator.)
7.5 Framing SIP messages
Unlike HTTP, SIP MAY use UDP or other unreliable datagram protocols.
Each such datagram carries one request or response. Datagrams,
including all headers, SHOULD NOT be larger than the path maximum
transmission unit (MTU) if the MTU is known, or 1500 bytes if the MTU
is unknown. However, implementations MUST be able to handle messages
up to the maximum datagram packet size. For UDP, this size is 65,535
bytes, including headers.
The MTU of 1500 bytes accommodates encapsulation within the
"typical" ethernet MTU without IP fragmentation. Recent
studies [15] indicate that an MTU of 1500 bytes is a
reasonable assumption. The next lower common MTU values are
1006 bytes for SLIP and 296 for low-delay PPP (RFC 1191
[16]). Thus, another reasonable value would be a message
size of 950 bytes, to accommodate packet headers within the
SLIP MTU without fragmentation.
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with one or more CRLF, CR, or LFs, these characters MUST be ignored.
Likewise, Implementations processing SIP messages over stream
oriented transports MUST ignore noise between messages.
8 General User Agent Behavior
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A user agent represents an end system. It contains a User Agent
Client (UAC), which generates requests, and a User Agent Server (UAS)
which responds to them. A UAC is capable of generating a request
based on some external stimulus (the user clicking a button, or a
signal on a PSTN line), and processing a response. A UAS is capable
of receiving a request, and generating response, based on user input,
external stimulus, the result of a program execution, or some other
mechanism.
When a UAC sends a request, it will pass through some number of proxy
servers, which forward the request towards the UAS. When the UAS
generates a response, the response is forwarded towards the UAC.
UAC and UAS procedures depend strongly on two factors. First, whether
the request or response is inside or outside of a dialog, and second,
based on the method of a request. Dialogs are discussed thoroughly in
Section 12; they represent a peer-to-peer relationship between user
agents, and are established by specific SIP methods, such as INVITE.
In this section, we discuss the method independent rules for UAC and
UAS behavior when processing of requests that are outside of a
dialog. This includes, of course, the requests which themselves
establish a dialog.
8.1 UAC Behavior
8.1.1 Generating the Request
A valid SIP request formulated by a UAC MUST at a minimum contain the
following headers: To, From, CSeq, Call-ID, and Via; all of these
headers are mandatory in all SIP messages. These five headers are the
fundamental building blocks of a SIP message, as they jointly provide
for most of the critical message routing services including the
addressing of messages, the routing of responses, ordering of
messages, and the unique identification of transactions.
Examples of requests send outside of a dialog include an INVITE to
establish a session (Section 13) and an OPTIONS to query for
capabilities (Section 11).
The To general-header field first and foremost specifies the desired
"logical" recipient of the request, or the address of record of the
user or resource that is the target of this request. This may or may
not be the ultimate recipient of the request. The To header MAY
contain a SIP URI, but it may also make use of other URI schemes (for
example as the tel URL [13]) when appropriate. The To header field
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allows for a display name; this is meant to contain a descriptive
version of the URI, and is intended to be displayed to a user
interface.
A UAC may learn how to populate the To header field for a particular
request in a number of ways. Usually the user will suggest the To
header field through a human interface, perhaps inputting the URI
manually or selecting it from some sort of address book.
A request outside of a dialog MUST NOT contain a tag; the tag in the
To field of a request identifies the peer of the dialog. Since no
dialog is established, no tag is present.
For further information on the To header see Section 22.37.
The following is an example of valid To header:
To: Carol <sip:carol@chicago.com>
The From general-header field indicates the logical identity of the
initiator of the request, possibly the user's address of record. Like
the To field, it contains a URI and optionally a display name. It is
used by SIP elements to determine processing rules to apply to a
request (for example, automatic call rejection). As such, it is very
important that the URI not contain IP addresses or host names, since
these are not logical names.
The From header field allows for a display name; this is meant to
contain a descriptive version of the URI, and is intended to be
displayed to a user interface. A UAC SHOULD use the display name
"Anonymous" if the identity of the client is to remain hidden.
Usually the value that populates the From header field in requests
generated by a particular user agent is pre-provisioned by the user
or by the administrators of the user's local domain. If a particular
user agent is used by multiple users, it might have switchable
profiles that include a URI corresponding to the identity of the
profiled user. Recipients of requests can authenticate the originator
of a request in order to ascertain that they are who their From
header field claims they are (see Section 20.2 for more on
authentication).
The From field MUST contain a new "tag" parameter, chosen by the UAC.
See Section 21.3 for details on choosing a tag.
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For further information on the From header see Section 22.20.
Examples:
From: "Bob" <sip:bob@biloxi.com> ;tag=a48s
From: sip:+12125551212@server.phone2net.com;tag=887s
From: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8
The Call-ID general-header field acts as a unique identifier to group
together series of messages. It is always the same for all requests
and responses sent by either UA in a dialog. It is also the same in
each registration from a UA within a single boot cycle.
In a new request created by a UAC outside of any dialog, unless
overridden by method specific behavior, it MUST be selected by the
UAC as a a globally unique identifier over space and time; all SIP
user agents must have a means to guarantee that the Call-ID headers
they produce will not be inadvertently generated by any other user
agent.
Use of cryptographically random identifiers [17] in the generation of
Call-IDs is RECOMMENDED. Implementations MAY use the form
"localid@host". Call-IDs are case-sensitive and are simply compared
byte-by-byte.
Using cryptographically random identifiers provides some
protection against session hijacking, and reduces the
likelihood of unintentional Call-ID collisions.
No provisioning or human interface is required for the selection of
the Call-ID header field value for a request.
For further information on the Call-ID header see Section 22.8.
Example:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
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The Cseq header serves as a way to identify and order transactions.
It consists of a sequence number and a method. The method MUST match
that of the request. The sequence number value is arbitrary, but MUST
be expressible as a 32-bit unsigned integer and MUST be less than
2**31.
As long as it follows the above guidelines, a client may use any
mechanism it would like to select CSeq header field values.
For further information on the CSeq header see Section 22.16.
Example:
CSeq: 4711 INVITE
The Via header is used to determine the transport to use for sending
a request, and for identifying the IP address and port where the
response is to be sent. Rules for setting and using the values in
this header are described in Section 19.
For further information on the Via header see Section 22.40.
The Contact header provides a SIP URI that can be used to contact
that specific instance of the user agent for subsequent requests. The
Contact header MUST be present in any request that can result in the
establishment of a dialog. For the methods defined in this
specification, that includes only the INVITE request. For these
requests, the scope of the Contact is the dialog. That is, the
Contact header refers to the URL that the UA would like to receive
requests at, for requests that are part of that dialog only. Only a
single URI MUST be present.
For further information on the Contact header, see Section 22.10.
The initial Request-URI of the message SHOULD be set to the value of
the URI in the To field. One notable exception is the REGISTER
method; behavior for setting the Request-URI of register is given in
Section 10. Another exception is the case of pre-existing Route
headers; in that case, the procedures of Section 12.2.1.1 as they
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pertain to the Request- URI are followed, even though there is no
dialog.
8.1.1.8 Supported and Require
If the UAC supports extensions to SIP that can be applied by the
server to the response, the UAC SHOULD include a Supported header in
the request listing the option tags for those extensions.
The option-tags listed MUST only refer to extensions defined in
standards track RFCs. This is to prevent servers from insisting that
clients implement non-standard, vendor defined features in order to
receive service. Extensions defined by experimental and informational
RFCs are explicitly excluded from usage with the Supported header in
a request, since they too are often used to document vendor defined
extensions.
If the UAC wishes to insist that a UAS understand an extension that
the UAC will apply to the request in order to process the request, it
MUST insert a Require header into the request listing the option tag
for that extension. If the UAC wishes to apply an extension to the
request and insist that a proxy understand that extension, it MUST
insert a Proxy-Require header into the request listing the option tag
for that extension.
8.1.1.9 Additional Message Components
After a new request has been created, the headers described above
have been properly constructed, any additional optional headers are
added, as are any headers specific to the method.
SIP requests MAY contain a MIME-encoded message-body. Regardless of
the type of body that a request contains, certain headers must be
formulated to characterize the contents of the body. For further
information on these headers see Section 7.4.
8.1.2 Sending the Request
The destination for the request is then computed. This can be a
preconfigured IP address, port and transport of an outbound proxy, or
it can be determined through DNS procedures applied to the Request-
URI. These procedures are described in Section 24, which yield an
ordered set of address, port and transports to attempt. The UAC
SHOULD follow the procedures defined there for stateful elements,
trying each address until a server is contacted. Each try constitutes
a new transaction, and therefore a new client transaction MUST be
constructed for each.
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8.1.3 Processing Responses
Responses are first processed by the transport layer, and then passed
up to the transaction layer. The transaction layer performs its
processing, and then passes it up to the TU. The majority of response
processing in the TU is method specific. However, there are some
general behaviors independent of the method.
8.1.3.1 Unrecognized Responses
A UAC MUST treat any response they do not recognize as being
equivalent to the x00 response code of that class, and MUST be able
to process the x00 response code for all classes. For example, if a
UAC receives an unrecognized response code of 431, it can safely
assume that there was something wrong with its request and treat the
response as if it had received a 400 (Bad Request) response code.
If more than one Via header field is present in a response, the UAC
SHOULD discard the message.
The presence of additional Via header fields that precede
the originator of the request suggests that the message was
misrouted or possibly corrupted.
8.1.3.3 Processing 3xx responses
Upon receipt of a redirection response (e.g. a 3xx response status
code), clients SHOULD use the URI(s) in the Contact header field to
formulate a new request.
To do that, the client copies all but the "method-param" and "header"
elements of the addr-spec part of the Contact header field into the
Request-URI of the request. It uses the "header" parameters to create
headers for the request, replacing any default headers normally used.
In all other respects, requests sent upon receipt of a redirect
response SHOULD re-use the headers and bodies of the original
request.
The Contact values present in redirection responses SHOULD NOT be
cached across calls, as they may not represent the most desirable
location for a particular destination address.
8.1.3.4 Processing 4xx responses
Certain 4xx response codes require specific UA processing,
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independent of the method.
If a 401 or 407 response is received, the UAC SHOULD follow the
authorization procedures of Section 20.2.2 and Section 20.2.3 to
retry the request with credentials.
If a 413 response is received (Section 23.4.11), it means that the
request contained a body that was longer than the UAS was willing to
accept. If possible, the UAC SHOULD retry the request, either
omitting the body or using one of a smaller length.
If a 415 response is received (Section 23.4.13), it means the request
contained media types not supported by the UAS. The UAC SHOULD retry
sending the request, this time only using content with types listed
in the Accept header in the response, with encodings listed in the
Accept-Encoding header in the response, and with languages listed in
the Accept-Language in the response.
If a 420 response is received (Section 23.4.14), it means the request
contained a Require or Proxy-Require header listing an option-tag for
a feature not supported by a proxy or UAS. The UAC SHOULD retry the
request, this time omitting any extensions listed in the Unsupported
header in the response.
In all of the above cases, retrying the request is accomplished by
creating a new request with the appropriate modifications. This new
request SHOULD have the same value of the Call-ID, To, and From of
the previous request, but the CSeq should contain a new sequence
number that is one higher than the previous.
With other 4xx responses, a retry may or may not be possible
depending on the method and the use case.
8.2 UAS Behavior
When a request outside of a dialog is processed by a UAS, there are a
set of processing rules which are followed, independent of the
method. Section 12 gives guidance on how a UAS can tell whether a
request is inside or outside of a dialog.
8.2.1 Authentication/Authorization
A UAS MAY authenticate the originator of a request, and this process
may require the server to issue a challenge for credentials. The
required behavior is independent of the method of the request, and is
detailed in Section 20.2.
8.2.2 Method Inspection
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Once a request is authenticated (or no authentication was desired),
the UAS MUST inspect the method of the request. If the UAS does not
support the method of a request it MUST generate a 405 (Method Not
Allowed) response. Procedures for generation of responses are
described in Section 8.2.7. The UAS MUST also add an Allow header to
the 405 response. The Allow header field MUST list the set of methods
supported by the UAS generating the message.
The Allow header is presented in Section 22.5.
If the method is one supported by the server, processing continues.
8.2.3 Header Inspection
If a UAS does not understand a header field in a request (i.e. the
header is not defined in this specification or in any supported
extension), the server MUST ignore that header and continue
processing the message. A UAS SHOULD ignore any malformed headers
which are not necessary for processing requests.
8.2.3.1 To and Request-URI
The To header field identifies the original recipient of the request
designated by the user identified in the From field. The original
recipient may or may not be the UAS processing the request, do to
call forwarding or other proxy operations. A UAS MAY apply any policy
it wishes in determination of whether to accept requests when the To
field is not the identity of the UAS. However, it is RECOMMENDED that
a UAS accept requests even if they do not recognize the URI scheme
(e.g., a tel: URI) in the To header, or if the To header does not
address a known or current user of this UAS. If, on the other hand,
the UAS decides to reject the request, it SHOULD generate a response
with a 403 status code and send it to the server transaction for
transmission.
However, the Request-URI identifies the UAS that is to process the
request. If the Request-URI does not identify an address that the UAS
is willing to accept requests for, it SHOULD reject the request with
a 404 (Not Found) response. If the Request-URI does not provide
sufficient information for the UAS to determine whether it is willing
to process the request, it SHOULD return a 485 (Ambiguous) response.
This response SHOULD contain a Contact header field containing URIs
of new addresses to be tried.
Typically, a UA which uses the REGISTER method to bind its address of
record to a specific contact address, will see requests whose
Request-URI equals those contact addresses.
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Assuming the UAS decides that it is the proper element to process the
request, it examines the Require header field, if present.
The Require general-header field is used by UAC to tell UAS about SIP
extensions that the UAC expects the UAS to support in order to
properly process the request. If a UAS does not understand an option
listed in a Require header field, it MUST respond by generating a
response with status code 420 (Bad Extension). The UAS MUST add a
Unsupported, and list in it those options it does not understand
amongst those in the Require header of the request. Upon receipt of
the 420 the client SHOULD retry the request, this time without using
those extensions listed in the Unsupported header in the response.
Example:
UACC->UAS: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells
UASS->UAC: SIP/2.0 420 Bad Extension
Unsupported: com.example.billing
This is to make sure that the client-server interaction
will proceed without delay when all options are understood
by both sides, and only slow down if options are not
understood (as in the example above). For a well-matched
client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation
mechanisms. In addition, it also removes ambiguity when the
client requires features that the server does not
understand. Some features, such as call handling fields,
are only of interest to end systems.
8.2.4 Content Processing
Assuming the UAS understands any extensions required by the client,
the UAS examines the body of the message, and the headers that
describe it. If there are any bodies whose type (indicated by the
Content-Type), language (indicated by the Content-Language) or
encoding (indicated by the Content-Encoding) are not understood, and
that body part is not optional (as indicated by the Content-
Disposition) header, the UAS MUST reject the request with a 415
(Unsupported Media Type) response. The response MUST contain a Accept
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header listing the types of all bodies it understands, in the event
the request contained bodies of types not supported by the UAS. If
the request contained content encodings not understood by the UAS,
the response MUST contain an Accept-Encoding header listing the
encodings understood by the UAS. If the request contained content
with languages not understood by the UAS, the response MUST contain
an Accept-Language header indicating the languages understood by the
UAS.
Beyond these checks, body handling is method and type specific.
For further information on the processing of Content-specific headers
see Section 7.4.
8.2.5 Applying Extensions
A UAS that wishes to apply some extension when generating the
response MUST only do so if support for that extension is indicated
in the Supported header in the request. If the desired extension is
not supported, the server SHOULD rely only on baseline SIP and any
other extensions supported by the client. To ensure that the SHOULD
can be fulfilled, any specification of a new extension MUST include
discussion of how to gracefully return to baseline SIP when the
extension is not present. In rare circumstances, where the server
cannot process the request without the extension, the server MAY send
a 421 (Extension Required) response. This response indicates that the
proper response cannot be generated without support of a specific
extension. The needed extension(s) MUST be included in a Require
header in the response. This behavior is NOT RECOMMENDED, as it will
generally break interoperability.
Any extensions applied to a non-421 response MUST be listed in a
Require header included in the response. Of course, the server MUST
NOT apply extensions not listed in the Supported header in the
request. As a result of this, the Require header in a response will
only ever contain option tags defined in standards track RFCs.
8.2.6 Processing the Request
Assuming all of the checks in the previous subsections are passed,
the UAS processing becomes method specific. Section 10 deals with the
REGISTER request, section 11 deals with the OPTIONS request, section
13 deals with the INVITE request, and section 15 deals with the BYE
request.
8.2.7 Generating the Response
When a UAS wishes to construct a response to a request, it follows
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these procedures. Additional procedures may be needed depending on
the status code of the response and the circumstances of its
construction. These additional procedures are documented elsewhere.
The From field of the response MUST equal the From field of the
request. The Call-ID field of the response MUST equal the Call-ID
field of the request. The Cseq field of the response MUST equal the
Cseq field of the request. The Via headers in the response MUST equal
the Via headers in the request, and MUST maintain the same ordering.
If a request contained a To tag in the request, the To field in the
response MUST equal that of the request. However, if the To field in
the request did not contain a tag, the URI in the To field in the
response MUST equal the URI in the To field in the request.
Additionally, the UAS MUST add a tag to the To field in the response.
This serves to identify the UAS that is responding, possibly
resulting in a component of a dialog ID. The same tag MUST be used
for all responses to that request, both provisional and final.
Procedures for generation of tags are defined in Section 21.3.
8.3 Redirect Servers
In some architectures it may be desirable to reduce the processing
load on proxy servers that are responsible for routing requests by
relying on redirection. Redirection allows servers to push routing
information for a request back in a response to the client, thereby
taking themselves out of the loop of further messaging for this
transaction while still aiding in locating the target of the request.
When the originator of the request receives the redirection it will
send a new request based on the routing information it has received.
By propagating routing information from the core of the network to
its edges, redirection allows for considerable network scalability.
A redirect server is logically constituted of a server transaction
layer and a transaction user that has access to a location service of
some kind (see Section 10 for more on registrars and location
services). This location service is effectively a database containing
mappings between a single URI and a set of one or more alternative
locations at which the target of that URI can be found.
A redirect server does not issue any SIP requests of its own. After
receiving a request other than CANCEL, the server gathers the list of
alternative locations from the location service and either returns a
final response of class 3xx or it refuses the request. For well-
formed CANCEL requests, it SHOULD return a 2xx response. This
response ends the SIP transaction. The redirect server maintains
transaction state for an entire SIP transaction. It is the
responsibility of clients to detect forwarding loops between redirect
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servers.
When a redirect server returns a 3xx response to a request, it
populates the list of (one or more) alternative locations into
Contact headers. An "expires" parameter to the Contact header may
also be supplied to indicate the lifetime of the Contact data.
The Contact header field contains URIs giving the new locations or
user names to try, or may simply specify additional transport
parameters. A 301 or 302 response may also give the same location and
username that was targeted by the initial request but specify
additional transport parameters such as a different server or
multicast address to try, or a change of SIP transport from UDP to
TCP or vice versa.
Note that the Contact header field MAY also refer to a different
entity than the one originally called. For example, a SIP call
connected to GSTN gateway may need to deliver a special informational
announcement such as "The number you have dialed has been changed."
A Contact response header field can contain any suitable URI
indicating where the called party can be reached, not limited to SIP
URIs. For example, it could contain URL's for phones, fax, or irc (if
they were defined) or a mailto: (RFC 2368, [18]) URL.
The "expires" parameter of the Contact header field indicates how
long the URI is valid. The parameter is either a number indicating
seconds or a quoted string containing a SIP-date. If this parameter
is not provided, the value of the Expires header field determines how
long the URI is valid. Implementations MAY treat values larger than
2**32-1 (4294967295 seconds or 136 years) as equivalent to 2**32-1.
Redirect servers MUST ignore features that are not understood
(including unrecognized headers, Required extensions, or even method
names) and proceed with the redirection of the session in question.
If a particular extension requires that intermediate devices support
it, the extension MUST be tagged in the Proxy-Require field as well
(see Section 22.28).
9 Canceling a Request
The previous section has discussed general UA behavior for generating
requests, and processing responses, for requests of all methods. In
this section, we discuss a general purpose method, called CANCEL.
The CANCEL request, as the name implies, is used to cancel a previous
request sent by a client. Specifically, it asks the user agent server
to cease processing the request, and generate an error response to
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that request. CANCEL has no effect on a request that has already been
responded to. Because of this, it is most useful to CANCEL requests
which can take a long time to respond to. For this reason, CANCEL is
most useful for INVITE requests, which can take a long time to
generate a response. In that usage, a UAS that receives a CANCEL
request for an INVITE, but has not yet sent a response, would "stop
ringing", and then respond to the INVITE with a specific error
response (a 487).
Cancel requests can be constructed and sent by any type of client,
including both proxies and user agent servers. Section 15 discusses
under what conditions a UAC would CANCEL an INVITE request, and
Section 16 discusses proxy usage of INVITE.
Because a stateful proxy can generate its own CANCEL, a stateful
proxy also responds to a CANCEL, rather than simply forwarding a
response it would receive from a downstream element. For that reason,
CANCEL is referred to as a "hop-by-hop" request, since it is
responded to at each stateful proxy hop.
9.1 Client Behavior
The following procedures are used to construct a CANCEL request. The
Request-URI, Call-ID, To, the numeric part of CSeq and From header
fields in the CANCEL request MUST be identical to those in the
request being cancelled, including tags. A CANCEL constructed by a
client MUST have only a single Via header, whose value matches the
top Via in the request being cancelled. Using the same values for
these headers allows the CANCEL to be matched with the request it
cancels (Section 9.2 indicates how such matching occurs). However,
the method part of the Cseq header MUST have a value of CANCEL. This
allows it to be identified and processed as a transaction in its own
right (See Section 17).
Once the CANCEL is constructed, the client SHOULD check whether any
response (provisional or final) has been received for the request
being cancelled (herein referred to as the "original request"). The
CANCEL request MUST NOT be sent if no provisional response has been
received, rather, the client MUST wait for the arrival of a
provisional response before sending the request. If the original
request has generated a final response, the CANCEL SHOULD NOT be
sent, as it is an effective no-op, since CANCEL has no effect on
requests which have already generated a final response. When the
client decides to send the CANCEL, it creates a client transaction
for the CANCEL, and passes it the CANCEL request along with the
destination address, port and transport. The destination address,
port, and transport for the CANCEL MUST be identical to those used to
send the original request.
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If it was allowed to send the CANCEL before receiving a
response for the previous request the server could receive
the CANCEL before the original request.
Note that both the transaction corresponding to the original request
and the CANCEL transaction will complete independently. However, a
UAC canceling a request cannot rely on receiving a 487 (Request
Terminated) response for the original request, as an RFC 2543-
compliant UAS will not generate such a response. If there is no final
response for the original request in 64*T1 seconds for an INVITE
transaction, and T3 seconds for a non-INVITE transaction, the client
SHOULD then consider the original transaction cancelled and SHOULD
destroy the client transaction handling the original request.
9.2 Server Behavior
The CANCEL method requests that the TU at the server side cancel a
pending request with the same Call-ID, To, From, top Via header and
Request-URI and CSeq (sequence number only) header field values.
The processing of a CANCEL request at a server depends on the type of
server. A stateless proxy will forward it, a stateful proxy might
respond to it and generate some CANCEL requests of its own, and a UAS
will respond to it. See Section 16.8 for proxy treatment of CANCEL.
When a UAS receives a CANCEL, it looks for any server transactions
which were created by requests with the same To, From, Call-ID, Cseq
numeric value, Request-URI and top Via header. If no matching
transactions are found, the CANCEL is responded to with a 481 (Call
Leg/Transaction Does Not Exist). If the transaction for the original
request still exists, the behavior of the UAS on receiving a CANCEL
request depends on whether it has already sent a final response for
original request. If it has, the CANCEL request has no effect on the
processing of the original request, no effect on any session state,
and no effect on the responses generated for the original request. If
the UAS has not issued a final response for the original request, it
immediately responds to the original request with a 487 (Request
Terminated).
The CANCEL request itself is answered with a 200 (OK) response in
either case. Once the response is constructed it is passed to the
server transaction for the CANCEL request.
10 Registrations
10.1 Overview of Usage
SIP is a protocol that offers a discovery capability. For one user to
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initiate a session with another, SIP must discover the current
host(s) that the called user is reachable at. This discovery process
is accomplished by SIP proxy servers, which are responsible for
receiving a request, determining where to send it based on knowledge
of the location of the user, and then sending it there. To do this,
proxies consult an abstract service known as a location service ,
which provides address bindings for a particular domain. These
address bindings map an incoming SIP URL, sip:bob@Biloxi.com , for
example, to one or more SIP URLs which are somehow "closer" to the
desired user, sip:bob@engineering.Biloxi.com , for example.
Ultimately, a proxy will consult a location service which maps a
received URL to the current host(s) that a user is logged in to.
There are many ways by which the contents of the location service can
be established. One way is administratively. In the above example,
Bob is known to be a member of the engineering department through
access to a corporate database. SIP provides a mechanism, however,
for a user agent to explicitly create a binding in the location
service of a proxy. This mechanism is known as registration.
The process of registration entails sending a REGISTER message to a
special type of UAS known as a registrar. The registrar acts as a
front end to the location service for a domain, reading and writing
mappings based on the contents of the REGISTER messages. This
location service will then be consulted by a proxy server that is
responsible for routing requests for that domain.
SIP does not mandate a particular mechanism for implementing the
location service. The only requirement is that a registrar for some
domain MUST be capable of reading and writing data to the location
service, and a proxy for that domain MUST be capable of reading that
same data. A registrar MAY be co-located with a particular SIP proxy
server for the same domain, allowing usage of an in memory database
for the location service. Usage of a shared database is another
implementation choice. The choice depends entirely on the
architectural requirements (redundancy, scalability, etc) of a
particular deployment.
Registration creates bindings in a location service for a particular
domain that associate an "address of record" URI with one or more
"contact addresses". This means that when a proxy for that domain
receives a request whose request URI matches the address of record,
the proxy will forward the request to the contact addresses
registered to that address of record. Generally, it only makes sense
to register an address of record at a location service for a domain
when requests for that address of record would be routed to that
domain. In most cases, this means that the domain of the registration
will need to match the domain in the URI of the address of record.
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The most important usage of the registration mechanism is to inform a
proxy of the mapping between the address of record and the current
host on which the UA resides. However, the registration process is a
general mechanism for establishing bindings, and can be used for
other purposes (for example, to set up call forwarding).
10.2 Construction of the REGISTER request
Several operations can be performed with a REGISTER method with
respect to a registrar. One of these is the basic registration
operation that is described above, which provides a new binding
between an address of record and one or more contact addresses.
Registration on behalf of a particular address of record may be
performed by a third party if they are authorized to do so. A client
may also remove previous bindings, or query to determine which
bindings are currently in place for an address of record.
Aside from the exceptions noted in this and the following sections,
the construction of the REGISTER method, and behavior of clients
sending a REGISTER is identical to the general UAC behavior described
in Section 8.1 and Section 17.1. Regardless of the operation that is
performed by a REGISTER, the following header fields MUST be
formulated as follows:
Request-URI: The Request-URI names the domain of the location
service that the registration is meant for (e.g.
"chicago.com"). The user name MUST be empty.
To: The To header field contains the address of record whose
registration is to be created or modified. Note that the
initial To header field and the Request-URI field SHOULD
therefore be different in a REGISTER message.
From: The From header field contains the address of record of
the person responsible for the registration, which MAY be
identical to the value of the To header field. For third-
party registrations the From header field and To header
field are different.
Call-ID: All registrations from a user agent client SHOULD use
the same Call-ID header value, at least within the same
reboot cycle.
If different Call-IDs were used for overlapping
REGISTER messages coming from the same client, the
registrar might have trouble determining their
ordering.
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Contact: REGISTER requests MAY contain one or more Contact
header fields. Contact addresses are presented in the
Contact header fields of REGISTER requests.
Note that user agents MUST NOT send a new registration (containing
new Contact header fields, as opposed to a retransmission) until they
have received a response from the registrar for the previous one.
The following optional Contact header parameters also contain
behavior specific to the registration process.
action: The "action" parameter has been deprecated. UACs SHOULD
NOT use the "action" parameter.
expires: The "expires" parameter indicates how long the UAC
would like the binding to be valid. The parameter is either
a number indicating seconds or a quoted string containing a
SIP-date. If this parameter is not provided, the value of
the Expires header field determines how long the binding is
valid. Implementations MAY treat values larger than 2**32-1
(4294967295 seconds or 136 years) as equivalent to 2**32-1.
10.2.1 Adding Bindings with REGISTER
For a simple registration, a REGISTER request sent to a registrar
includes contact addresses to which requests should be forward for
the originating user's address of record. The address of record
itself (i.e. 'sip:carol@chicago.com') MUST populate the To header of
the REGISTER. The Contact header fields of the request typically
contain SIP URIs that identify particular SIP endpoints (i.e.
'sip:carol@cube2214a.chicago.com'), but they MAY use any URI scheme;
this way a SIP UA can choose to register telephone numbers (with the
tel URL, [13]) or email addresses (with a mailto URL, [18]) as
Contacts for an address of record.
For example, if Carol, whose address of record is to register with
the registrar associated with the location service of chicago.com.
This location service would then be accessed by a proxy server that
receives requests targeting users in the chicago.com domain, and
hence new requests for Carol's address of record will be routed to
her SIP endpoint.
Once a client has established bindings at a registrar, it MAY send
subsequent registrations containing new bindings or modifications to
pre-existing bindings as necessary. The 2xx response to the REGISTER
message will contain (in Contact header fields) a complete list of
bindings that have been registered for this address of record at this
registrar.
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bob
+----+
| UA |
| |
+----+
|
|3)INVITE
| carol@chicago.com
chicago.com +--------+ V
+---------+ 2)Store|Location|4)Query +-----+
|Registrar|=======>| Service|<=======|Proxy|sip.chicago.com
+---------+ +--------+=======>+-----+
A 5)Resp |
| |
| |
1)REGISTER| |
| |
+----+ |
| UA |<-------------------------------+
cube2214a| | 6)INVITE
+----+ carol@cube2214a.chicago.com
carol
Figure 2: REGISTER example
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10.2.1.1 Setting the Expiration Interval of Contact Addresses
When a client sends a REGISTER request, it MAY suggest an expiration
interval that indicates how long the client would like the
registration to be valid (although as is detailed in Section 10.3,
the registrar has the ultimate say).
There are two ways in which a client can suggest an expiration
interval for a binding: through an Expires header, or an "expires"
Contact header parameter. The latter allows expiration intervals to
be suggested on a per-binding basis when more than one binding is
given in a single REGISTER, whereas the former suggests an expiration
interval for all Contact header fields that do not contain the
"expires" parameter.
If neither mechanism for expressing a suggested expiration time is
present in a REGISTER, a default suggestion of one hour is assumed.
10.2.1.2 Setting Preference among Contact Addresses
If more than one Contact is sent in a REGISTER, then the registering
UA intends to associate all of the URIs given in these Contact
headers with the address of record present in the To field. This list
can be prioritized with the "q" mechanism.
q: The "q" parameter indicates a relative preference for the
particular Contact header field compared to other bindings
present in this REGISTER message or existing within the
location service of the registrar. For an example of how a
proxy server uses "q" values, see Section 16.5.
10.2.2 Removing Bindings with REGISTER
Registrations are removed from the registrar through an expiration
process; registrations are soft state and need to be refreshed
periodically. A client may attempt to influence the expiration
intervals selected by the registrar as described in Section 10.2.1.
A registering user agent requests the immediate removal of a binding
by specifying an expiration interval of "0" for that contact address
in a REGISTER. It is RECOMMENDED that user agents support this
mechanism so that bindings can be removed (for whatever reason)
before their expiration interval has passed.
The REGISTER-specific Contact header field value of "*" applies to
all registrations, but it MUST only be used when the Expires header
is present with a value of "0".
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Use of the "*" Contact header field value allows a
registering user agent to remove all of its bindings
expediently.
10.2.3 Fetching Bindings with REGISTER
If no Contact headers are present in a REGISTER, then the UA is not
in fact registering any new bindings, and the list of bindings is
therefore left unchanged. As noted above, in a successful response to
this REGISTER message, the complete list of existing bindings is
returned, and thus a REGISTER without Contact headers serves as a
fetch operation.
10.2.4 Refreshing Registrations
When a 2xx response has been received by the client for a REGISTER
request, the client MUST determine when each of the bindings
enumerated in the response needs to be refreshed. This may include
bindings that were registered in previous REGISTER transactions.
Since the list of bindings returned in the response to a REGISTER may
contain bindings that were not included in this REGISTER transaction,
the client must correlate Contact header fields in the response with
the Contact header fields it sent in the request in order to
establish proper expiration timers. This correlation should be
performed in accordance with the URI comparison rules given in
Section 21.1.4.
The registering UA MUST re-register each contact address at least as
often as the mandated expiration interval. A REGISTER that refreshes
a binding SHOULD have the same Call-ID as the request which created
the binding. The CSeq header SHOULD have a numeric sequence number
that is one higher than the value sent in the last request with the
same Call-ID.
Note that a UA MUST must update its expiration timers for refreshing
each binding every time it receives a response to a registration
request.
Registration refreshes SHOULD be sent to the same address as the
original registration, unless redirected.
10.2.5 Discovering a Registrar
Depending on the policy of their administrative domain, SIP UAs can
be configured with the address of a local registrar. Some UAs may be
equipped with protocol tools (outside the scope of SIP) that allow
them to discover their local registrar dynamically.
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Note that as an alternate means of discovering a registrar if no
local registrar is configured in the user agent, clients MAY register
via multicast. Multicast registrations are addressed to the well-
known "all SIP servers" multicast address "sip.mcast.net"
(224.0.1.75). This request MUST be scoped to ensure it is not
forwarded beyond the boundaries of the administrative system. This
MAY be done with either TTL or administrative scopes (see [19]),
depending on what is implemented in the network. SIP user agents MAY
listen to that address and use it to become aware of the location of
other local users (see [20]); however, they do not respond to the
request.
Multicast registration may be inappropriate in some
environments, for example, if multiple businesses share the
same local area network.
If a SIP UA knows of an appropriate registrar it SHOULD attempt to
register with this server periodically - management of registration
intervals is detailed below.
10.3 Processing of REGISTER at the Registrar
A registrar is a UAS that responds to a REGISTER request, and stores
the information gathered from that request in a location service that
is in turn accessible to proxy servers within its administrative
domain. A registrar handles requests as a UAS (in conformity with
Section 8.2 and Section 17.2) but it accepts only the REGISTER method
and generates only the responses detailed in this section. Note that
the REGISTER method also does not support the Record-Route or Route
header, and that proxy servers MUST NOT add Record-Route headers to
REGISTER requests.
A registrar must know (through provisioning or some other mechanism)
the set if administrative domain(s) for which its associated location
service(s) are responsible. REGISTER requests MUST be processed by a
registrar in the order that they are received.
Upon the arrival of a REGISTER message, the registrar MUST inspect
the Request-URI to determine whether it has access to a location
service responsible for the domain to which this request is
addressed. If this message is for some other administrative domain,
then if the registrar can act as a proxy server, it SHOULD forward
the request to the addressed domain (following the general behavior
for proxying messages described in Section 16).
When a registrar receives a REGISTER message, it is RECOMMENDED that
the registrar authenticate the user agent client. Mechanisms for the
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authentication of SIP user agents are described in Section 20.2;
registration behavior in no way overrides the generic authentication
framework for SIP. If no authentication mechanism is available, the
registrar MAY take the From address as the asserted identity of the
originator of the request.
Once the identity of the registering user has been ascertained, it is
RECOMMENDED that the registrar determine if the authenticated user
agent is authorized to request and/or modify registrations for this
address of record. For example, a registrar might consult a
authorization database (directly or through an appropriate protocol)
that maps credentials or other tokens of identity resulting from
authentication to one or more addresses of record for which this
identity is responsible.
Note that in architectures that support third-party
registration, one entity may be responsible for updating
the registrations associated with multiple addresses of
record.
When the registrar has determined that the client is permitted to
make the request, the registrar MUST extract the address of record
from the To header field of the REGISTER. Note that the registrar
MUST extract the entire To header field URI in order to use it as an
index in the location service.
Next, the registrar MUST query its location service (the repository
of previously registered bindings) for the set of bindings associated
with this address of record. If the address of record is not valid
for this administrative domain (for example, because the username is
not assigned), then the registration attempt fails (see below). A
full URI comparison (as described in Section 21.1.4) MUST be
performed to determine whether a given binding matches this address
of record.
The registrar now MUST extract all the Contact header fields from the
REGISTER message (note that there may be no Contact header field).
Each contact address in a REGISTER MUST now be compared to all
existing registrations at this location service according to the
rules in Section 21.1.4. Note that URIs other than SIP URIs in
contact addresses MUST be compared according to the standard URI
equivalency rules for the URI schema in question.
If a match is found among pre-existing registrations, the registrar
MUST copy all parameters associated with the current Contact header
field from the REGISTER message into the pre-existing binding in its
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location service (overwriting with changed values any existing
parameters as necessary, with the exception of "expires"). Expiration
intervals for this contact address MUST also be reset, based on any
suggested expiration in the REGISTER (remember that this can be "0").
If no match is found among the set of pre-existing registrations, the
registrar MUST create a new binding in its location service between
the address of record and the current Contact header field. All
Contact header field parameters are copied verbatim into this new
binding (again with the exception of "expires"). An expiration
interval MUST be selected by the registrar, taking into account any
suggested expiration for this contact address in the REGISTER.
Allowing the registrar to set the registration interval
protects it against excessively frequent registration
refreshes while limiting the state that it needs to
maintain and decreasing the likelihood of registrations
going stale.
The expiration interval mandated by the registrar may be either
longer or shorter than the interval suggested by the sender of the
REGISTER, though the registrar SHOULD abide by the registering
client's suggestion.
A server MAY decide to lengthen the expiration interval if
the refresh rate of a particular client exceeds a
threshold, for example.
After the expiration interval selected by the registrar for a binding
has passed, if the binding has not been refreshed (increasing the
expiration interval), the registrar SHOULD silently discard the
binding.
Once all bindings in the location service have been updated to
reflect any changes present to contact addresses in the REGISTER
message, the registrar MUST remove any bindings that expire
immediately.
The REGISTER might have set the expiration interval for
some bindings to "0" to remove them before their expiration
interval passes.
Finally, the registrar must generate a response. If the address of
record given in the To header field of the REGISTER method is valid
for its administrative domain, then a 200 response MUST be sent,
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which MUST contain a complete list (within Contact header fields) of
the currently valid bindings in the location service associated with
the address of record contained in the To field of the REGISTER
request. This list MAY be empty (in which case the 200 would not
contain any Contact headers).
In a successful response to a REGISTER, wherein the bindings for this
address of record are enumerated as described above, the registrar
MUST supply an expiration interval for each contact address in either
an "expires" parameter of a Contact header or an Expires header. This
interval specifies the expiration interval that has been mandated by
the registrar (taking into account the registering UA's suggestion).
If the registration failed because the address of record contained in
the To field of the REGISTER is not valid for this domain, then a 404
MUST be sent.
11 Querying for Capabilities
The SIP method OPTIONS allows a client to query another client or
server as to its capabilities. This allows a client to discover
information about the methods, content types, extensions, codecs etc.
supported without actually "ringing" the other party. For example,
before a client inserts a Require header field into an INVITE listing
an option that it is not certain the destination UAS supports, the
client can query the destination UAS with an OPTIONS to see if this
option is returned in a Supported header field.
The target of the OPTIONS request is identified by the Request-URI,
which could identify another User Agent or a SIP Server.
Alternatively, a server receiving an OPTIONS request with a Max-
Forwards header value of 0 MAY respond to the request regardless of
the Request-URI.
This behavior is common with HTTP/1.1.
An OPTIONS request sent as part of an established dialog does not
have any impact on the dialog.
11.1 Construction of OPTIONS Request
An OPTIONS request is constructed using the standard rules for a SIP
request as discussed Section 8.1.1.
A Contact header field MAY be present in an OPTIONS.
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OPEN ISSUE #197: What is the semantic of this Contact
An Accept header field SHOULD be included to indicate the type of
message body the UAC wishes to receive in the response.
Example OPTIONS request:
OPTIONS sip:carol@chicago.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=23411513a6
Via: SIP/2.0/UDP 10.1.3.3:5060
To: <sip:carol@chicago.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 63104 OPTIONS
Contact: <sip:alice@10.1.3.3>
Accept: application/sdp
Contact-Length: 0
11.2 Processing of OPTIONS Request
The response to an OPTIONS is constructed using the standard rules
for a SIP response as discussed in Section 8.2.7. The response code
chosen is the same that would have been chosen had the request been
an INVITE. That is, a 200 (OK) would be returned if the UAS is ready
to accept a call, a 486 (Busy Here) would be returned if the UAS is
busy, etc. This allows an OPTIONS request to be used to determine the
basic state of a UAS, which can be an indication of whether the UAC
will accept an INVITE request.
Note that this use of OPTIONS has limitations due the differences in
proxy handling of OPTIONS and INVITE requests. While a forked INVITE
can result in multiple 200 OK responses being returned, a forked
OPTIONS will only result in a single 200 OK response, since it is
treated by proxies using the non-INVITE handling. See Section 13.2.1
for the normative details.
Allow, Accept, Accept-Encoding, Accept-Language, and Supported header
fields SHOULD be present in a 200 OK response to an OPTIONS request.
A Contact header field MAY be present in a 200 OK response.
A Warning header field MAY be present.
A message body MAY be sent, the type of which is determined by the
Accept header in the OPTIONS request.
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Example OPTIONS response (corresponding to the request in Section
11.1):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=23411513a6
Via: SIP/2.0/UDP 10.1.3.3:5060
To: <sip:carol@chicago.com>;tag=93810874
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 63104 OPTIONS
Contact: <sip:carol@10.3.6.6>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE
Accept: application/sdp
Accept-Encoding: gzip
Accept-Language: en
Supported: foo
Content-Type: application/sdp
Contact-Length: 274
v=0
o=carol 28908764872 28908764872 IN IP4 10.3.6.6
s=-
t=0 0
c=IN IP4 10.3.6.6
m=audio 0 RTP/AVP 0 1 3 99
a=rtpmap:0 PCMU/8000
a=rtpmap:1 1016/8000
a=rtpmap:3 GSM/8000
a=rtpmap:99 SX7300/8000
m=video 0 RTP/AVP 31 34
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
12 Dialogs
A key concept for a user agent is that of a dialog. A dialog
represents a peer- to-peer SIP relationship between a two user agents
that persists for some time. The dialog facilitates sequencing of
messages between the user agents, and proper routing of requests
between both them. The dialog represents a context in which to
interpret SIP messages. The previous section discussed method
independent UA processing for requests and responses outside of a
dialog. This section discusses how those requests and responses are
used to construct a dialog, and then how subsequent requests and
responses are sent within a dialog.
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A dialog is identified at each UA with a dialog ID, which consists of
a Call-ID value, a local URI and local tag (together called the local
address), and a remote URI and remote tag (together called the remote
address). The dialog ID at each UA involved in the dialog is not the
same. Specifically, the local URI and local tag at one UA are
identical to the remote URI and remote tag at the peer UA. The tags
are opaque tokens that facilitate the generation of unique dialog
IDs.
A dialog ID is also associated with all responses, and with any
request that contains a tag in the To field. The rules for computing
the dialog ID of a message depend on whether the entity is a UAC or
UAS. For a UAC, the Call-ID value of the dialog ID is set to the
Call-ID of the message, the remote address is set to the To field of
the message, and the local address is set to the From field of the
message (these rules apply to both requests and responses). As one
would expect, for a UAS, the Call-ID value of the dialog ID is set to
the Call-ID of the message, the remote address is set to the From
field of the message, and the local address is set to the To field of
the message.
A dialog contains certain pieces of state needed for further message
transmissions within the dialog. This state consists of the Call-ID,
a local sequence number (used to order requests from the UA to its
peer), a remote sequence number (used to order requests from its peer
to the UA), and a route set, which is an ordered list of URIs. The
route set is the set of servers that need to be traversed to send a
request to the peer. A dialog can also be in the "early" state, which
occurs when it is created with a provisional response, and then
transition to the "established" state when the final response comes.
12.1 Creation of a Dialog
Dialogs are created through the generation of non-failure responses
to requests with specific methods. Within this specification, only
the 2xx and 1xx responses to INVITE establish a dialog. A dialog
established by a non-final response to a request is called an early
dialog. Extensions MAY define other means for creating dialogs.
Section 13 gives more details that are specific to the INVITE method.
Here, we describe the process for creation of dialog state that is
not dependent on the method.
When a UAS responds to a request with a response that establishes a
dialog (such as a 2xx to INVITE), the UAS MUST copy all Record-Route
headers from the request into the response, and MUST maintain the
order of those headers. This includes the URIs, URI parameters, and
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any Record-Route header parameters, whether they are known or unknown
to the UAS. The UAS MUST add a Contact header field to the response.
The Contact header field contains an address where the UAS would like
to be contacted for subsequent requests in the dialog (which includes
the ACK for a 2xx response in the case of an INVITE). Generally, the
host portion of this URI is the IP address of the host, or its FQDN.
The URI provided in the Contact header MUST be a SIP URL.
The UAS then constructs the state of the dialog. This state MUST be
maintained for the duration of the dialog. First, the route set MUST
be computed by following these steps:
1. The list of URIs in the Record-Route headers in the
request, if present, are taken, including any URI
parameters.
2. The URI in the Contact header from the request if present,
is taken, including any URI parameters. The URI is appended
to the bottom of the list of URIs from the previous step.
Contact was not mandatory in RFC2543. Thus, if the UAS
is talking to an older UAC, the UAC might not have
inserted the Contact header.
3. The resulting list of URIs is called the route set
These rules clearly imply that a UA MUST be able to parse
and process Record-Route header fields. This is a change
from RFC2543, where all record-route and route processing
was optional for user agents.
It is possible for the route set to be empty. This will occur if
neither Record-Route headers nor a Contact header were present in the
request. The UAS MUST also remember whether the bottom-most entry in
the route set was constructed from a Contact header or not. This is
effectively a boolean value, which we refer to as CONTACT_SET. This
is needed in order for the UA to determine whether the bottom most
value can be updated from subsequent requests; if it was constructed
from a Contact, it can be updated.
The remote sequence number sequence number MUST be set to the value
of the sequence number in the Cseq header of the request. The local
sequence number MUST be empty. The call identifier component of the
dialog ID MUST be set to the value of the Call-ID in the request. The
local address component of the dialog ID MUST be set to the To field
in the response to the request (which therefore includes the tag),
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and the remote address component of the dialog ID MUST be set to the
From field in the request. A UAS MUST be prepared to receive a
request without a tag in the From field, in which case the tag is
considered to effectively have a value of null.
This is to maintain backwards compatibility with RFC2543,
which did not mandate From tags.
When a UAC receives a response that establishes a dialog, it
constructs the state of the dialog. This state MUST be maintained for
the duration of the dialog. First, the route set MUST be computed by
following these steps:
1. The list of URIs present in the Record-Route headers in the
response are taken, if present, including all URI
parameters, and their order is reversed.
2. The URI in the Contact header from the response, if
present, is taken, including all URI parameters, and
appended to the end of the list from the previous step.
3. The list of URIs resulting from the above two operations is
referred to as the route set
It is possible for the route set to be empty. This will occur if
neither Record-Route headers nor a Contact header were present in the
response. The UAC MUST also remember whether the bottom-most entry in
the route set was constructed from a Contact header or not. This is
effectively a boolean value, which we refer to as CONTACT_SET. This
is needed in order for the UA to determine whether the bottom most
value can be updated from subsequent requests; if it was constructed
from a Contact, it can be updated.
The local sequence number sequence number MUST be set to the value of
the sequence number in the Cseq header of the request. The remote
sequence number MUST be empty (it is established when the UA sends a
request within the dialog). The call identifier component of the
dialog ID MUST be set to the value of the Call-ID in the request. The
local address component of the dialog ID MUST be set to the From
field in the request, and the remote address component of the dialog
ID MUST be set to the To field of the response. A UAC MUST be
prepared to receive a response without a tag in the To field, in
which case the tag is considered to effectively have a value of null.
This is to maintain backwards compatibility with RFC2543,
which did not mandate To tags.
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12.2 Requests within a Dialog
Once a dialog has been established between two UAs either of them MAY
initiate new transactions as needed within the dialog. However, a
dialog imposes some restrictions on the use of simultaneous
transactions.
A TU MUST NOT initiate a new regular transaction within a dialog
while a regular transaction is in progress (in either direction)
within that dialog.
OPEN ISSUE #113: Should we relax the constraint on non-
overlapping regular transactions?
A refresh request sent within a dialog is defined as a request that
can modify the route set of the dialog. For dialogs that have been
established with an INVITE, the only refresh request defined is re-
INVITE (see Section 14). Other extensions may define different
refresh requests for dialogs established in other ways.
Note that an ACK is NOT a refresh request.
12.2.1 UAC Behavior
12.2.1.1 Generating the Request
A request within a dialog is constructed by using many of the
components of the state stored as part of the dialog.
The To header field of the request MUST be set to the remote address,
and the From header field MUST be set to the local address (both
including tags, assuming the tags are not null).
The Call-ID of the request MUST be set to the Call-ID of the dialog.
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction. Therefore, if the local sequence number is not
empty, the value of the local sequence number MUST be incremented by
one, and this value MUST placed into the Cseq header. If the local
sequence number is empty, an initial value MUST be chosen using the
guidelines of Section 8.1.1.4. The method field in the Cseq header
MUST match the method of the request.
With a length of 32 bits, a client could generate, within a
single call, one request a second for about 136 years
before needing to wrap around. The initial value of the
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sequence number is chosen so that subsequent requests
within the same call will not wrap around. A non-zero
initial value allows clients to use a time-based initial
sequence number. A client could, for example, choose the 31
most significant bits of a 32-bit second clock as an
initial sequence number.
The Request-URI of requests is determined according to the following
rules:
The UAC takes the list of URI in the route set inserted into the
request URI of the request, including all URI parameters. Any URI
parameters not allowed in the request URI MUST then be stripped. Each
of the remaining URIs (if any) from the route set , including all URI
parameters, MUST be placed into a Route header field into the
request, in order.
A TU SHOULD follow the rules just mentioned to build the Request-URI
of the request, regardless of whether the UA uses an outbound proxy
server or not. However, in some instances, a UA may not be willing or
capable of sending the request to the top element in the route set is
not capable of DNS, and therefore may not be able to follow those
procedures. In these cases, the UA MAY send the request to a local
outbound server. In this case, it MUST NOT remove the top Route
header.
In dialogs created by an INVITE, if the UA is the caller,
it sets the Request-URI to the same value it used for the
initial request, and sends it to its local outbound server.
Bug#161: Which Request-URI does the callee use?
A UAC SHOULD include a Contact header in any refresh requests within
a dialog, and unless there is a need to change it, the URI SHOULD be
the same as used in previous requests within the dialog. As discussed
in Section 12.2.2, a Contact header in a refresh request updates the
route set. This allows a UA to provide a new contact address, should
its address change during the duration of the dialog.
However, requests that are not refresh requests do not affect the
route set for the dialog.
Once the request has been constructed, the address of the server is
computed and the request is sent, using the same procedures for
requests outside of a dialog (Section 8.1.1).
12.2.1.2 Processing the Responses
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The UAC will receives responses to the request from the transaction
layer.
The behavior of a UAC that receives a 3xx response for a request sent
within a dialog is the same as if the request would have been sent
outside a dialog. This behavior is described in Section 13.2.2.
Note however that when the UAC tries alternative locations
it still uses the route set for the dialog to build the
Route header of the request.
If a UAC has a route set for a dialog, and receives a 2xx response to
a refresh it sent, the Contact header field of the response is
examined. If not present, the route set remains unchanged. If the
response had a Contact header field, and the boolean variable
CONTACT_SET is false, the URL in the Contact header field in the
response is added to the bottom of the route set , and CONTACT_SET is
set to true. If the refresh request response had a Contact header
field, and CONTACT_SET is true, the URL in the Contact header field
of the response to the refresh request replaces the bottom value in
the route set is responded with a non-2xx final response the route
set remains unchanged as if no refresh request had been issued.
If the response for the a request within a dialog is a 481
(Call/Transaction Does Not Exist) or a 408 (Request Timeout) the UAC
SHOULD terminate the dialog.
For INVITE initiated dialogs terminating the dialog
consists of sending a BYE.
12.2.2 UAS behavior
The UAS will receive the request from the transaction layer. If the
request has a tag in the To header field, the UAS core computes the
dialog identifier corresponding to the request and compares it with
existing dialogs. If there is a match, this is a mid-dialog request.
In that case, the same processing rules for requests outside of a
dialog, discussed in Section 8.2, are applied by the UAS once the
request is received from the transaction layer.
Requests that do not change in any way the state of a dialog may be
received within a dialog (e.g., an OPTIONS request). They are
processed as if they had been received outside the dialog.
Requests within a dialog MAY contain Record-Route and Contact header
fields. However, requests that are not refresh requests do not update
the route set for the dialog. This specification only defines one
refresh request: re-INVITE (see Section 14).
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Special rules apply when updated Record-Route or Contact header
fields are received inside a refresh request. If a UAS has a route
set for a dialog, and receives a refresh for that dialog containing
Record-Route header fields, it MUST copy those header fields into any
2xx response to that request. If the boolean variable CONTACT_SET is
true, the Contact header field in the request (if present) replaces
the last entry in the route set the UAS MUST add the URL in the
Contact header field in the re- INVITE to the bottom of the route set
, and then set CONTACT_SET to true. If the request did not contain a
Contact header field, the route-set at the UAS remains unchanged.
If the remote sequence number is empty, it MUST be set to the value
of the sequence number in the Cseq header in the request. If the
remote sequence number was not empty, but the sequence number of the
request is lower than the remote sequence number, the request is out
of order and MUST be rejected with a 500 response. If the remote
sequence number was not empty, and the sequence number of the request
is greater than the remote sequence number, the request is in order.
It is possible for the CSeq header to be higher than the remote
sequence number by more than one. This is not an error condition, and
a UAS SHOULD be prepared to receive and process requests with CSeq
values more than one higher than the previous received request. The
UAS MUST then set the remote sequence number to the value of the
sequence number in the Cseq header in the request.
12.3 Termination of a Dialog
Dialogs can end in several different ways, depending on the method.
When a dialog is established with INVITE, it is terminated with a
BYE. No other means to terminate a dialog are described in this
specification, but extensions can define other ways.
13 Initiating a Session
13.1 Overview
When a user agent client desires to initiate a session (for example,
audio, video, or a game), it formulates an INVITE request. The INVITE
request asks a server to establish a session. This request is
forwarded by proxies, eventually arriving at one or more UAS which
can potentially accept the invitation. These UAS's will frequently
need to query the user about whether to accept the invitation. After
some time, those UAS can accept the invitation (meaning the session
is to be established) by sending a 2xx response. If the invitation is
not accepted, a 3xx,4xx,5xx or 6xx response is sent, depending on the
reason for the rejection. Before sending a final response, the UAS
can also send a provisional response (1xx) to advise the UAC of
progress in contacting the called user.
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After possibly receiving one or more provisional responses, the UA
will get one or more 2xx responses or one non-2xx final response.
Because of the protracted amount of time it can take to receive final
responses to INVITE, the reliability mechanisms for INVITE
transactions differ from those of other requests (like OPTIONS). Once
it receives a final response, the UAC needs send an ACK for every
final response it receives. The procedure for sending this ACK
depends on the type of response. For final responses between 300 and
699, the ACK processing is done in the transaction layer, and follows
one set of rules (See Section 17). For 2xx responses, the ACK is
generated by the UAC core.
A 2xx response to an INVITE establishes a session, and it also
creates a dialog between the UA that issued the INVITE and the UA
that generated the 2xx response. Therefore, when multiple 2xx
responses are received from different remote UAs (because the INVITE
forked), each 2xx establishes a different dialog. All these dialogs
are part of the same call.
This section provides details on the establishment of a session using
INVITE.
13.2 Caller Processing
13.2.1 Creating the Initial INVITE
Since the initial INVITE represents a request outside of a dialog,
its construction follows the procedures of Section 8.1.1. Additional
processing is required for the specific case of INVITE.
An Allow header field (Section 22.5) SHOULD be present in the
INVITE. It indicates what methods can be invoked within a dialog, on
the UA sending the INVITE, for the duration of the dialog. For
example, a UA capable of receiving INFO requests within a dialog [21]
SHOULD include an Allow header listing the INFO method.
A Supported header field (Section 22.35) SHOULD be present in the
INVITE. It enumerates all the extensions understood by the UAC.
An Accept (Section 22.1) header field MAY be present in the INVITE.
It indicates which content-types are acceptable to the UA, in both
the response received by it, and in any subsequent requests sent to
it within dialogs established by the INVITE. The Accept header is
especially useful for indicating support of various session
description formats.
The UA MAY add an Expires header field (Section 22.19) to limit the
validity of the invitation. If the time indicated in the Expires
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header field is reached and no final answer for the INVITE has been
received the UAC core SHOULD generate a CANCEL request for the
original INVITE.
A UAC MAY also find useful to add, among others, Subject (Section
22.34), Organization (Section 22.24) and User-Agent (Section 22.39)
header fields. They all contain useful information related to the
INVITE.
The UAC MAY choose to add a message body to the INVITE. Section
8.1.1.9 deals with how to construct the header fields- Content-Type
among others- needed to describe the message body.
There are special rules for message bodies that contain a session
description - their corresponding Content-Disposition is "session".
SIP uses an offer/answer model where one UA sends a session
description, called the offer, which contains a proposed description
of the session. The offer indicates the desired communications means
(audio, video, games), parameters of those means (such as codec
types) and addresses for receiving media from the offerer. The other
UA responds with another session description, called the answer,
which indicates which communications means are accepted, the
parameters which apply to those means, and addresses for receiving
media from the answerer. The offer/answer model can be mapped into
the INVITE transaction in two ways. The first, which is the most
intuitive, is that the INVITE contains the offer, the 2xx response
contains the answer, and no session description is provided in the
ACK. In this model, the UAC is the offerer, and the UAS is the
answerer. A second model is that the INVITE contains no session
description, the 2xx response contains the offer, and the ACK
contains the answer. In this model, the UAS is the offerer, and the
UAC is the answerer. The second model is useful for gateways from
H.323v1 to SIP, where the H.323 media characteristics are not known
until the call is established. This is also useful for sessions that
use third-party call control. As a result of these models, if the
INVITE contains a session description, the ACK MUST NOT contain one.
Conversely, if the caller chooses to omit the session description in
the INVITE, the ACK MUST contain one (if a 2xx response is received).
2xx responses to an INVITE MUST always contain a session description.
All user agents that support INVITE MUST support both models.
The Session Description Protocol (SDP) [6] MUST be supported by all
user agents as a means to describe sessions, and its usage for
construction offers and answers MUST follow the procedures defined in
[22].
Note that the restrictions of the offer-answer model (session
description only in the INVITE OR in the ACK, but not in both) just
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described only apply to bodies whose Content-Disposition header field
is "session". Therefore, it is possible that both the INVITE and the
ACK contain a body message (e.g., the INVITE carries a photo
(Content-Disposition: render) and the ACK a session description
(Content-Disposition: session) ).
If the Content-Disposition header field is missing, bodies
of Content-Type application/sdp imply the disposition
"session", while other content types imply "render".
Once the INVITE has been created, the UAC follows the procedures
defined for sending requests outside of a dialog (Section 8). This
results in the construction of a client transaction that will
ultimately send the request and deliver responses to the UAC.
If a UA A sends an INVITE request to B and receives an INVITE request
from B before it has received the response to its request from B, A
MAY return a 500 (Internal Server Error), which SHOULD include a
Retry- After header field specifying when the request should be
resubmitted.
13.2.2 Processing INVITE Responses
Once the INVITE has been passed to the INVITE client trasaction, the
UAC waits for responses for the INVITE. Responses are matched to
their corresponding INVITE because they have the same Call-ID, the
same From header field, the same To header field, excluding the tag,
and the same CSeq. Rules for comparisons of these headers are
described in Section 22.
Zero, one or multiple provisional responses may arrive before one or
more final responses are received. Provisional responses for an
INVITE request can create "early dialogs". If a provisional response
has a tag in the To field, and if the dialog ID of the response does
not match an existing dialog, one is constructed using the procedures
defined in Section 12.1.0.2.
The early dialog will only be needed if the UAC needs to send a
request to its peer within the dialog before the initial INVITE
transaction completes. Header fields present in a provisional
response are applicable for the duration of the early dialog (e.g.,
an Allow header field in a provisional response contains the methods
that can be used in the early dialog).
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A 3xx response may contain a Contact header field providing new
addresses where the callee might be reachable. Depending on the
status code of the 3xx response (see Section 23.3) the UAC MAY choose
to try those new addresses.
13.2.2.3 4xx, 5xx and 6xx responses
A single non-2xx final response may be received for the INVITE. 4xx,
5xx and 6xx responses may contain a Contact header field indicating
the location where additional information about the error can be
found.
All early dialogs are considered terminated upon reception of the
non-2xx final response.
After having received the non-2xx final response the UAC core
considers the INVITE transaction completed. The INVITE client
transaction handles generation of ACKs for the response (see Section
17).
Multiple 2xx responses may arrive at the UAC for a single INVITE
request due to a forking proxy. Each response is distinguished by the
tag parameter in the To header field, and each represents a distinct
dialog, with a distinct dialog identifier.
If the dialog identifier in the 2xx response matches the dialog
identifier of an existing dialog, the dialog MUST be transitioned to
the "established", and the route set for the dialog MUST be
recomputed based on the 2xx response using the procedures of Section
12.1.0.2. Otherwise, a new established dialog is constructed in the
same fashion.
The route set only is recomputed for backwards
compatibility. RFC 2543 did not mandate mirroring of
Record-Route headers in a 1xx, only 2xx. However, we cannot
update the entire state of the dialog, since mid-dialog
requests may have been sent within the early call leg,
modifying the sequence numbers, for example.
The UAC core MUST generate an ACK request for each 2xx received from
the transaction layer. The header fields of the ACK are constructed
in the same way as for any request sent within a dialog (see Section
12) with the exception of the CSeq. The sequence number of the CSeq
header field MUST be the same as the INVITE being acknowledged, but
the CSeq method MUST be ACK. If the INVITE did not contain an offer,
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the 2xx will contain one, and therefore the ACK MUST carry an answer
in its body.
Once the ACK has been constructed, the procedures of Section 24 are
used to send it. However, the request is passed to the transport
layer directly for transmission, rather than a client transaction.
This is because the UAC core handles retransmissions of the ACK, not
the transaction layer. The ACK MUST be passed to the client transport
every time a retransmission of the 2xx final response that triggered
the ACK arrives.
The UAC core considers the INVITE transaction completed 62*T1 seconds
after the reception of the first 2xx response. At this point all the
early dialogs that have not transitioned to established dialogs are
terminated. Once the INVITE transaction is considered completed by
the UAC core, no more new 2xx responses are expected to arrive.
If, after acknowledging any 2xx response to an INVITE, the caller
does not want to continue with that dialog, then the caller MUST
terminate the dialog by sending a BYE request as described in Section
15.
13.3 Callee Processing
13.3.1 Processing of the INVITE
The UAS core will receive INVITE requests from the transaction layer.
It first performs the request processing procedures of Section 8.2,
which are applied for both requests inside and outside of a dialog.
Assuming these processing states complete without generating a
response, the UAS core performs the additional processing steps:
1. If the request is an INVITE that contains an Expires header
field the UAS core inspects this header field. If the
INVITE has already expired a 487 response is generated.
2. If the request has no tag in the To the UAS core checks
ongoing transactions. If the To, From, Call-ID, CSeq
exactly match (including tags) those of any request
received previously, but the branch-ID in the topmost Via
is different from those received previously, the UAS core
SHOULD generate a 482 (Loop detected) response and pass it
to the server transaction.
The same request that was generated by the UAC has
arrived to the UAS more than once following different
paths. The UAS processes the request that was received
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first and responds with 482 (Loop detected) to the
rest of them.
If no match is found, the request does not belong to any
existing dialog. If the request is an INVITE the UAS core
follows the procedures described in this section.
3. If the request is a mid-dialog request, the method-
independent processing described in Section 12.2.2 is first
applied. It might also modify the session; Section 14
provides details.
4. If the request has a tag in the To header field but the
dialog identifier does not match any of the existing
dialogs, the UAS may have crashed and restarted, or may
have received a request for a different (possibly failed)
UAS. The UAS MAY either accept or reject the request.
Accepting the request provides robustness, so that dialogs
can persist even through crashes. UAs wishing to support
this capability must choose monotonically increasing CSeq
sequence numbers even across reboots. This is because
subsequent requests from the crashed-and-rebooted UA
towards the other UA need to have a CSeq sequence number
higher than previous requests in that direction.
Note also that the crashed-and-rebooted UA will have lost
any Route headers which would need to be inserted into a
subsequent request. Therefore, it is possible that the
requests may not be properly forwarded by proxies.
RTP media agents allowing restarts need to be robust
by accepting out-of-range timestamps and sequence
numbers.
If the UAS wishes to reject the request, because it does
not wish to recreate the dialog, it MUST respond to the
request with a 481 (Call/Transaction Does Not exist) status
code and pass that to the server transaction.
Processing from here forward assumes that the INVITE is outside of a
dialog, and is thus for the purposes of establishing a new session.
The INVITE may contain a session description, in which case the UAS
is being presented with an offer for that session. It is possible
that the user is already a participant in that session, even though
the INVITE is outside of a dialog. This can happen when a user is
invited to the same multicast conference by multiple other
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participants. If desired, the UAS MAY use identifiers within the
session description to detect this duplication. For example, SDP
contains a session id and version number in the origin (o) field. If
the user is already a member of the session and the session
parameters contained in the session description have not changed, the
UAS MAY silently accept the INVITE
The INVITE may not contain a session description at all, in which
case the UAS is being asked to participate in a session, but the UAC
has asked that the UAS provide the offer of the session.
The callee can indicate progress, accept, redirect, or reject the
invitation. In all of these cases, it formulates a response using the
procedures described in Section 8.2.7.
The UAS may not be able to answer the invitation immediately, and
might choose to indicate some kind of progress to the caller (for
example, an indication that a phone is ringing). This is accomplished
with a provisional response between 101 and 199. These provisional
responses establish early dialogs and therefore follow the procedures
of Section 12.1.0.1 in addition to those of Section 8.2.7. A UAS MAY
send as many provisional responses as it likes. Each of these MUST
indicate the same dialog ID. SIP, however, does not guarantee that
these provisional responses are reliably delivered to the UAC.
13.3.1.2 The INVITE is redirected
If the UAS decides to redirect the call, a 3xx response is sent. A
300 (Multiple Choices), 301 (Moved Permanently) or 302 (Moved
Temporarily) response SHOULD contain a Contact header field
containing URIs of new addresses to be tried. The response is passed
to the INVITE server transaction, which will deal with its
retransmissions.
13.3.1.3 The INVITE is rejected
A common scenario occurs when the callee is currently not willing or
able to take additional calls at this end system. A 486 (Busy Here)
SHOULD be returned in such scenario. If the UAS knows that no other
end system will be able to accept this call a 600 (Busy Everywhere)
response SHOULD be sent instead. However, it is unlikely that a UAS
will be able to know this in general, and thus this response will not
usually be used. The response is passed to the INVITE server
transaction, which will deal with its retransmissions.
13.3.1.4 The INVITE is accepted
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The UAS core generates a 2xx response. This response establishes a
dialog, and therefore follows the procedures of Section 12.1.0.1 in
addition to those of Section 8.2.7.
A 2xx response to an INVITE SHOULD contain the Allow header field and
the Supported header field, and MAY contain the Accept header field.
Including these header fields allows the UAC to determine the
features and extensions supported by the UAS for the duration of the
call, without probing.
If the INVITE request contained an offer, the 2xx MUST contain an
answer. If the INVITE did not contain an offer, the 2xx MUST contain
an offer.
Once the response has been constructed it is passed to the INVITE
server transaction. Note, however, that the INVITE server transaction
does not retransmit 2xx responses to an INVITE. Therefore, it is
necessary to pass periodically the response to the server transaction
until the ACK arrives. The 2xx response is resubmitted to the server
transaction with an interval that starts at T1 seconds and doubles
for each retransmission until it reaches T2 seconds (T1 and T2 are
defined in Section 17). Response retransmissions cease when an ACK
request is received with the same dialog ID as the response. This is
independent of whatever transport protocols are used to send the
response.
Since 2xx is retransmitted end-to-end, there may be hops
between UAS and UAC which are UDP. To ensure reliable
delivery across these hops, the response is retransmitted
periodically even if the transport at the UAS is reliable.
If the server retransmits the 2xx response for 64*T1 seconds without
receiving an ACK, it considers the dialog completed, the session
terminated, and therefore it SHOULD send a BYE.
14 Modifying an Existing Session
A successful INVITE request (see Section 13) establishes both a
dialog between two user agents and a session (using the offer/answer
model). Section 12 explains how to modify an existing dialog using a
refresh request (e.g., changing the route set of the dialog). This
section describes how to modify the actual session. This modification
can involve changing addresses or ports, adding a media stream,
deleting a media stream, and so on. This is accomplished by sending a
new INVITE request within the same dialog that established the
session. An INVITE request sent within an existing dialog is known as
a re-INVITE.
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Note that a single re-INVITE can modify at the same time
the dialog and the parameters of the session.
Either the caller or callee can modify an existing session.
14.1 UAC Behavior
The same offer-answer model that applies to session descriptions in
INVITEs (Section 13.2.1) applies to re-INVITEs. As a result, a UAC
that wants to add a media stream, for example, will create a new
offer that contains this media stream, and send that in an INVITE
request to its peer. It is important to note that the full
description of the session, not just the change, is sent. This
maintains the idempotency of SIP, supports stateless session
processing in various elements, and supports failover and recovery
capabilities. Of course, a UAC MAY send a re-INVITE with no session
description, in which case the response to the re-INVITE will contain
the offer.
If the session description format has the capability for version
numbers, the offerer SHOULD indicate that the version of the session
description has changed.
The To, From, Call-ID, CSeq, and Request-URI of a re-INVITE are set
following the same rules as for regular requests within an existing
dialog, described in Section 12.
Note that, as opposed to initial INVITEs (see Section 13), re-
INVITEs contain tags in the To header field and are sent using the
route set for the dialog. Therefore, a single final (2xx or non-2xx)
response is received for re-INVITEs.
Note that a UAC MUST NOT initiate a new INVITE transaction within a
dialog while another transaction (INVITE or non-INVITE) is in
progress. However, a UA MAY initiate a regular transaction within an
early dialog - while an INVITE transaction is in progress.
If a re-INVITE is responded with a non-2xx final response the session
parameters MUST remain unchanged, as if no re-INVITE had been issued.
The rules for transmitting a re-INVITE and for generating an ACK for
a 2xx response to re-INVITE are the same as for an INVITE (Section
13.2.1).
14.2 UAS Behavior
Section 13.3.1 describes the steps to follow in order to distinguish
incoming re-INVITEs from incoming initial INVITEs. This Section
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describes the procedures to follow upon reception of a re-INVITE for
an existing dialog.
A UAS that receives a second INVITE before it sent the final response
to a first INVITE with a lower CSeq sequence number on the same
dialog MUST return a 500 response to the second INVITE and MUST
include a Retry-After header field with a randomly chosen value of
between 0 and 10 seconds. Similarly, a UAS the receives an INVITE on
a dialog while an INVITE it had sent on that dialog is in progress
MUST return a 500 response to the received INVITE and MUST include a
Retry-After header field with a randomly chosen value of between 0
and 10 seconds.
If a user agent receives a re-INVITE for an existing dialog it MUST
check any version identifiers in the session description or, if there
are no version identifiers, the content of the session description to
see if it has changed. If the session description has changed, the
user agent server MUST adjust the session parameters accordingly,
possibly after asking the user for confirmation.
Versioning of the session description can be used to
accommodate the capabilities of new arrivals to a
conference, add or delete media or change from a unicast to
a multicast conference.
If a UAS generates a 2xx response and never receives an ACK, it
SHOULD generate a re-INVITE itself with an offer equal to the last
session description sent to the peer. The purpose of this is to
ensure that both caller and callee have a consistent view of the
session parameters.
A UAS providing an offer in a 2xx (because the INVITE did not contain
an offer) MUST offer the same session description as last provided to
the peer, with the exception of being able to change the IP
address/port if so desired.
Under error conditions (e.g., the UAS has crashed and
restarted) the session description in the 2xx response for
an empty re-INVITE may be different than the one in use at
that moment. If the new session description is not
acceptable for the UAC it SHOULD then send a BYE (after
ACKing the 2xx response).
15 Terminating a Session
Terminating a session is done either with the BYE request, or the
CANCEL request, depending on the state of the dialog. Either caller
or callee can terminate, and may do so for any reason. Sections 13
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and 12 document some cases where call termination is normative
behavior. As a general rule, if a UA decides that the session is to
be terminated, it MUST follow the procedures here to initiate
signaling action to convey that.
Note that both the session and the dialog between both user
agents will be terminated.
When a UAC sends an INVITE request to create a session, if a 1xx
response with a tag in the To field is received, an early dialog is
created. When a 2xx response is received, the dialog becomes
established. For either state of the dialog, if the UAC desires to
terminate the session, the UAC SHOULD follow the procedures described
in Section 15.1.1 to terminate the session. If the callee for a new
session wishes to terminate the dialog, it uses the procedures of
Section 15.1.1, but MUST NOT do so until it has receive an ACK or
until the server transaction times out.
This does not mean a user can't hang up right away; it just
means that the software in their phone needs to maintain
state for a short while in order to properly clean up.
OPEN ISSUE #202: Is this the right solution.
If the UAC desires to end the session before any type of dialog has
been created, it SHOULD send a CANCEL for the INVITE request that
requested establishment of the session that is to be terminated. The
UAC constructs and sends the CANCEL following the procedures
described in Section 9. This CANCEL will normally result in a 487
response to be returned to the INVITE, indicating successful
cancellation. However, it is possible that the CANCEL and a 2xx
response to the INVITE "pass on the wire". In this case, the UAC will
receive a 2xx to the INVITE. It SHOULD then terminate the call by
following the procedures described in Section 15.1.1.
15.1 Terminating a Dialog with a BYE
15.1.1 UAC Behavior
A user agent client uses BYE request, sent within a dialog, to
indicate to the server that it wishes to terminate the session. This
will also terminate the dialog. A BYE request MAY be issued by either
caller or callee. A BYE request SHOULD NOT be sent before the
creation of a dialog (either early or established). In that case the
UAC SHOULD follow the procedures described in Section 9 instead.
Proxies ensure that a CANCEL request is routed in the same
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way as the INVITE was. However, a proxy performing load
balancing may route a BYE without a Route header field in a
different way than the INVITE, since both requests have
different CSeq sequence numbers.
The To, From, Call-ID, CSeq, and Request-URI of a BYE are set
following the same rules as for regular requests sent within a
dialog, described in Section 12.
Once the BYE is constructed, it creates a new non-INVITE client
transaction, and passes it the BYE request. The user agent SHOULD
stop sending media as soon as the BYE request is passed to the client
transaction.
15.1.2 UAS Behavior
A UAS core receiving a BYE request checks to see if it matches an
existing dialog. If the BYE does not match an existing dialog, the
UAS core SHOULD generate a 481 response and pass that to the server
transaction.
A UAS core receiving a BYE request for an existing dialog MUST follow
the procedures of Section 12.2.2 to process the request. Once done,
the UAS MUST cease transmitting media streams for the session being
terminated. The UAS core MUST generate a 2xx response to the BYE, and
MUST pass that to the server transaction for transmission.
The UAS MUST still respond to any pending requests received for that
dialog, (which can only be an INVITE). It is RECOMMENDED that a 487
(Request Terminated) response is generated to those pending requests.
16 Proxy Behavior
16.1 Overview
SIP proxies are elements that route SIP requests to user agent
servers and SIP responses to user agent clients. A request may
traverse several proxies on its way to a UAS. Each will make routing
decisions, modifying the request before forwarding it to the next
element. Responses will route through the same set of proxies
traversed by the request in the reverse order.
It is important to note that being a proxy is a logical role for a
SIP element. When a request arrives, an element that can play the
role of a proxy must first decide if it needs to respond to the
request on its own. For instance, the request could be malformed or
the element may need credentials from the client before acting as a
proxy. The element MAY respond with any appropriate error code.
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When responding directly to a request, the element is playing the
role of a UAS and MUST behave as described in Section 8.2.
A proxy can operate in either a stateful or stateless mode for each
new request.
When stateless, a proxy acts as a simple forwarding element. It
forwards each request downstream to a single element determined by
making a routing decision based on the request. It simply forwards
every response it receives upstream. A stateless proxy discards
information about a message once it has been forwarded.
On the other hand, a stateful proxy remembers information
(specifically, transaction state) about each incoming request and any
requests it sends as a result of processing the incoming request. It
uses this information to affect the processing of future messages
associated with that request. A stateful proxy MAY chose to "fork" a
request, routing it to multiple destinations. Any request that is
forwarded to more than one location MUST be handled statefully. Any
request processed using TCP (or any other mechanism that is
inherently stateful), MUST be handled statefully.
Much of the processing involved when acting statelessly or statefully
for a request is identical. The next several subsections are written
from the point of view of a stateful proxy. The last section calls
out those places where a stateless proxy behaves differently.
16.2 Stateful Proxy
When stateful, a proxy is purely a SIP transaction processing engine.
Its behavior is modeled here in terms of the Server and Client
Transactions defined in Section 17. A stateful proxy has a server
transaction associated with one or more client transactions by a
higher layer proxy processing component (see figure 3), known as a
proxy core. An incoming request is processed by a server transaction.
Requests from the server transaction are passed to a proxy core. The
proxy core determines where to route the request, choosing one or
more next-hop locations. An outgoing request for each next-hop
location is processed by its own associated client transaction. The
proxy core collects the responses from the client transactions and
uses them to send responses to the server transaction.
A stateful proxy creates a new server transaction for each new
request received. Any retransmissions of the request will then be
handled by that server transaction per Section 17.
Note that this is a model of proxy behavior, not of software. An
implementation is free to take any approach that replicates the
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external behavior this model defines.
+--+
| C|
| l|
| i|
| e|
| n|
+-t+
+----------------------+ +---+
| | | C |
+--+ | | | l |
| | | | | i |
| | | Proxy | | e |
| S| | "higher" layer | | n |
| e| | | +-t-+
| r| | |
| v| | |
| e| | |
| r| | | +---+
| | | | | |
| | | | | C |
| | | | | l |
+--+ +----------------------+ | i |
| e |
| n |
| t |
| |
+---+
Figure 3: Stateful Proxy Model
For all new requests, including any with unknown methods, an element
intending to proxy the request MUST:
1. Validate the request (Section 16.3)
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2. Make a routing decision (Section 16.4)
3. Forward the request to each chosen destination (Section
16.5)
4. Process all responses (Section 16.6)
16.3 Request Validation
Before an element can proxy a request, it MUST verify the message's
validity. A valid message must pass the following checks:
1. Reasonable Syntax
2. Max-Forwards
3. Loop Detection
4. Proxy-Require
5. Proxy-Authorization
If any of these checks fail, the element MUST behave as a user agent
server (see Section 8.2) and respond with an error code.
1. Reasonable Syntax check
The request MUST be well-formed enough to be handled with a
server transaction. Any components involved in the
remainder of these Request Validation steps or the Request
Processing section MUST be well-formed. Any other
components, well-formed or not, SHOULD be ignored. For
instance, an element SHOULD NOT reject a request because of
a malformed Date header field.
This protocol is designed to be extended. Future extensions
may define new methods and header fields at any time. An
element MUST NOT refuse to proxy a request because it
contains a method or header field it does not know about.
2. Max-Forwards check
The Max-Forwards header (Section 22.22) is used to limit
the number of elements a SIP request can traverse.
If the request does not contain a Max-Forwards header
field, this check is passed.
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If the request contains a Max-Forwards header field with a
field value greater than zero, the check is passed.
If the request contains a Max-Forwards header field with a
field value of zero (0), the element MUST NOT forward the
request. If the request was for OPTIONS, the element MAY
act as the final recipient and respond per Section 11.
Otherwise, the element MUST return a 483 (Too many hops)
response.
3. Loop Detection check
An element MUST check for forwarding loops before
forwarding a request. If the request contains a Via header
field value with A sent-by value that equals a value placed
into previous requests by the proxy, the request has been
forwarded by this element before. The request has either
looped or is legitimately spiraling through the element.
To determine if the request has looped, the element MUST
perform the branch parameter calculation described in
Section 3 on this message and compare it to the parameter
received in that Via field value. If the parameters match,
the request has looped. If they differ, the request is
spiraling, and processing continues. If a loop is detected,
the element MUST return a 482 (Loop Detected) response.
An element MUST NOT forward a request to a multicast group
which already appears in any of the Via headers.
4. Proxy-Require check
Future extensions to this protocol may introduce features
that require special handling by proxies. Endpoints will
include a Proxy-Require header in requests that use these
features, telling the proxy it should not process the
request unless the feature is understood.
If the request contains a Proxy-Require header (Section
22.28) with one or more option-tags this element does not
understand, the element MUST return a 420 (Bad Extension)
response. The response MUST include an Unsupported
(Section 22.38) header field listing those option-tags the
element did not understand.
5. Proxy-Authorization check
If an element requires credentials before forwarding a
request, the request MUST be inspected as described in
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Section 20.2.3. That section also defines what the element
must do if the inspection fails.
16.4 Making a Routing Decision
At this point, the proxy must decide where to forward the request.
This can be modeled as computing a set of destinations for the
request. This set will either be predetermined by the contents of
the request or will be obtained from an abstract location service.
Each destination is represented as a URI and an optional IP address,
port and transport. This combination is referred to as a "next-hop
location".
First, the proxy core checks the received request for Route headers.
If any Route header fields are present in the request, the element
MUST use the URL (including all of its parameters) from the topmost
Route header field as only next hop URI in the destination set, with
no IP address, port and transport set for that next hop. The
destination set is complete, containing only this URL, and the proxy
MUST proceed to the Request Processing of Section 16.5.
The Route mechanism is used to control the path a request takes
through SIP elements, much like strict IP source routing. The UAC
will insert Route header fields (see Section 12), usually based on
information provided by proxies through Record-Route header fields
(see Section 6).
Assuming there were no Route headers in the received request, the
proxy checks the Request-URI of the received request. If it has an
maddr parameter, and that parameter does not indicate an interface
the proxy is listening on, the Request-URI MUST be placed into the
destination set as the only next hop URI, with no IP address, port
and transport set for that next hop, and the proxy MUST proceed to
Section 16.5. If the maddr parameter was present, but did indicate an
interface the proxy is listening on, the proxy MUST strip the maddr
and continue processing as if no maddr were present.
OPEN ISSUE #213: Do we strip just the maddr, or the port
and transport as well?
OPEN ISSUE #218: Are we really sure this ordering of
precedence of Route, maddr, and domain is correct?? It is
not yet clear. This needs resolution asap finally, since it
affects things like loose source routing, outbound proxy
processing at a UA, and so on.
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If the domain of the Request-URI indicates a domain this element is
not responsible for, it SHOULD set the next hop URI to the Request-
URI, and leave the IP address, port and transport of the next hop
empty. That next hops MUST be placed into the destination set as the
only next hop, and the element MUST proceed to the task of Request
Processing (Section 16.5.
There are many circumstances in which a proxy might receive
a request for a domain it is not responsible for. A
firewall proxy handling outgoing calls (the way HTTP
proxies handle outgoing requests) is an example of where
this is likely to occur.
If the destination set for the request has not been predetermined as
described above, this implies that the element is responsible for the
domain in the Request-URI, and the element MAY use whatever mechanism
it desires to determine where to send the request. Any of these
mechanisms can be modeled as accessing an abstract Location Service.
This may consist of obtaining information from a location service
created by a SIP Registrar, reading a database, consulting a presence
server, utilizing other protocols, or simply performing an
algorithmic substitution on the Request-URI. The output of these
mechanisms is used to construct the destination set.
Any information in or about the request or the current environment of
the element MAY be used in the construction of the destination set.
For instance, different sets may be constructed depending contents or
presence of header fields and bodies, the time of day of the
request's arrival, the interface on which the request arrived,
failure of previous requests, or even the element's current level of
utilization.
As potential destinations are located through these services, their
next hops are added to the destination set. Next-hop locations may
only be placed in the destination set once. If a next-hop location is
already present in the set (based on the definition of equality for
the URI type and equality of the optional parameters), it MUST NOT be
added again.
A proxy MAY continue to add destinations to the set after beginning
Request Processing. It MAY use any information obtained during that
processing to determine new locations. For instance, a proxy may
choose to incorporate contacts obtained in a redirect response (3xx
class) into the destination set. If a proxy uses a dynamic source of
information while building the destination set (for instance, if it
consults a SIP Registrar), it SHOULD monitor that source for the
duration of processing the request. New locations SHOULD be added to
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the destination set as they become available. As above, any given URI
MUST NOT be added to the set more than once.
Allowing a URI to be added to the set only once reduces
unnecessary network traffic, and in the case of
incorporating contacts from redirect requests prevents
infinite recursion.
An example trivial location service is achieved by configuring an
element with a default outbound destination. All requests are
forwarded to this location. The Request-URI of the request is placed
in the destination set with the optional next-hop IP address, port
and transport parameters set to the default outbound destination. The
destination set is complete, containing only this URI, and the
element proceeds to the task of Request Processing.
If the Request-URI indicates a resource at this proxy that does not
exist, the proxy MUST return a 404 (Not Found) response.
If the destination set remains empty after applying all of the above,
the proxy MUST return an error response, which SHOULD be the 480
(Temporarily Unavailable) response.
16.5 Request Processing
As soon as the destination set is non-empty, a proxy MAY begin
forwarding the request. A stateful proxy MAY process the set in any
order. It MAY process multiple destinations serially, allowing each
client transaction to complete before starting the next. It MAY start
client transactions with every destination in parallel. It also MAY
arbitrarily divide the set into groups, processing the groups
serially and processing the destinations in each group in parallel.
A common ordering mechanism is to use the qvalue parameter of
destinations obtained from Contact header fields (see Section 22.10).
Destinations are processed from highest qvalue to lowest.
Destinations with equal qvalues may be processed in parallel.
A stateful proxy must have a mechanism to maintain the destination
set as responses are received and associate the responses to each
forwarded request with the original request. For the purposes of this
model, this mechanism is a "response context" created by the proxy
layer before forwarding the first request.
For each destination, the proxy forwards the request following these
steps:
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1. Make a copy of the received request
2. Update the Request-URI
3. Add a Via header field value
4. Update the Max-Forwards field if present
5. Update the Route header field if present
6. Optionally add a Record-route header field value
7. Optionally add additional headers
8. send the new request
Each of these steps is detailed below:
1. Copy request
The proxy starts with a copy of the received request. The
copy MUST initially contain all of the header fields from
the received request. Only those fields detailed in the
processing described below may be removed. The copy SHOULD
maintain the ordering of the header fields as in the
received request. The proxy MUST NOT reorder field values
with a common field name (See Section 7.3.1).
An actual implementation need not perform a copy; the
primary requirement is that the processing of each
next hop begin with the same request.
2. Request-URI
The Request-URI in the copy's start line MUST be replaced
with the URI for this destination. If the URI contains any
parameters not allowed in a Request-URI, they MUST be
removed.
This is the essence of a proxy's role. This is the
mechanism through which a proxy routes a request toward its
destination.
3. Via
The proxy MUST insert a Via header field into the copy
before the existing Via header fields. The Via header
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maddr, ttl, and sent-by components will be set when the
request is processed by the transport layer (Section 19).
The Via headers ensure that responses will follow the same
set of elements that the request traversed.
The proxy MUST include a "branch" parameter (Section 22.40)
in the Via header. When the path of a request through one
or more forking proxies is graphed, the result is a tree.
The branch parameter identifies the "branch" each request
was forwarded on. The branch parameter value MUST be
unique for each client transaction to which the request is
forwarded. The precise format of the branch. token is
implementation-defined. In order to be able to both detect
loops and associate responses with the corresponding
request, the parameter SHOULD consist of two parts
separable by the implementation. The first part is used to
detect loops and distinguish loops from spirals. The second
is used to match responses to requests.
Loop detection is performed by verifying that those fields
having an impact on the routing decision have not changed.
The value placed in the this part of the branch parameter
SHOULD reflect all of those fields (which include any
Proxy-Require and Proxy-Authorization headers). This is to
ensure that if the request is routed back to the proxy, and
one of those fields changes, it is treated as a spiral and
not a loop (Section 3). A common way to create this value
is to compute a cryptographic hash of the To, From, Call-ID
header fields, the Request-URI of the request received
(before translation) and the sequence number from the CSeq
header field, in addition to any Proxy-Require and Proxy-
Authorization fields that may be present. The algorithm
used to compute the hash is implementation-dependent, but
MD5 [23], expressed in hexadecimal, is a reasonable choice.
(Note that base64 is not permissible for a token.)
In order to correctly match responses to requests (Section
17.1.3), the value SHOULD also contain a part that is a
globally unique function of of the branch on which this
request will be forwarded. One example is a hash of a
sequence number, local IP address and request-URI of the
request
For example: 7a83e5750418bce23d5106b4c06cc632.1
The "branch" parameter MUST depend on all information
used for routing decisions, including the incoming
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request-URI and any header values affecting the
routing choices. This is necessary to distinguish
looped requests from requests whose routing parameters
have changed before returning to this server.
Note that the request method MUST NOT be included in the
calculation of the branch parameter. In particular, CANCEL
and ACK requests MUST have the same branch value as the
corresponding request they cancel or acknowledge. The
branch parameter is used in correlating those requests at
server handling them (see Section 17.2.3 and 9.2).
4. Max-Forwards
If the copy contains a Max-Forwards header field, the proxy
must decrement its value by one (1).
5. Route
If the copy contains a Route header field, the proxy must
remove the first (topmost) value. Note that this value was
placed in the destination set and then into the Request-URI
of this copy in previous steps.
6. Record-Route
If this proxy wishes to request to remain on the path of
future requests in this dialog, it MUST insert a Record-
Route header value (Section refsec:record-route) into the
copy before any existing Record-Route header values. See
Section 12 for details on whether this request will be
honored. Each proxy in the path of a request makes this
request independently the presence of a Record-Route header
does not obligate this proxy to add a value.
If the request is honored, the information the proxy places
in the Record-Route header value will be used at the
endpoints to construct Route headers. As shown in the
processing steps above, Route headers determine forwarding
destinations much like strict IP source routing.
The URL placed in the Record-Route header value MUST be a
SIP URL. This URL MAY be different for each destination the
request is forwarded to. The URL SHOULD NOT contain the
transport parameter unless the proxy has knowledge (such as
in a private network) that the next downstream element that
will be in the path of subsequent requests supports that
transport.
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The URL this proxy provides will be used by some other
element to make a routing decision. This proxy, in
general, has no way to know what the capabilities of
that element are, so it must restrict itself to the
mandatory elements of a SIP implementation: SIP URLs
and UDP transports.
The URL placed in the Record-Route header value MUST
resolve to this element when the server location procedures
of Section 24 are applied to it. This ensures subsequent
requests are routed back to this element.
The URL placed in the Record-Route header value SHOULD be
such that if a subsequent request is received with this URL
in the Request-URI, the proxy's normal request processing
will cause it to be forwarded to one of the previous
elements, including the originating client, traversed by
the original request. This improves robustness, ensuring
that the Request-URI contains enough information to forward
subsequent requests to a reasonable destination even in the
absence of Route headers.
The URL placed in the Record-Route header value MUST vary
with the Request-URI in the received request. A request
may legitimately pass through this proxy more than once on
the way to its final destination (this is called a
spiraling request). The Request-URI will be different each
time the request passes through. If this proxy places the
same URL in the Record-Route header field each time,
subsequent requests will be rejected as looped requests. It
is insufficient to simply copy the Request-URI from each
request into the Record-Route header. Some modification,
such as adding an maddr parameter, is necessary.
URLs satisfying the above paragraphs can be constructed in
many ways. One way is to use a URL that is nearly the same
as the Contact header in the initial request (if present,
else the From field), but with the maddr and port set to
resolve to the proxy, and with a transaction identifier
added to the user part of the request-URI (in order to meet
the requirement that the URL in the Record-Route be
different for each distinct Request-URI). A call stateful
proxy could use a URL of the form sip:proxy.example.com and
use information from the stored call state to meet the
requirements.
The proxy MAY include Record-Route header parameters in the
value it provides. These will be returned in some responses
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to the request (200 responses to INVITE for example) and
may be useful for pushing state into the message.
The Record-Route process is designed to work for any SIP
request that initiates a dialog. The only such request in
this specification is INVITE. Extensions to the protocol
MAY define others, and the mechanisms described here will
apply. The request that initiates a dialog and all
refreshes (re-INVITE for example) MUST have Record-Route
header values added to them if the proxy wishes to remain
in the request path. This means a proxy will often need to
record-route requests that contain Route headers. Section
12 describes how this will affect a dialog.
Including Record-Route even when Route headers already
exist in a request improves robustness in the presence
of a preloaded Route header field and recovery from
endpoint failure.
If a proxy needs to be in the path of any type of dialog
(such as one straddling a firewall), it SHOULD add a
Record-Route header value to every request with a method it
doesn't understand.
Generally, the choice about whether to record-route or not
is a tradeoff of features vs. performance. Faster request
processing and higher scalability is achieved when proxies
do not record route. However, provision of certain services
may require a proxy to observe all messages in a dialog. It
is RECOMMENDED that proxies do not automatically record
route. They should do so only if specifically required.
7. Adding Additional Headers
The proxy MAY add any other appropriate headers to the copy
at this point.
8. Forward Request
A stateful proxy creates a new client transaction for this
request as described in Section 17.1. If the next-hop
location used in building this request contains the
optional addressing parameters, the transaction is
instructed to send the request based on those parameters.
Otherwise, the proxy uses the procedures of Section 24 to
compute an ordered set of addresses from the Request-URI,
and as described there, attempts to contact the first one
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by instructing the client transaction to send the request
there. If this fails, the stateful proxy continues down the
list. Each attempt is a new client transaction, and
therefore represents a new branch, so that the processing
described above for each branch would need to be repeated.
This results in a requirement to use a different branch ID
parameter for each attempt.
16.6 Response Processing
When a response is received by an element, it first tries to locate a
client transaction (Section 17.1.3) matching the response. If none is
found, the element MUST process the response (even if it is an
informational response) as a stateless proxy (described below). If a
match is found, the response is handed to the client transaction.
Forwarding responses for which a client transaction (or
more generally any knowledge of having sent an associated
request) is not found improves robustness. In particular,
it ensures that "late" 2xx class responses to INVITE
requests are forwarded properly.
As client transactions pass responses to the proxy layer, the
following processing MUST take place:
1. Find the appropriate response context
2. Remove the topmost Via
3. Add the response to the response context
4. Check to see if this response should be forwarded
The following processing MUST be performed on each response that is
forwarded. Note that more than one response to each request will
likely be forwarded - each provisional and one final at the least.
1. Aggregate authorization header fields if necessary
2. Forward the response
3. Generate any necessary CANCEL requests
If no final response has been forwarded after every client
transaction associated with the response context has been terminated,
the proxy must choose and forward the "best" response from those it
has seen so far.
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Each of the above steps are detailed below:
1. Find Context
The proxy locates the "response context" it created before
forwarding the original request using the key described in
Section 16.5. The remaining processing steps take place in
this context.
2. Via
The proxy removes the topmost Via field value from the
response. The address in this value necessarily matches
the proxy since the response matched a client transaction
above. The branch parameter from this value can be used to
determine which branch the response corresponds to.
If no Via field values remain in the response, the response
was meant for this element and MUST NOT be forwarded. The
remainder of the processing described in this section is
not performed on this message. This will happen, for
instance, when the element generates CANCEL requests as
described in Section sec:proxy-response-processing-cancel.
3. Add response to context
Final responses received are stored in the response context
until a final response is generated on the server
transaction associated with this context. The response may
a candidate for the best final response to be returned on
that server transaction. Information from this response may
be needed in forming the best response even if this
response is not chosen.
If the proxy chooses to recurse on a 3xx class response, it
MUST NOT add the response to the response context
4. Check response for forwarding
Until a final response has been sent on the server
transaction, the following responses MUST be forwarded
immediately:
- Any provisional response other than 100 Trying
- Any 2xx response
If a 6xx response is received, it is not immediately
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forwarded, but the stateful proxy SHOULD cancel all pending
transactions as described in Section 9.
This is a change from RFC2543, which mandated that the
6xx be forwarded immediately. The problem with this is
that it is possible for a 2xx to arrive on another
branch, in which case the proxy would have to forward
that in the case of an INVITE transaction. The result
is that the UAC could receive a 6xx followed by a 2xx,
which should never be allowed to happen. So, instead,
upon receiving a 6xx, a proxy will CANCEL, which will
generally result in 487s to all outstanding client
transactions, and then at that point the 6xx is
forwarded upstream.
After a final response has been sent on the server
transaction, the following responses MUST be forwarded
immediately:
- Any 2xx class response to an INVITE request
A stateful proxy MUST NOT immediately forward any other
responses. In particular, a stateful proxy MUST NOT forward
any 100 Trying response. Those responses that are
candidates for forwarding later as the "best" response have
been gathered as described in step "Add Response to
Context".
Any response chosen for immediate forwarding MUST be
processed as described in steps "Aggregate authorization
headers" through "Record-Route".
5. Choosing the best response
A stateful proxy MUST send a final response to a response
context's server transaction if no final responses have
been immediately forwarded by the above rules and all
client transactions in this response context have been
terminated.
The stateful proxy MUST choose the "best" final response
among those received and stored in the response context.
If there are no final responses in the context, the proxy
MUST send a 408 (Request Timeout) response to the server
transaction.
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Otherwise, the proxy MUST forward one of the responses from
the lowest response class stored in the response context.
The proxy MAY select any response within that lowest class.
The proxy SHOULD give preference to responses that provide
information affecting resubmission of this request, such as
401, 407, 415, 420, and 484.
A proxy which receives a 503 response SHOULD NOT forward it
upstream unless it can determine that any subsequent
requests it might proxy will also generate a 503. In other
words, forwarding a 503 means that the proxy knows it
cannot service any requests, not just the one for the
Request-URI in the request which generated the 503.
The forwarded response MUST be processed as described in
steps "Aggregate authorization headers" through "Record-
Route".
For example, if a proxy forwarded a request to 4 locations,
and received 503, 407, 501, and 404 responses, it may
choose to forward the 407 response.
The tag in the To header field serves to distinguish
responses at the UAC. If the forwarded response did not
have one, it MUST NOT be inserted into the response by the
proxy.
6. Aggregate authorization headers
If the selected response is a 401 or 407, the proxy MUST
collect any WWW-Authenticate and Proxy-Authenticate header
fields from all other 401 and 407 responses received so for
in this response context and add them to this response
before forwarding.
This is necessary because any or all of the destinations
the request was forwarded to may have requested
credentials. The client must receive all of those
challenges and supply credentials for each of them when it
retries the request. Motivation for this behavior is
provided in Section 20.
7. Record-Route
If the selected response contains a Record-Route header
field value originally provided by this proxy, the proxy
MAY chose to rewrite the value before forwarding the
response. This allows the proxy to provide different URLs
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for itself to the next upstream and downstream elements. A
proxy may choose to use this mechanism for any reason. For
instance, it is useful for multi-homed hosts.
The new URL provided by the proxy MUST satisfy the same
constraints on URLs placed in Record-Route header fields in
requests (see Section 6) with the following modifications:
The URL SHOULD NOT contain the transport parameter unless
the proxy has knowledge that the next upstream (as opposed
to downstream) element that will be in the path of
subsequent requests supports that transport.
The URL placed in the Record-Route header value SHOULD be
such that if a subsequent request is received with this URL
in the Request-URI, the proxy's normal request processing
will cause it to be forwarded to the same next-hop element
(as opposed to some previous element) as the originally
forwarded request.
When a proxy does decide to modify the Record-Route header
in the response, one of the operations it must perform is
to locate the Record-Route that it had inserted. If the
request spiraled, and the proxy inserted a Record-Route in
each iteration of the spiral, locating the correct header
in the response (which must be the proper iteration in the
reverse direction) is tricky. Note that the rules above
dictate that a proxy insert a different URI into the
Record-Route for each distinct Request-URI received. The
two issues can be solved jointly. A RECOMMENDED mechanism
is for the proxy to append a piece of data to the user
portion of the URL. This piece of data is a hash of the
transaction key for the incoming request, concatenated with
a unique identifier for the proxy instance. Since the
transaction key includes the Request-URI, this key will be
unique for each distinct Request-URI. When the response
arrives, the proxy modifies the first Record-Route whose
identifier matches the proxy instance. The modification
results in a URI without this piece of data appended to the
user portion of the URI. Upon the next iteration, the same
algorithm (find the topmost Record-Route header with the
parameter) will correctly extract the next Record-Route
header inserted by that proxy.
8. Forward response
After performing the processing described in steps
"Aggregate authorization headers" through "Record-Route",
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the proxy may perform any feature specific manipulations on
the selected response. Unless otherwise specified, the
proxy MUST NOT remove the message body or any header values
other than the Via header value discussed in Section
refsec:proxy-response-processing-via. The proxy MUST pass
the response to the server transaction associated with the
response context. This will result in the response being
sent to the location now indicated in the topmost Via field
value. If the server transaction is no longer available to
handle the transmission, the element MUST forward the
response statelessly by sending it to the server transport.
Even after forwarding a final response, the proxy MUST
maintain the response context until all of its associated
transactions have been terminated.
9. Generate CANCELs
OPEN ISSUE #7: If CANCEL is restricted to INVITE only, this
behavior must restrict itself to INVITE requests.
OPEN ISSUE #122: The MUST below reflects list discussion,
but the question of how strong this requirement should be
was not formally closed.
If the forwarded response was a final response, the proxy
MUST generate a CANCEL request for all pending client
transactions associated with this response context. A proxy
SHOULD also generate a CANCEL request for all pending
client transactions associated with this response context
when it receives a 6xx response. A pending client
transaction is one that has received a provisional
response, but no final response and has not had an
associated CANCEL generated for it. Generating CANCEL
requests is described in Section 9.1.
16.7 Handling transport errors
If the transport layer notifies a proxy of an error when it tries to
forward a request (see Section 19.4), the proxy MUST behave as if the
forwarded request received a 400 response.
If the proxy is notified of an error when forwarding a response, it
drops the response. The proxy SHOULD NOT cancel any outstanding
client transactions associated with this response context due to this
notification.
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If a proxy cancels its outstanding client transactions, a
single malicious or misbehaving client can cause all
transactions to fail through its Via header field.
16.8 CANCEL Processing
A stateful proxy may generate a CANCEL to any other request it has
generated at any time. For instance, it may choose to generate
CANCELs based on having a transaction exceed the time specified in
the Expire header of certain requests, or as a result of any logic it
applies while forwarding requests. A proxy MUST cancel any pending
client transactions associated with a response context when it
receives a matching CANCEL request.
OPEN ISSUE #185: Should generating CANCEL at a proxy based
on Expires in INVITE be deprecated?
While a CANCEL request is handled in a stateful proxy by its own
server transaction, a new response context is not created for it.
Instead, the proxy layer searches its existing response contexts for
the server transaction handling the request associated with this
CANCEL. If a matching response context is found, the element MUST
immediately return a 200 OK response to the CANCEL request. In this
case, the element is acting as a user agent server as defined in
Section 8.2. Furthermore, the element MUST generate CANCEL requests
for all pending client transactions in the context as described in
Section 9.
If a response context is not found, the element does not have any
knowledge of the request to apply the CANCEL to. It MUST forward the
CANCEL request statelessly (it may have statelessly forwarded the
associated request previously).
16.9 Stateless proxy
When acting statelessly, a proxy is a simple message forwarder. Much
of the processing performed when acting statelessly is the same as
when behaving statefully. The differences are detailed here.
A stateless proxy does not have any notion of a transaction, or of
the response context used to describe stateful proxy behavior.
Instead, the stateless proxy takes messages, both requests and
responses, directly from the transport layer (See section 19). As a
result, stateless proxies do not retransmit messages on their own.
They do, however, forward all retransmission they receive (they do
not have the ability to distinguish a retransmission from the
original message). Furthermore, when handling a request statelessly,
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an element MUST NOT generate its own 100 Trying (or any other
provisional) response.
A stateless proxy must validate a request as described in Section
16.3
A stateless proxy must make a routing decision as described in
Section 16.4 with the following exception:
o A stateless proxy MUST choose one and only one destination
from the destination set. This choice MUST only rely on fields
in the message and time-invariant properties of the server. In
particular, a retransmitted request MUST be forwarded to the
same destination each time it is processed. Furthermore,
CANCEL and non-Routed ACK requests MUST generate the same
choice as their associated INVITE.
A stateless proxy must process the request before forwarding as
described in Section 16.5 with the following exceptions:
o The branch parameter on the inserted Via header field MUST be
the same each time a retransmitted request is forwarded. Thus
for a stateless proxy, the branch parameter calculation MUST
only depend on message parameters affecting the routing of the
request which are invariant on retransmission.
o The request is sent directly to the transport layer instead of
through a client transaction. If the next-hop destination
parameters don't provide an explicit destination, the element
applies the procedures of Section 24 to the Request-URI to
determine where to send the request.
Stateless proxies MUST NOT perform special processing for CANCEL
requests. They are processed by the above rules as any other
requests.
Response processing as described in Section 16.6 does not apply to a
proxy behaving statelessly. When a response arrives at a stateless
proxy, the proxy inspects the address in the first (topmost) Via
header value. If that address matches the proxy, the proxy MUST
remove that value from the response and forward the result to the
location indicated in the next Via header value. Unless specified
otherwise, the proxy MUST NOT remove any other header values or the
message body. If the address does not match the proxy, the message
MUST be silently discarded.
17 Transactions
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SIP is fundamentally a transactional protocol. This means that
interactions between components take place in a series of independent
message exchanges. Specifically, a SIP transaction consists of a
single request, and any responses to that request (which include zero
or more provisional responses and one or more final responses). In
the case of a transaction where the request was an INVITE (known as
an INVITE transaction), the transaction also includes the ACK only if
the final response was not a 2xx response. If the response was a 2xx,
the ACK is not considered part of the transaction.
The reason for this separation is rooted in the importance
of delivering all 200 OK responses to an INVITE to the UAC.
To deliver them all to the UAC, the UAS alone takes
responsibility for retransmitting them, and the UAC alone
takes responsibility for acknowledging them with ACK. Since
this ACK is retransmitted only by the UAC, it is
effectively considered its own transaction.
Transactions have a client side and a server side. The client side is
known as a client transaction, and the server side, as a server
transaction. The client transaction sends the request, and the server
transaction sends the response. The client and server transactions
are logical functions that are embedded in any number of elements.
Specifically, they exist within user agents and stateful proxy
servers. Consider the example of Section 4. In this example, the UAC
executes the client transaction, and its outbound proxy executes the
server transaction. The outbound proxy also executes a client
transaction, which sends the request to a server transaction in the
inbound proxy. That proxy also executes a client transaction, which
in turn, sends the request to a server transaction in the UAS. This
is shown pictorially in Figure 4.
A stateless proxy does not contain a client or server transaction.
The transaction exists between the UA or stateful proxy on one side
of the stateless proxy, and the UA or stateful proxy on the other
side. As far as SIP transactions are concerned, stateless proxies are
effectively transparent. The purpose of the client transaction is to
receive a request from the element the client is embedded in (call
this element the "Transaction User" or TU; it can be a UA or a
stateful proxy), and reliably deliver the request to that server
transaction. The client transaction is also responsible for receiving
responses, and delivering them to the TU, filtering out any
retransmissions or disallowed responses (such as a response to ACK).
In the case of an INVITE transaction, that includes generation of the
ACK request for any final response excepting a 2xx response.
Similarly, the purpose of the server transaction is to receive
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+---------+ +---------+ +---------+ +---------+
| +-+|Request |+-+ +-+|Request |+-+ +-+|Request |+-+ |
| |C||------->||S| |C||------->||S| |C||------->||S| |
| |l|| ||e| |l|| ||e| |l|| ||e| |
| |i|| ||r| |i|| ||r| |i|| ||r| |
| |e|| ||v| |e|| ||v| |e|| ||v| |
| |n|| ||e| |n|| ||e| |n|| ||e| |
| |t|| ||r| |t|| ||r| |t|| ||r| |
| | || || | | || || | | || || | |
| |T|| ||T| |T|| ||T| |T|| ||T| |
| |r|| ||r| |r|| ||r| |r|| ||r| |
| |a|| ||a| |a|| ||a| |a|| ||a| |
| |n|| ||n| |n|| ||n| |n|| ||n| |
| |s||Response||s| |s||Response||s| |s||Response||s| |
| +-+|<-------|+-+ +-+|<-------|+-+ +-+|<-------|+-+ |
+---------+ +---------+ +---------+ +---------+
UAC Outbound Inbound UAS
Proxy Proxy
Figure 4: Transaction relationships
requests from the transport layer, and deliver them to the TU. The
server transaction filters any request retransmissions from the
network. The server transaction accepts responses from the TU, and
delivers them to the transport layer for transmission over the
network. In the case of an INVITE transaction, it absorbs the ACK
request for any final response excepting a 2xx response.
The 2xx response, and the ACK for it, have special treatment. This
response is retransmitted only by a UAS, and its ACK generated only
by the UAC. This end-to-end treatment is needed so that a caller
knows the entire set of users that have accepted the call. Because of
this special handling, retransmissions of the 2xx response are
handled by the UA core, not the transaction layer. Similarly,
generation of the ACK for the 2xx is handled by the UA core. Each
proxy along the path merely forwards each 2xx response to INVITE, and
its corresponding ACK.
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17.1 Client transaction
The client transaction provides its functionality through the
maintenance of a state machine.
The TU communicates with the client transaction through a simple
interface. When the TU wishes to initiate a new transaction, it
creates a client transaction, and passes it the SIP request to send,
a value for timer C (described below), and an IP address, port, and
transport to send it to. The client transaction begins execution of
its state machine. Valid responses are past up to the TU from the
client transaction.
There are two types of client transaction state machines, depending
on the method the request passed by the TU. One handles client
transactions for INVITE request. This type of machine is referred to
as an INVITE client transaction. Another type handles client
transactions for all requests except INVITE and ACK. This is referred
to as a non-INVITE client transaction. There is no client transaction
for ACK. If the TU wishes to send an ACK, it passes one directly to
the transport layer for transmission.
The INVITE transaction is different from those of other methods
because of its extended duration. Normally, human input is required
in order to respond to an INVITE. The long delays expected for
sending a response argue for a three way handshake. Requests of other
methods, on the other hand, are expected to completely rapidly. In
fact, because of its reliance on just a two way handshake, TUs SHOULD
respond immediately to non-INVITE requests. Protocol extensions which
require longer durations for generation of a response (such as a new
method that does require human interaction) SHOULD instead use two
transactions - one to send the request, and another in the reverse
direction to convey the result of the request.
17.1.1 INVITE Client Transaction
17.1.1.1 Overview of INVITE Transaction
The INVITE transaction consists of a three-way handshake. The client
transaction sends an INVITE, the server transaction sends responses,
and the client transaction sends an ACK. For unreliable transports
(such as UDP), the client transaction will retransmit requests at an
interval that starts at T1 seconds and doubles after every
retransmission. The request is not retransmitted over reliable
transports. After receiving a 1xx response, any retransmissions cease
altogether, and the client waits for further responses. The server
transaction can send additional 1xx responses, which are not
transmitted reliably. Eventually, the server transaction decides to
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send a final response. For unreliable transports, that response is
retransmitted periodically, and for reliable transports, its sent
once. For each final response that is received at the client
transaction, the client transaction sends an ACK, the purpose of
which is to quench retransmissions of the response.
17.1.1.2 Formal Description
The state machine for the INVITE client transaction is shown in
Figure 5. The initial state, "calling", MUST be entered when the TU
initiates a new client transaction with an INVITE request. The client
transaction MUST pass the request to the transport layer for
transmission (see Section 19). If an unreliable transport is being
used, the client transaction SHOULD start timer A with a value of T1,
and SHOULD NOT start timer A when a reliable transport is being used
(Timer A controls request retransmissions). For any transport, the
client transaction MUST start timer B with a value of 64*T1 seconds
(Timer B controls transaction timeouts).
When timer A fires, the client transaction SHOULD retransmit the
request by passing it to the transport layer, and SHOULD reset the
timer with a value of 2*T1. When the timer fires 2*T1 seconds later,
the request SHOULDbe retransmitted again (assuming the client
transaction is still in this state). This process SHOULDcontinue, so
that the request is retransmitted with intervals that double after
each transmission. These retransmissions SHOULDonly be done while the
client transaction is in the "calling" state.
The default value for T1 is 500ms. T1 is an estimate of the RTT
between the client and server transactions. The optional RTT
estimation procedure of Section 17.3 MAY be followed, in which case
the resulting estimate MAY be used instead of 500ms. If no RTT
estimation is used, other values MAYbe used in private networks where
it is known that RTT has a different value. On the public Internet,
T1 MAY be chosen larger, but SHOULD NOT be smaller.
If the client transaction is still in the "calling" when timer B
fires, the client transaction SHOULD inform the TU that a timeout has
occurred. The client transaction MUST NOT generate an ACK. The value
of 64*T1 is equal to the amount of time required to send seven
requests in the case of an unreliable transport.
If the client transaction receives a provisional response while in
the "calling" state, it transitions to the "proceeding" state. Upon
entering this state, the client transaction MUST start timer C with
the value provided by the TU when the client transaction was created.
This timeout dictates how long the client transaction waits for a
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final response before giving up (i.e., roughly how long does it "let
the phone ring"). In the "proceeding" state, the client transaction
SHOULD NOT retransmit the request any longer. Furthermore, the
provisional response MUST be passed to the TU. Any further
provisional responses MUST be passed up to the TU while in the
"proceeding" state. When timer C fires, the client transaction MUST
transition to the terminated state, and it MUST inform the TU of the
timeout.
When in either the "calling" or "proceeding" states, reception of a
response with status code from 300-699 MUST cause the client
transaction to transition to "completed". The client transaction MUST
pass the received response up to the TU, and it MUST generate an ACK
request, even if the transport is reliable (guidelines for
constructing the ACK from the response are given in Section 17.1.1.3)
and then pass the ACK to the transport layer for transmission. The
ACK MUST be sent to the same address, port and transport that the
original request was sent to. The client transaction SHOULD start
timer D when it enters the "completed" state, with a value of T3
seconds for unreliable transports, and zero seconds for reliable
transports. T3 is the total amount of time that the server
transaction can remain in the "completed" state when unreliable
transports are used. For the default values of the timers below, this
is 16 seconds.
OPEN ISSUE #210: Timer D should be based on the values of
the timers selected at the server, but these values aren't
known by the client. We could alternatively specify an
absolute minimum.
Any retransmissions of the final response that are received while in
the "completed" state SHOULD cause the ACK to be re-passed to the
transport layer for retransmission, but the newly received response
MUST NOT be passed up to the TU. A retransmission of the response is
defined as any response which would match the same client
transaction, based on the rules of Section 17.1.3.
If timer D fires while the client transaction is in the "completed"
state, the client transaction MUST move to the terminated state, and
it MUST inform the TU of the timeout.
When in either the "calling" or "proceeding" states, reception of a
2xx response MUST cause the client transaction to enter the
terminated state, and the response MUST be passed up to the TU. The
handling of this response depends on whether the TU is a proxy core
or a UAC core. A UAC core will handle generation of the ACK for this
response, while a proxy core will always forward the 200 OK upstream.
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|INVITE from TU
Timer A fires |INVITE sent
Reset A, V Timer B fires
INVITE sent +-----------+ t.o. to TU
+---------| |---------------+
| | Calling | |
+-------->| |-------------->|
+-----------+ 2xx |
300-699 | | 2xx to TU |
ACK sent | |1xx |
+---------------+ |1xx to TU |
| | |
| 1xx V Timer C fires |
| 1xx to TU -----------+ t.o. to TU |
| +---------| |-------------->|
| | |Proceeding | |
| +-------->| |-------------->|
| +-----------+ 2xx |
| 300-699 | 2xx to TU |
| ACK sent, | |
| resp. to TU| |
| | | NOTE:
| 300-699 V |
| ACK sent +-----------+ | transitions
| +---------| | | labeled with
| | | Completed | | the event
| +-------->| | | over the action
| +-----------+ | to take
| ^ | |
| | | Timer D fires |
+--------------+ | - |
| |
V |
+-----------+ |
| | |
| Terminated|<--------------+
| |
+-----------+
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The differing treatment of 200 OK between proxy and UAC is the reason
that handling of it does not take place in the transaction layer.
The client transaction MUST be destroyed the instant it enters the
terminated state. This is actually necessary to guarantee correct
operation. The reason is that 2xx responses to an INVITE are treated
differently; each one is forwarded by proxies, and the ACK handling
in a UAC is different. Thus, each 2xx needs to be passed to a proxy
core (so that it can be forwarded) and to a UAC core (so it can be
acknowledged). No transaction layer processing takes place. Whenever
a response is received by the transport, if the transport layer finds
no matching client transaction (using the rules of Section 17.1.3,
the response is passed directly to the core. Since the matching
client transaction is destroyed by the first 2xx, subsequent 2xx will
find no match and therefore be passed to the core.
17.1.1.3 Construction of the ACK Request
The ACK request constructed by the client transaction MUST contain
values for the Call-ID, From, and Request-URI which are equal to the
values of those headers in the request that created the client
transaction (call this the "original request"). The To field in the
ACK MUST equal the To field in the response being acknowledged, and
will therefore usually differ from the To field in the original
request by the addition of the tag parameter. The ACK MUST contain a
single Via header, and this MUST be equal to the top Via header of
the original request. The ACK request MUST NOT contain any Route
headers. The CSeq header in the ACK MUST contain the same value for
the sequence number as was present in the original request, but the
method parameter MUST be equal to "ACK".
These rules for construction of ACK only apply to the client
transaction. A UAC core which generates an ACK for 2xx MUST instead
follow the rules described in Section 13.
For example, consider the following request:
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 10.1.3.3
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=88sja8x
Call-ID: 987asjd97y7atg@10.1.3.3
CSeq: 986759 INVITE
The ACK request for a non-2xx final response to this request would
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look like:
ACK sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 10.1.3.3
To: Bob <sip:bob@biloxi.com>;tag=99sa0xk
From: Alice <sip:alice@atlanta.com>;tag=88sja8x
Call-ID: 987asjd97y7atg@10.1.3.3
CSeq: 986759 ACK
17.1.2 non-INVITE Client Transaction
17.1.2.1 Overview of the non-INVITE Transaction
non-INVITE transactions do not make use of ACK. They are a simple
request-response interaction. For unreliable transports, requests are
retransmitted at an interval which starts at T1, and doubles until it
hits T2. If a provisional response is received, retransmissions
continue for unreliable transports, but at an interval of T2. The
server transaction retransmits the last response it sent (which can
be a provisional or final response) only when a retransmission of the
request is received. This is why request retransmissions need to
continue even after a provisional response, they are what ensure
reliable delivery of the final response.
Unlike an INVITE transaction, a non-INVITE transaction has no special
handling for the 2xx response. The result is that only a single 2xx
response to a non-INVITE is ever delivered to a UAC.
17.1.2.2 Formal Description
The state machine for the non-INVITE client transaction is shown in
Figure 6. It is very similar to the state machine for INVITE.
The "Trying" state is entered when the TU initiates a new client
transaction with a request. When entering this state, the client
transaction SHOULD set Timer F to fire in T3 seconds. The request
MUST be passed to the transport layer for transmission. If an
unreliable transport is in use, the client transaction MUST set timer
E to fire in T1 seconds. If timer E fires while still in this state,
the timer is reset, but this time with a value of MIN(2*T1, T2). When
the timer fires again, it is reset to a MIN(4*T1, T2). This process
continues, so that retransmissions occur with an exponentially
increasing inverval that caps at T2. The default value of T2 is 4s,
and it represents the amount of time a non-INVITE server transaction
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will take to respond to a request, if it does not respond
immediately. For the default values of T1 and T2, this results in
intervals of 500 ms, 1 s, 2 s, 4 s, 4 s, 4s, etc.
If Timer F fires while the client transaction is still in the
"Trying" state, the client transaction SHOULD inform the TU about the
timeout, and then it SHOULDenter the "Terminated" state. If a
provisional response is received while in the "Trying" state, the
response MUST be passed to the TU, and then the client transaction
SHOULD move to the "Proceeding" state. If a final response (status
codes 200-699) is received while in the "Trying" state, the response
MUST be passed to the TU, and the client transaction MUST transition
to the "Completed" state.
If Timer E fires while in the "Proceeding" state, the request MUST be
passed to the transport layer for retransmission, and Timer E MUST be
reset with a value of T2 seconds. If timer F fires while in the
"Proceeding" state, the TU MUST be informed of a timeout, and the
client transaction MUST transition to the terminated state. If a
final response (status codes 200-699) is received while in the
"Proceeding" state, the response MUST be passed to the TU, and the
client transaction MUST transition to the "Completed" state.
Once the client transaction enters the "Completed" state, it MUST set
Timer K to fire in T4 seconds for unreliable transports, and zero
seconds for reliable transports. The "Completed" state exists to
buffer any additional response retransmissions that may be received
(which is why the client transaction remains there only for
unreliable transports). T4 represents the amount of time the network
will take to clear messages between client and server transactions.
The default value of T4 is 5s. A response is a retransmission when it
matches the same transaction, using the rules specified in Section
17.1.3. If Timer K fires while in this state, the client transaction
MUST transition to the "Terminated" state.
OPEN ISSUE #211: This special treatment for reliable
transports, where the state machine transactions directly
to terminated, is new.
Once the transaction is in the terminated state, it MUST be
destroyed. As with client transactions, this is needed to ensure
reliability of the 2xx responses to INVITE.
17.1.3 Matching Responses to Client Transactions
When the transport layer in the client receives a response, it has to
figure out which client transaction will handle the response, so that
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|Request from app
|send request
Timer E V Timer F
send request +-----------+ t.o. to TU
+---------| |-------------------+
| | Trying | |
+-------->| | |
+-----------+ |
200-699 | | |
resp. to TU | |1xx |
+---------------+ |resp. to TU |
| | |
| Timer E V Timer F |
| send req +-----------+ t.o.to TU |
| +---------| |------------------>|
| | |Proceeding | |
| +-------->| |-----+ |
| +-----------+ |1xx |
| | ^ |resp to TU |
| 200-699 | +--------+ |
| resp. to TU | |
| | |
| V |
| +-----------+ |
| | | |
| | Completed | |
| | | |
| +-----------+ |
| ^ | |
| | | Timer K |
+--------------+ | - |
| |
V |
NOTE: +-----------+ |
| | |
transitions | Terminated|<------------------+
labeled with | |
the event +-----------+
over the action
to take
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Internet Draft SIP October 26, 2001
the processing of Sections 17.1.1 and 17.1.2 can take place.
A response matches a client transaction through a comparison process
with fields in the request that created the transaction.
Specifically, the From, Call-ID, CSeq, and the topmost Via header
MUST match the same fields in the request, using the matching
operations for those headers defined in Section 22. If the To field
in the request had a tag, the To field in the response MUST match the
To field in the request, as described in Section 22.37. However, if
the To field in the request did not contain a tag, the To field in
the response MUST match that in the request, except that the tag MUST
NOT be considered as part of the matching process. This is needed
since a UAS will add a tag to the To field of the response.
17.1.4 Handling Transport Errors
When the client transaction sends a request to the transport layer to
be sent, the following procedures are followed if the transport layer
indicates a failure.
The client transaction SHOULD inform the TU that a transport failure
has occurred, and the client transaction SHOULD transition directly
to the terminated state.
17.2 Server Transaction
The server transaction is responsible for the delivery of requests to
the TU, and the reliable transmission of responses. It accomplishes
this through a state machine. Server transactions are created by the
core when a request is received, and transaction handling is desired
for that request (this won't always be the case).
As with the client transactions, the state machine depends on whether
the received request is an INVITE request or not.
17.2.1 INVITE Server Transaction
The state diagram for the INVITE server transaction is shown in
Figure 7.
When a server transaction is constructed with a request, it enters
the "Proceeding" state. The server transaction MUST generate a 100
response (not any status code - the specific value of 100) unless it
knows that the TU will generate a provisional or final response
within 200 ms, in which case it MAY generate a 100 response. This
provisional response is needed to rapidly quench request
retransmissions in order to avoid network congestion. The request
Various Authors [Page 100]
Internet Draft SIP October 26, 2001
MUST be passed to the TU.
The TU passes any number of provisional responses to the server
transaction. So long as the server transaction is in the "Proceeding"
state, each of these MUST be passed to the transport layer for
transmission. They are not sent reliably (they are not
retransmitted), and do not cause a change in the state of the server
transaction. If a request retransmission is received while in the
"Proceeding" state, the most recent provisional response that was
received from the TU MUST be passed to the transport layer for
retransmission. A request is a retransmission if it matches the same
server transaction based on the rules of Section 17.2.3.
If, while in the "proceeding" state, the TU passes a 2xx Response to
the server transaction, the server transaction MUST pass this
response to the transport layer for transmission. It is not
retransmitted by the server transaction; retransmissions of 2xx
responses are handled by the TU. The server transaction MUST then
transition to the "terminated" state.
While in the "Proceeding" state, if the TU passes a response with
status code from 300 to 699 to the server transaction, the response
MUST be passed to the transport layer for transmission, and the state
machine MUST enter the "Completed" state. For unreliable transports,
timer G is set to fire in T1 seconds, and is not set to fire for
reliable transports.
This is a change from RFC2543, where responses were always
retransmitted, even over reliable transports.
When the "Completed" state is entered, timer H MUST be set to fire in
64*T1 seconds, for all transports. Timer H determines when the server
transaction gives up retransmitting the response. Its value is chosen
to equal Timer B, the amount of time a client transaction will
continue to retry sending a request. If timer G fires, the response
is passed to the transport layer once more for retransmission, and
timer G is set to fire in MIN(2*T1, T2) seconds. From then on, when
timer G fires, the response is passed to the transport again for
transmission, and timer G is reset with a value that doubles, unless
that value exceeds T2, in which case it is reset with the value of
T2. This is identical to the retransmit behavior for requests in the
"Trying" state of the non- INVITE client transaction. Furthermore,
while in the "completed" state, if a request retransmission is
received, the server SHOULD pass the response to the transport for
retransmission.
If an ACK is received while the server transaction is in the
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|INVITE
|pass to TU, send 100
INVITE V
send response+-----------+
+--------| |--------+101-199 from TU
| | Proceeding| |send response
+------->| |<-------+
+-----------+
300-699 from TU | |2xx from TU
send response | |send response
| +-------------------+
| |
INVITE V Timer G fires |
send response+-----------+ send response |
+--------| |--------+ |
| | Completed | | |
+------->| |<-------+ |
+-----------+ |
| | |
ACK | | |
- | +------------------>+
| Timer H fires |
V fail to TU |
+-----------+ |
| | |
| Confirmed | |
| | |
+-----------+ |
| |
|Timer I fires |
|- |
| |
V |
+-----------+ |
| | |
| Terminated|<---------------+
| |
+-----------+
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"Completed" state, the server transaction MUST transition to the
"confirmed" state. As Timer G is ignored in this state, any
retransmissions of the response will cease.
If timer H fires while in the "Completed" state, it implies that the
ACK was never received. In this case, the server transaction MUST
transition to the terminated state, and MUST indicate to the TU that
a transaction failure has occurred.
The purpose of the "confirmed" state is to absorb any additional ACK
messages that arrive, triggered from retransmissions of the final
response. When this state is entered, timer I is set to fire in T4
seconds for unreliable transports, and zero seconds for reliable
transports. Once timer I fires, the server MUST transition to the
"Terminated" state.
Once the transaction is in the terminated state, it MUST be
destroyed. As with client transactions, this is needed to ensure
reliability of the 2xx responses to INVITE.
17.2.2 non-INVITE Server Transaction
The state machine for the non-INVITE server transaction is shown in
Figure 8.
The state machine is initialized in the "Trying" state, and is passed
a request other than INVITE or ACK when initialized. This request is
passed up to the TU. Once in the "Trying" state, any further request
retransmissions are discarded. A request is a retransmission if it
matches the same server transaction, using the rules specified in
Section 17.2.3.
While in the "Trying" state, if the TU passes a provisional response
to the server transaction, the server transaction MUST enter the
"Proceeding" state. The response MUST be passed to the transport
layer for transmission. Any further provisional responses that are
received from the TU while in the "Proceeding" state MUST be passed
to the transport layer for transmission. If a retransmission of the
request is received while in the "Proceeding" state, the most
recently sent provisional response MUST be passed to the transport
layer for retransmission. If the TU passes a final response (status
codes 200-699) to the server while in the "Proceeding" state, the
transaction MUST enter the "Completed" state, and the response MUST
be passed to the transport layer for transmission.
When the server transaction enters the "Completed" state, it MUST set
Timer J to fire in T3 seconds for unreliable transports, and zero
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seconds for reliable transports. While in the "Completed" state, the
server transaction MUST pass the final response to the transport
layer for retransmission whenever a retransmission of the request is
received. Any other final responses passed by the TU to the server
transaction MUST be discarded while in the "Completed" state. The
server transaction remains in this state until Timer J fires, at
which point it MUST transition to the "Terminated" state.
The server transaction MUST be destroyed the instant it enters the
"Terminated" state.
17.2.3 Matching Requests to Server Transactions
When an INVITE or ACK request is received from the network by the
server, it has to be matched to an existing INVITE transaction. The
INVITE request matches a transaction if the Request-URI, To, From,
Call-ID, CSeq, and top Via header match those of the INVITE request
which created the transaction. The ACK request matches a transaction
if the Request-URI, From, Call-ID, CSeq method (not the number), and
top Via header match those of the INVITE request which created the
transaction, and the To field of the ACK matches the To field of the
response sent by the server transaction (which then includes the
tag). Matching is done based on the matching rules defined for each
of those headers. The usage of the tag in the To field helps
disambiguate ACK for 2xx from ACK for other responses at a proxy
which may have forwarded both responses (which can occur in unusual
conditions).
For all other request methods, a request is matched to a transaction
if the Request-URI, To, From, Call-ID and Cseq (including the method)
and top Via header match those of the request which created the
transaction. Matching is done based on the matching rules defined for
each of those headers.
Because the matching rules include the Request-URI, the server cannot
match a response to a transaction. When the TU passes a response to
the server, it must inform the TU which transaction the response is
for.
17.3 RTT Estimation
Most of the timeouts used in the transaction state machines derive
from T1, which is an estimate of the RTT between the client and
server transactions. This subsection defines optional procedures that
a client can use to build up estimates of the RTT to a particular IP
address. To perform this procedure, the client MUST maintain a table
of variables for each destination IP address to which an RTT estimate
is being made.
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|Request received
|pass to TU
V
+-----------+
| |
| Trying |-------------+
| | |
+-----------+ |200-699 from TU
| |send response
|1xx from TU |
|send response |
| |
Request V 1xx from TU |
send response+-----------+send response|
+--------| |--------+ |
| | Proceeding| | |
+------->| |<-------+ |
+-----------+ |
| |
| |
|200-699 from TU |
|send response |
Request V |
send response+-----------+ |
+--------| | |
| | Completed |-------------+
+------->| |
+-----------+
|
|Timer J fires
|-
|
V
+-----------+
| |
| Terminated|
| |
+-----------+
Internet Draft SIP October 26, 2001
OPEN ISSUE #212: Is destination IP address the right index
for an RTT estimate? How about Request-URI?
If a client wishes to measure RTT for a particular IP address, it
MUST include a Timestamp header into a request containing the time
when the request is initially created and passed to a new client
transaction, which transmits the request. If a 100 response (not any
1xx, only the 100 response) is received before the client transaction
generates a retransmission, an RTT estimate is made. This is
consistent with the RFC 2988 requirements on TCP for using Karn's
algorithm in RTT estimation.
The estimate, called R, is made by computing the difference between
the current time and the value of Timestamp header in the 100
response. The value of R is applied to the estimation of RTO as
described in Section 2 of RFC 2988 [24], with the following
differences. First, the initial value of RTO is 500 ms for SIP, not 3
s as is used for TCP. Second, there is no minimum value for the RTO,
as there is for TCP, if SIP is being run on a private network. When
run on the public Internet, the minimum is 500 ms, as opposed to 1 s
for TCP. This difference is because of the expected usage of SIP in
private networks where rapid call setup times are service critical.
Once RTO is computed, the timer T1 is set to the value of RTO, and
all other timers scale proportionally as described above.
18 Reliability of Provisional Responses
Placeholder.
Reliability of provisional responses will be incorporated into bis.
This is a heads up on that.
19 Transport
The transport layer is responsible for the actual transmission of
requests and responses over network transports. This includes
determination of the connection to use for a request or response, in
the case of connection oriented transports.
The transport layer is responsible for managing any persistent
connections (for transports like TCP, TLS and SCTP) including ones it
opened, as well as ones opened to it. This includes connections
opened by the client or server transports, so that connections are
shared between client and server transport functions. It is
RECOMMENDED that connections be kept open for some implementation
defined time after the last message was sent or received over that
connection. This time SHOULD be at least 16 seconds in order to
ensure with high probability that responses can be sent over the same
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connection a request was sent.
All SIP elements MUST support UDP at a minimum.
19.1.1 Sending Requests
The client side of the transport layer is responsible for sending the
request and receiving responses. The user of the transport layer
passes the client transport the request, an IP address, port,
transport, and possibly TTL for multicast destinations.
A client that sends a request to a multicast address MUST add the
"maddr" parameter to its Via header field, and SHOULD add the "ttl"
parameter. (In that case, the maddr parameter SHOULD contain the
destination multicast address, although under exceptional
circumstances it MAY contain a unicast address.) Requests sent to
multicast groups SHOULD be scoped to ensure that they are not
forwarded beyond the administrative domain to which they were
targeted. This scooping MAY be done with either TTL or administrative
scopes [19], depending on what is implemented in the network.
It is important to note that the layers above the transport layer do
not operate differently for multicast as opposed to unicast requests.
This means that SIP treats multicast more like anycast, assuming that
there is a single recipient generating responses to requests. If this
is not the case, the first response will end up "winning", based on
the client transaction rules. Any other responses from different UA
will appear as retransmissions and be discarded. This limits the
utility of multicast to cases where an anycast type of function is
desired, such as registrations.
OPEN ISSUE #7: This is a proposed resolution to whether or
not multicast should be removed entirely.
Before a request is sent, the client transport MUST insert a value of
the sent-by field into the Via header. This field contains an IP
address or host name, and port. In certain cases discussed in Section
19.2.2, this IP address and port are used to construct a SIP URL for
sending the response. The transport layer MUST be prepared to receive
incoming connections (and receive responses sent over such
connections) on any IP addresses and ports that this SIP URL might
resolve to using the procedures defined in Section 24. The transport
layer MUST also be prepared to receive an incoming connection on the
source IP address that the request was sent from, and port number in
the sent-by field. The client transport MUST also be prepared to
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receive the response on the same connection used to send the request.
For unreliable unicast transports, the client transport MUST be
prepared to receive responses on the source IP address that the
request is sent from (as responses are sent back to the source
address), but the port number in the sent-by field. Furthermore, as
with reliable transports, in certain cases the IP address and port
are used to construct a URL for sending the response. The client
transport MUST be prepared to receive responses on any IP
address/port combinations that this SIP URL might resolve to using
the procedures of Section 24.
For multicast, the client transport MUST be prepared to receive
responses on the same multicast group and port that the request is
sent to.
If a request is destined to an IP address, port, and transport to
which an existing connection is open, it is RECOMMENDED that this
connection be used to send the request, but another connection MAY be
opened and used.
If a request is sent using multicast, it is sent to the group
address, port, and TTL provided by the transport user. If a request
is sent using unicast unreliable transports, it is sent to the IP
address and port provided by the transport user.
19.1.2 Receiving Responses
When a response is received, the client transport examines the top
Via header. If the value of the sent-by parameter in that header does
not correspond to a value that the client transport is configured to
insert into requests, the response MUST be rejected.
If there are any client transactions in existence, the client
transport uses the matching procedures of Section 17.1.3 to attempt
to match the response to an existing transaction. If there is a
match, the response MUST be passed to that transaction. Otherwise,
the response MUST be passed to the core (whether it be stateless
proxy, stateful proxy, or UA) for further processing. Handling of
these "stray" responses is dependent on the core (a stateless proxy
will forward all responses, for example).
19.2.1 Receiving Requests
When the server transport receives a request over any transport, it
MUST examine the value of the sent-by parameter in the top Via header
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field. If the host portion of the sent-by parameter contains a domain
name, or if it contains an IP address that differs from the packet
source address, the server MUST add a "received" attribute to that
Via header field. This attribute MUST contain the source address that
the packet was received from. This is to assist the server transport
layer in sending the response, since it must be sent to the source IP
address that the request came from.
Consider a request received by the server transport which looks like,
in part:
INVITE sip:bob@Biloxi.com SIP/2.0
Via: SIP/2.0/UDP bobspc.biloxi.com:5060
The request is received with a source IP address of 1.2.3.4. Before
passing the request up, the transport would add a received parameter,
so that the request would look like, in part:
INVITE sip:bob@Biloxi.com SIP/2.0
Via: SIP/2.0/UDP bobspc.biloxi.com:5060
Next, the client transport attempts to match the request to the
client transaction. It does so using the matching rules described in
Section 17.2.3. If a matching server transaction is found, the
request is passed to that transaction for processing. If no match is
found, the request is passed to the core, which may decide to
construct a new server transaction for that request.
19.2.2 Sending Responses
The server transport uses the value of the top Via header in order to
determine where to send a response. It MUST follow the following
process:
o If the "sent-protocol" is a reliable transport protocol such
as TCP, TLS or SCTP, the response MUST be sent using the
existing connection to the source of the original request that
created the transaction, if that connection is still open.
This does require the server transport to maintain an
association between server transactions and transport
connections. If that connection is no longer open, the server
MAY open a connection to the IP address in the received
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parameter, if present, using the port in the sent-by value, or
the default port for that transport, if no port is specified
(5060 for UDP and TCP, 5061 for TLS and SSL). If that
connection attempt fails, the server SHOULD construct a SIP
URL of the form "sip:<sent-by host>;transport=<sent-protocol>"
and then use the procedures defined in Section 24 to determine
the IP address and port to open the connection and send the
response to.
o Otherwise, if the Via header field contains a "maddr"
parameter, forward the response to the address listed there,
using the port indicated in "sent-by", or port 5060 if none is
present. If the address is a multicast address, the response
SHOULD be sent using the TTL indicated in the "ttl" parameter,
or with a TTL of 1 if that parameter is not present.
o Otherwise (for unreliable unicast transports), if the top Via
has a received parameter, send the response to the address in
the "received" parameter, using the port indicated in the
"sent-by" value, or using port 5060 if none is specified
explicitly. If this fails, e.g., elicits an ICMP "port
unreachable" response, send the response to the address in the
"sent-by" parameter. The address to send to is determined by
constructing a SIP URL of the form "sip:<sent-by>", and then
using the DNS procedures defined in Section 24 to send the
response.
o Otherwise, if it is not receiver-tagged, send the response to
the address indicated by the "sent-by" value.
In the case of message oriented transports (such as UDP), if the
message has a Content-Length header, the message body is assumed to
contain that many bytes. If there are additional bytes in the
transport packet below the end of the body, they MUST be discarded.
If the transport packet ends before the end of the message body, this
is considered an error. If the message is a response, it MUST be
discarded. If its a request, the element SHOULD generate a 400 class
response. If the message has no Content-Length header, the message
body is assumed to end at the end of the transport packet.
In the case of stream oriented transports (such as TCP), the
Content-Length header indicates the size of the body. The Content-
Length header MUST be used with stream oriented transports.
19.4 Error Handling
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Error handling is independent of whether the message was a request or
response.
If the transport user asks for a message to be sent over an
unreliable transport, and the result is an ICMP error, the behavior
depends on the type of ICMP error. A host, network, port or protocol
unreachable errors, or parameter problem errors SHOULD cause the
transport layer to inform the transport user of a failure in sending.
Source quench and TTL exceeded ICMP errors SHOULD be ignored.
If the transport user asks for a request to be sent over a reliable
transport, and the result is a connection failure, the transport
layer SHOULD inform the transport user of a failure in sending.
20 Security Considerations
The fundamental security issues confronting SIP are: preserving the
confidentiality and integrity of messaging, preventing replay attacks
or message spoofing, ensuring the privacy of the participants in a
session, and preventing denial of service attacks.
SIP messages frequently contain sensitive information about their
senders not just what they have to say, but with whom they
communicate, when they communicate and for how long, and from where
they participate in sessions. Many applications and their users
require that this sort of private information be hidden from any
parties that do not need to know it.
Encryption provides the best means to preserve the confidentiality of
signaling it can also guarantee that messages are not modified by any
malicious intermediaries. However, SIP requests and responses cannot
be encrypted end-to-end (that is, between a pair of distinct user
agents who share encryption keys) in their entirety because message
fields such as the Request-URI, Route and Via need, in most network
architectures, to be visible to proxies so that SIP requests are
routed correctly. Note that proxy servers need to modify signaling as
well (adding Via headers) in order for SIP to function. Proxy servers
must therefore be a part of trust relationships in SIP networks.
Note that there are also less direct ways in which private
information can be divulged. If a user or service chooses to be
reachable at an address that is guessable from the person's name and
organizational affiliation (which describes most addresses of
record), the traditional method of ensuring privacy by having an
unlisted "phone number" is compromised. A user location service can
infringe on the privacy of the recipient of a session invitation by
divulging their specific whereabouts to the caller; an implementation
consequently SHOULD be able to restrict, on a per-user basis, what
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kind of location and availability information is given out to certain
classes of callers.
SIP entities also have a need to identify one another in a secure
fashion. Ordinarily a SIP UA asserts an identity for the initiator of
a request in the From header field, but in many systems this
information is controlled directly by the end user, and thus spoofing
the contents of the From is trivial. When a SIP endpoint asserts the
identity of its user to a peer user agent or to a proxy server, that
identity should in some way be verifiable. A cryptographic
authentication mechanism is provided in SIP to address this
requirement.
The most comprehensive mechanisms for securing SIP messages
(providing confidentiality and integrity guarantees for signaling as
well as authentication) make use of transport or network layer
encryption. encryption encrypts the entire SIP request or response
on the wire so that packet sniffers or other eavesdroppers cannot see
who is calling whom.
Note that the security of SIP signaling itself has no bearing on the
security of protocols used in concert with SIP such as RTP, or with
any MIME types carried as SIP bodies, such as SDP. Any media
associated with a session can be encrypted end-to-end without any of
the problems associated with encrypting SIP signaling. Media
encryption is outside the scope of this document.
20.1 Transport and Network Layer Security
SIP requests and responses MAY be protected by security mechanisms at
the transport or network layer. No particular mechanism is
recommended by this document, but two popular alternatives are
briefly examined: protection at the transport layer can be afforded
by TLS [25], and network layer security is provided by IPSec [26].
Transport or network layer security encrypts signaling traffic,
guaranteeing message confidentiality and integrity (note however that
the originator and recipient of a session may be deducible by
observers performing a network traffic analysis). The keys used to
establish encrypt traffic can also be used to verify an asserted
identity in many architectures, and therefore provide a means of
authentication.
IPSec is a network layer protocol essentially, a secure replacement
for traditional IP (Internet Protocol). IPSec is most suited to VPN
(virtual private network) architectures in which a set of SIP hosts
(mingled user agents and proxy servers) or bridged administrative
domains have a trust relationship with one another.
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TLS is a transport protocol and hence, like TCP and UDP, TLS can be
specified as the desired transport protocol within a Via header field
or a SIP-URI. TLS is most suited to architectures in which a chain of
trust joins together a set of hosts (e.g. Alice trusts her local
proxy server, which in turn trust Bob's local proxy server, which Bob
trusts, hence Bob and Alice can communicate securely).
TLS must be tightly coupled with a SIP application. Note that
transport mechanisms are specified on a hop-by-hop basis in SIP, and
that in some networks TLS might be used for only certain portions of
the signaling path.
It is RECOMMENDED that SIP endpoints support TLS as a secure
transport for SIP.
20.2 SIP Authentication
SIP provides a stateless challenged-based mechanism for
authentication. Any time that a proxy server or user agent receives
a request, they MAY challenge the initiator of the request to provide
assurance of their identity. Once the originator has been identified,
the recipient of the request SHOULD ascertain whether or not this
user is authorized to make the request in question. No authorization
systems are recommended or discussed in this document.
The "basic" and "digest" authentication mechanisms described in this
section provide message authentication only, without message
integrity or confidentiality. Protective measures above and beyond
authentication need to be taken to prevent active attackers from
modifying and/or replaying SIP requests and responses.
Due to its weak security, the usage of "basic" authentication is NOT
RECOMMENDED. However, servers MAY support it to handle older RFC 2543
clients that might still use it.
The framework for SIP authentication closely parallels that of HTTP
(RFC 2617 [27]). In particular, the BNF for auth- scheme, auth-param,
challenge, realm, realm-value, and credentials is identical. The 401
response is used by user agent servers in SIP to challenge the
identity of a user agent client. Additionally, registrars and
redirect servers MAY make use of 401 (Unauthorized) responses for
authentication, but proxies MUST NOT, and instead MAY use the 407
(Proxy Authentication Required) response. The requirements for
inclusion of the Proxy-Authenticate, Proxy- Authorization, WWW-
Authenticate, and Authorization in the various messages are identical
to those described in RFC 2617 [27].
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Since SIP does not have the concept of a canonical root URL, the
notion of protection spaces is interpreted differently in SIP. The
realm is a protection domain for all SIP URIs with the same value for
the userinfo, host and port part of the SIP Request-URI. For example:
INVITE sip:bob@biloxi.com SIP/2.0
WWW-Authenticate: Basic realm="business"
and
INVITE sip:robert@biloxi.com SIP/2.0
WWW-Authenticate: Basic realm="business"
Generally, SIP authentication is for a specific request Request-URI
and realm, a protection domain. Thus, for basic and digest
authentication, each such protection domain has its own set of user
names and secrets. If a user agent does not care about different
Request-URIs, it makes sense to establish a "global" user name,
secret and realm that is the default challenge if a particular
Request-URI does not have its own realm or set of user names (e.g. an
INVITE to 'sip:10.3.6.6'). Similarly, SIP entities representing many
users, such as PSTN gateways, MAY try a pre- configured global user
name and secret when challenged, independent of the Request-URI.
20.2.2 User to User Authentication
When a UAS receives a request from a UAC, the UAS MAY authenticate
the originator before the request is processed. If no credentials (in
the Authorization header field are provided in the request, the UAS
can challenge the originator to provide credentials by rejecting the
request with a 401 (Unauthorized) status code.
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages. The field value consists of at
least one challenge that indicates the authentication scheme(s) and
parameters applicable to the Request-URI. See [H14.47] for a
definition of the syntax.
An example of the WWW-Authenticate in a 401 challenge is:
WWW-Authenticate: Basic realm="business"
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When the originating UAC receives the 401 it SHOULD, if it is able,
re-originate the request with the proper credentials. The UAC may
require input from the originating user before proceeding. The
content of the "realm" parameter of the WWW-Authenticate header
SHOULD be displayed to the user. Once authentication credentials have
been supplied (either directly by the user, or discovered in a
keyring), user agents SHOULD cache the credentials for a given value
of the Request-URI and "realm" and attempt to re-use these values on
the next request for that destination.
Any user agent that wishes to authenticate itself with a UAS or
registrar -- usually, but not necessarily, after receiving a 401
response -- MAY do so by including an Authorization header field with
the request. The Authorization field value consists of credentials
containing the authentication information of the user agent for the
realm of the resource being requested.
An example of the Authorization header is:
Authorization: Basic QWxhZGRpbjpvcGVuIHNlc2FtZQ==
When a UAC resubmits a request with its credentials after receiving a
401 (or 407) response, it MUST increment the CSeq header field as it
would normally do when sending an updated request.
20.2.3 Proxy to User Authentication
Similarly, when a UAC sends a request to a proxy server, the proxy
server MAY authenticate the originator before the request is
processed. If no credentials (in the Proxy-Authorization header
field) are provided in the request, the UAS can challenge the
originator to provide credentials by rejecting the request with a 407
(Proxy Authentication Required) status code. The proxy MUST populate
the 407 (Proxy Authentication Required) message with a Proxy-
Authenticate header applicable to the proxy for the requested
resource.
The use of the Proxy-Authentication and Proxy-Authorization parallel
that described in [27], with one difference. Proxies MUST NOT add the
Proxy-Authorization header. 407 (Proxy Authentication Required)
responses MUST be forwarded upstream towards the UAC following the
procedures for any other response. It is the client's responsibility
to add the Proxy-Authorization header containing credentials for the
realm of the proxy which has asked for authentication.
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If a proxy were to resubmit a request with a Proxy-
Authorization header field, it would need to increment the
CSeq in the new request. However, this would mean that the
UAC which submitted the original request would discard a
response from the UAS, as the CSeq value would be
different.
When the originating UAC receives the 407 it SHOULD, if it is able,
re-originate the request with the proper credentials. It should
follow the same procedures for the display of the "realm" parameter
that are given above for responding to 401.
Any user agent that wishes to authenticate itself to a proxy server
-- usually, but not necessarily, after receiving a 407 response --
MAY do so by including an Proxy-Authorization header field with the
request. The Proxy-Authorization request-header field allows the
client to identify itself (or its user) to a proxy which requires
authentication. The Proxy-Authorization field value consists of
credentials containing the authentication information of the user
agent for the proxy and/or realm of the resource being requested.
A Proxy-Authorization header field applies only to the proxy whose
realm is identifier in the "realm" parameter (this proxy may
previously have demanded authentication using the Proxy-Authenticate
field). When multiple proxies are used in a chain, the Proxy-
Authorization header field MUST NOT be consumed by any proxy whose
realm does not match the "realm" parameter specified in the Proxy-
Authorization header.
Note that if an authentication scheme is used in the Proxy-
Authorization that does not support realms, a proxy server MUST
attempt to parse all Proxy-Authorization headers to determine whether
or not one of them has what it considers to be valid credentials.
Because this is potentially very time consuming in large networks,
proxy servers SHOULD use an authentication scheme that supports
realms in the Proxy-Authorization header.
It is also possible that a 401 or 407 response will contain several
challenges, from a mixture of proxies and user agent servers, if the
request was forked. If at least one user agent responds to a request
with a challenge, than a 401 should be used; otherwise a 407 should
be used. When resubmitting its request in response to the challenge,
the UAC needs to include an Authorization for each WWW-Authenticate
and Proxy- Authorization for each Proxy-Authenticate.
See [H14.34] for a definition of the syntax of Proxy- Authentication
and Proxy-Authorization.
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20.2.4 Authentication Schemes
SIP implementations MAY use HTTP's basic and digest authentication
mechanisms ([27]) to provide a rudimentary form of security. This
section overviews usage of these mechanisms in SIP. The scheme usage
is almost completely identical to that for HTTP [27]. This section
outlines this operation, pointing to RFC 2617 ([27]) for details and
noting the differences that arise when using SIP. Since RFC 2543 is
based on HTTP basic and digest as defined in RFC 2069 [28], SIP
servers supporting RFC 2617 MUST ensure they are backwards compatible
with RFC 2069. Procedures for this backwards compatibility are
specified in RFC 2617.
The rules for basic authentication follow those defined in [27] but
with the words "origin server" replaced with "user agent server,
redirect server , or registrar".
Since SIP URIs are not hierarchical, the paragraph in [27] that
states that "all paths at or deeper than the depth of the last
symbolic element in the path field of the Request-URI also are within
the protection space specified by the Basic realm value of the
current challenge" does not apply for SIP. SIP clients MAY
preemptively send the corresponding Authorization header with
requests for SIP URIs within the same protection realm (as defined
above) without receipt of another challenge from the server.
The rules for digest authentication follow those defined in [27],
with "HTTP 1.1" replaced by "SIP/2.0" in addition to the following
differences:
1. The URI included in the challenge has the following BNF:
URI = SIP-URL
2. The BNF in RFC 2617 has an error in that the URI is not
enclosed in quotation marks. (The example in Section 3.5 is
correct.) For SIP, the URI MUST be enclosed in quotation
marks.
3. The BNF for digest-uri-value is:
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digest-uri-value = Request-URI ; as defined in
Section 26
4. The example procedure for choosing a nonce based on Etag
does not work for SIP.
5. The text in RFC 2617 [27] regarding cache operation does
not apply to SIP.
6. RFC 2617 [27] requires that a server check that the URI in
the request line, and the URI included in the Authorization
header, point to the same resource. In a SIP context, these
two URI's may actually refer to different users, due to
forwarding at some proxy. Therefore, in SIP, a server MAY
check that the Request-URI in the Authorization header
corresponds to a user for whom that the server is willing
to accept forwarded or direct calls.
RFC2543 did not allow usage of the Authentication-Info header (it
effectively used RFC 2069). However, we now allow usage of this
header, since it provides integrity checks over the bodies and
provides mutual authentication. RFC2617 [27] defines mechanisms for
backwards compatibility using the qop attribute in the request. These
mechanisms MUST be used by a server to determine if the client
supports the new mechanisms in RFC 2617 that were not specified in
RFC 2069.
20.3 SIP Encryption
No mechanism is currently specified for encrypting entire SIP
messages end-to-end for the purpose of confidentiality. This is a
hard problem because network intermediaries (like proxy servers) need
to view certain headers in order to route messages correctly, and if
these intermediaries are excluded from security associations then SIP
messages will essentially be unroutable.
That much said, SIP messages carry MIME bodies and the MIME standard
includes mechanisms for securing MIME contents to ensure both
integrity and confidentiality (including the 'multipart/encrypted'
MIME type, see [29]), but detailed description of the use of secure
MIME types are outside the scope of this document. Implementors
should note, however, that there may be rare network intermediaries
(not typical proxy servers) that rely on viewing or modifying the
bodies of SIP messages (especially SDP), and that secure MIME may
prevent these sorts of intermediaries from functioning.
This applies particularly to certain types of firewalls.
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End-to-end encryption relies on keys shared by the two user agents
involved in the request. Typically, the message is sent encrypted
with the public key of the recipient, so that only that recipient can
read the message. SIP does not define any mechanism for end-to-end
key exchange.
Note that the PGP mechanism for encrypting the headers and
bodies of SIP messages described in RFC2543 has been
deprecated.
20.4 Denial of Service
Denial of service attacks focus on rendering a particular network
element unavailable, usually by directing an excessive amount of
network traffic at its interfaces. A distributed denial of service
attack allows one network user to cause multiple network hosts to
flood a target host with a large amount of network traffic.
In many architectures SIP proxy servers face the public Internet in
order to accept requests from worldwide IP endpoints. When the host
on which a SIP proxy server is operating is routable from the public
Internet, it should be deployed in an administrative domain with
secure routing policies (blocking source-routed traffic, preferably
filtering ping traffic).
SIP creates a number of potential opportunities for distributed
denial of service attacks that must be recognized and addressed by
the implementors and operators of SIP systems.
Floods of messages directed at proxy servers can lock up proxy server
resources and prevent desirable traffic from reaching its
destination. There is a computational expense associated with
processing a SIP transaction at a proxy server, and that expense is
greater for stateful proxy servers that it is for stateless proxy
servers. Therefore stateful proxies are more susceptible to flooding
than stateless proxy servers.
Attackers can create bogus requests that contain a falsified Via
header field which identifies a targeted host as the originator of
the message and then send this message to a large number of SIP
network elements, thereby using hapless SIP UAs or proxies to
generate denial of service traffic aimed at the target.
Similarly, attackers might use falsified Route headers in a request
that identify the target host and then send such messages to forking
proxies that will amplify messaging sent to the target. Record-Route
could be used to similar effect when the attacker is certain that the
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SIP dialog initiated by the request will result in numerous
transactions originating in the backwards direction.
One could prevent one's host from being commandeered for such an
attack by disallowing requests that do not make use of a persistent
security association established through a transport or network layer
security instrument such as TLS or IPsec. This could be an
appropriate security solution for two proxy servers that trust one
another and exchange significant amounts of signaling traffic with
one another, or between a user agent and its outbound proxy.
Both TLS and IPSec can also make use of bastion hosts at the edges of
administrative domains that participate in the security associations
to aggregate secure tunnels and sockets. These bastion hosts can also
take the brunt of denial of service attacks, ensuring that SIP hosts
within the administrative domain are not encumbered with superfluous
messaging.
If such a persistent security association is not feasible, user
agents and proxy servers SHOULD challenge questionable requests with
only a single 401 (Unauthorized) or 407 (Proxy Authentication
Required) forgoing the normal response retransmission algorithm.
Retransmitting the 401 or 407 status response amplifies the
problem of an attacker using a falsified header (such as
Via) to direct traffic to a third party.
A number of denial of service attacks open up if REGISTER requests
are not properly authenticated and authorized by registrars.
Attackers could de-register some or all users in an administrative
domain, thereby preventing these users from being invited to new
sessions. An attacker could also register a large number of contacts
designating the same host for a given address of record in order to
use the registrar and any associated proxy servers as amplifiers in a
denial of service attack. Attackers might also attempt to deplete
available memory and disk resources of a registrar by registering
huge numbers of bindings.
With either TCP or UDP, a denial of service attack exists by a rogue
proxy sending 6xx responses. Although a client SHOULD choose to
ignore such responses if it requested authentication, a proxy cannot
do so. It is obliged to forward the 6xx response back to the client.
The client can then ignore the response, but if it repeats the
request it will probably reach the same rogue proxy again, and the
process will repeat.
The use of multicast to transmit SIP requests can greatly increase
the potential for denial of service attacks.
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21 Common Message Components
There are certain components of SIP messages that appear in various
places within SIP messages (and sometimes, outside of them), which
merit separate discussion.
21.1 SIP Uniform Resource Locators
A SIP URL identifies a communications resource. Like all URLs, SIP
URLs may be placed in web pages, email messages or printed
literature. They contain sufficient information to initiate and
maintain a communication session with the resource.
Examples of communications resources include
o a user of an online service
o an appearance on a multiline phone
o a mailbox on a messaging system
o a PSTN phone number at a gateway service
o a group (such as "sales" or "helpdesk") in an organization
21.1.1 SIP URL components
The "sip:" scheme follows the guidelines in RFC 2396 [9]. It uses a
form similar to the mailto URL, allowing the specification of SIP
request-header fields and the SIP message- body. This makes it
possible to specify the subject, media type, or urgency of sessions
initiated by using a URL on a web page or in an email message. The
formal syntax for a SIP URL is presented in Section 26. Its general
form is
sip:user:password@host:port;url-parameters?headers
These tokens, and some of the tokens in their expansion, have the
following meanings.
user: The identifier of a particular resource at the host being
addressed.
Note that "host" as used here may, and frequently does,
refer to a domain.
The "userpart" of a URL consists of this user field, the
password field and the @ sign following them. The userpart
of a URL is optional and MAY be absent when the destination
host does not have a notion of users or when the host
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itself is the resource being identified. If the @ sign is
present in a SIP URL, the user field MUST NOT be empty.
If the host being addressed is capable of processing
telephone numbers, an Internet telephony gateway for
instance, a telephone- subscriber field defined in RFC 2806
[13] MAY be used to populate the user field. There are
special escaping rules for encoding telephone-subscriber
fields in SIP URLs described in Section 21.1.2.
password: A password associated with the user
While the SIP URL syntax allows this field to be present,
its use is NOT RECOMMENDED, because the passing of
authentication information in clear text (such as URIs) has
proven to be a security risk in almost every case where it
has been used. For instance, transporting a PIN number in
this field exposes the PIN.
host: The entity hosting the SIP resource
The host part contains either a fully-qualified domain name
or numeric IPv4 or IPv6 address. Using the fully-qualified
domain name form is RECOMMENDED whenever possible.
port: The port number where the request is to be sent.
URL parameters: Parameters affecting a request constructed from
the URL.
URL parameters are added after the hostport component and
are separated by semi-colons. This extensible mechanism
includes the transport, maddr, ttl, user, and method
parameters.
The transport parameter determines the transport mechanism
to be used for sending SIP messages. SIP can use any
network transport protocol. Parameter names are defined
for UDP [30], TCP [31], TLS [25], and SCTP [32].
The maddr parameter indicates the server address to be
contacted for this user, overriding any address derived
from the host field. Section 24 describes the proper
interpretation of the transport, maddr and hostport in
order to obtain the destination address, port and transport
for sending a request.
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The maddr field can be used as a simple form of loose
source routing. It allows a URL to specify a specific
proxy that must be traversed en-route to the
destination. This capability is useful for a roaming
user that is forced to use an outbound proxy, but
wishes to force requests through their home proxy.
The ttl parameter determines the time-to-live value of the
UDP multicast packet and MUST only be used if maddr is a
multicast address and the transport protocol is UDP. The
user parameter was described above. For example, to specify
to call alice@atlanta.com using multicast to 239.255.255.1
with a ttl of 15, the following URL would be used:
sip:alice@atlanta.com;maddr=239.255.255.1;ttl=15
The set of valid telephone-subscriber strings is a subset
of valid user strings. The user URL parameter exists to
distinguish telephone numbers from user names that happen
to look like telephone numbers. If the user string
contains a telephone number formatted as a telephone-
subscriber, the user parameter value "phone" SHOULD be
present. Even without this parameter, recipients of SIP
URLs MAY interpret the pre-@ part as a telephone number if
local restrictions on the name space for user name allow
it.
The method of the SIP request constructed from the URL can
be specified with the method parameter.
Since the url-parameter mechanism is extensible, SIP
elements MUST silently ignore any url-parameters that they
do not understand.
Headers: Headers to be included in a request constructed from
the URL.
Headers fields in the SIP request can be specified with the
"?" mechanism within a SIP URL. The header names and values
are encoded in ampersand separated hname = hvalue pairs.
The special hname "body" indicates that the associated
hvalue is the message-body of the SIP request.
Table 1 summarizes the use of SIP URL components based on the context
in which the URL appears. The external column describes URLs
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appearing anywhere outside of a SIP message, for instance on a web
page or business card. Entries marked "m" are mandatory, those
marked "o" are optional, and those marked "-" are not allowed.
Elements processing URLs SHOULD ignore any disallowed components if
they are present. The second column indicates the default value of
an optional element if it is not present. "--" indicates that the
element is either not optional, or has no default value.
SIP URLs in Contact header fields have different restrictions
depending on the context in which the header field appears. One set
applies to messages that establish and maintain dialogs (INVITE and
its 200 OK response). The other applies to registration and
redirection messages (REGISTER, its 200 OK response, and 3xx class
responses to any method).
OPEN ISSUE #203: maddr is disallowed in To/From, but not port. Should
port be disallowed?
OPEN ISSUE #204: Password is disallowed in From, but not To. Why?
OPEN ISSUE #205: Should we allow method and header URL components in
registration/redirect Contacts. What do they mean?
dialog
reg./redir. Contact/
default Req.-URI To From Contact R-R/Route external
user -- o o o o o o
password -- o o - o o o
host -- m m m m m m
port 5060 o o o o o o
user-param ip o o o o o o
method INVITE - - - o - o
maddr-param -- o - - o o o
ttl-param 1 o - - o - o
transp.-param udp o - - o o o
other-param -- o o o o o o
headers -- - - - o - o
Table 1: Use and default values of URL components for SIP headers,
Request-URI and references
21.1.2 Character escaping requirements
SIP follows the requirements and guidelines of RFC 2396 when defining
the set of characters that must be escaped in a SIP URL, and uses its
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""%" HEX HEX" mechanism for escaping. From RFC 2396:
The set of characters actually reserved within any given
URI component is defined by that component. In general, a
character is reserved if the semantics of the URI changes
if the character is replaced with its escaped US-ASCII
encoding. [9]. Excluded US-ASCII characters [9], such as
space and control characters and characters used as URL
delimiters, also MUST be escaped. URLs MUST NOT contain
unescaped space and control characters.
For each component, the set of valid BNF expansions defines exactly
which characters may appear unescaped. All other characters MUST be
escaped.
For example, "@" is not in the set of characters in the user
component, so the user "j@s0n" must have at least the @ sign encoded,
as in "j%40s0n".
Expanding the hname and hvalue tokens in Section 26 show that all URL
reserved characters in header names and values MUST be escaped.
The telephone-subscriber subset of the user component has special
escaping considerations. The set of characters not reserved in the
RFC 2806 [13] description of telephone-subscriber contains a number
of characters in various syntax elements that need to be escaped when
used in SIP URLs. Any characters occurring in a telephone-subscriber
that do not appear in an expansion of the BNF for the user rule MUST
be escaped.
21.1.3 Example SIP URLs
sip:alice@atlanta.com
sip:alice:secretword@atlanta.com;transport=tcp
sip:alice@atlanta.com?subject=project
sip:+1-212-555-1212:1234@gateway.com;user=phone
sip:1212@gateway.com
sip:alice@10.1.1.1
sip:atlanta.com;method=REGISTER?to=alice
sip:alice;day=tuesday@atlanta.com
The last example URL above has a user field value of
"alice;day=tuesday". The escaping rules defined above allow a
semicolon to appear unescaped in this field. Note, however, that for
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the purposes of this protocol, the field is opaque. The apparent
structure in that value is only useful to the entity responsible for
the resource.
21.1.4 SIP URL Comparison
SIP URLs are compared for equality according to the following rules:
o Comparisons of scheme name ("sip"), domain names, parameter
names and header names are case-insensitive, all other
comparisons are case-sensitive. (OPEN ISSUE #100 : There is a
proposal to make only quoted string comparisons case-
sensitive.)
o The ordering of parameters and headers is not significant in
comparing SIP URLs.
o Characters other than those in the "reserved" and "unsafe"
sets (see RFC 2396 [9]) are equivalent to their ""%" HEX HEX"
encoding.
o An IP address that is the result of a DNS lookup of a host
name does not match that host name.
o For two URLs to be equal, the user, password, host, and port
components must match. A URL omitting the optional port
component will match a URL explicitly declaring port 5060. A
URL omitting the user component will not match a URL that
includes one. A URL omitting the password component will not
match a URL that includes one.
o URL url-parameter components are compared as follows
- Any url-parameter appearing in both URLs must match.
- A user, transport, ttl, or method url-parameter appearing in
only one URL must contain its default value or the URLs do
not match.
- All other url-parameters appearing in only one URL are
ignored when comparing the URLs.
o URL header components are never ignored. Any present header
component MUST be present in both URLs and match for the URLs
to match. The matching rules are defined for each header in
Section sec:header-fields.
The URLs within each of the following sets are equivalent:
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sip:alice@%61tlanta.com
sip:alice@AtLanTa.CoM;Transport=udp
sip:carol@chicago.com
sip:carol@chicago.com;newparam=5
sip:carol@chicago.com;security=on
sip:biloxi.com;transport=tcp;method=REGISTER?to=sip:bob
sip:biloxi.com;method=REGISTER;transport=tcp?to=sip:bob
sip:alice@atlanta.com?subject=project
sip:alice@atlanta.com?priority=urgent&subject=project
The URLs within each of the following sets are not equivalent:
SIP:ALICE@AtLanTa.CoM;Transport=udp (different usernames)
sip:alice@AtLanTa.CoM;Transport=UDP
sip:bob@biloxi.com (different port and transport)
sip:bob@biloxi.com:6000;transport=tcp
sip:carol@chicago.com (different header component)
sip:carol@chicago.com?Subject=next
sip:bob@phone21.boxesbybob.com (even though that's what
sip:bob@10.4.1.4 phone21.boxesbybob.com resolves to)
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Note that equality is not transitive:
o sip:carol@chicago.com and sip:carol@chicago.com;security=on
are equivalent
o sip:carol@chicago.com and sip:carol@chicago.com;security=off
are equivalent
o sip:carol@chicago.com;security=on and
sip:carol@chicago.com;security=off are not equivalent
Comparing URLs is a major part of comparing several SIP headers (see
Section 22).
21.2 Option Tags
Option tags are unique identifiers used to designate new options
(extensions) in SIP. These tags are used in Require (Section 22.30),
Proxy-Require (Section 22.28, Supported (Section 22.35) and
Unsupported (Section 22.38) header fields. Note that these options
appear as parameters in those headers in an option-tag = token form
(see Section 26 for the definition of token).
The creator of a new SIP option MUST either prefix the option with
their reverse domain name or register the new option with the
Internet Assigned Numbers Authority (IANA) (See Section 27).
An example of a reverse-domain-name option is "com.foo.mynewfeature",
whose inventor can be reached at "foo.com". For these features,
individual organizations are responsible for ensuring that option
names do not collide within the same domain. The host name part of
the option MUST use lower-case; the option name is case-sensitive.
Options registered with IANA do not contain periods and are globally
unique. IANA option tags are case-sensitive.
The "tag" parameter is used in the To and From fields of SIP
messages. It serves as a general mechanism to identify a particular
instance of a user agent for a particular SIP URI.
As proxies can fork requests, the same request can reach multiple
instances of a user (mobile and home phones, for example). Since each
can respond, there needs to be a means for the originator of a
session to distinguish the responses. Tag fields in the To and From
disambiguate these multiple instances of the same user.
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This situation also arises with multicast requests.
When a tag is generated by a UA for insertion into a request or
response, it MUST be globally unique and cryptographically random
with at least 32 bits of randomness. A property of this selection
requirement is that a UA will place a different tag into the From
header of an INVITE as it would place into the To header of the
response to the same INVITE. This is needed in order for a UA to
invite itself to a session, a common case for "hairpinning" of calls
in PSTN gateways.
Besides the requirement for global uniqueness, the algorithm for
generating a tag is implementation specific. Tags are helpful in
fault tolerant systems, where a dialog is to be recovered on an
alternate server after a failure. A UAS can select the tag in such a
way that a backup can recognize a request as part of a dialog on the
failed server, and therefore determine that it should attempt to
recover the dialog and any other state associated with it.
22 Header Fields
The general syntax for header fields is covered in Section 7.3. This
section lists the full set of header fields along with notes on
syntax, meaning, and usage. Throughout this section, we use [HX.Y] to
refer to Section X.Y of the current HTTP/1.1 specification RFC 2617
[27]. Examples of each header field are given.
Information about header fields in relation to methods and proxy
processing is summarized in Tables 2 and 3.
The "where" column describes the request and response types in which
the header field can be used. Values in this column are:
R: refers to header fields that can be used in requests.
r: designates a header field as applicable to all responses,
while a list of numeric values indicates the status codes
with which the header field can be used.
c: indicates a header field is copied from the request to the
response.
The "proxy" column describes the operations a proxy may perform on a
header.
c: indicates that a proxy can add (concatenate) comma-separated
elements to the header
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m: indicates that a proxy can modify the header
a: indicates that a proxy can add the header if not present
r: indicates that a proxy must be be able to read the header.
Headers that need to be read cannot be encrypted.
The next six columns relate to the presence of a header field in a
method, with the contents indicating:
o: for optional
m: for mandatory
m*: indicates a header that SHOULD be sent, but servers need to
be prepared to receive messages without that header field.
*: indicates that the header fields are required if the message
body is not empty. See sections 22.14, 22.15 and 7.4 for
details.
-: for not applicable.
"Optional" means thata UA MAY include the header field in a request
or response, and a UA MAY ignore the header field if present in the
request or response (The exception to this rule is the Require header
field discussed in 22.30). A "mandatory" header field MUST be present
in a request, and MUST be understood by the UAS receiving the
request. A mandatory response header field MUST be present in the
response, and the header field MUST be understood by the UAC
processing the response. "Not applicable" means for header fields
that the header field MUST NOT be present in a request. If one is
placed in a request by mistake, it MUST be ignored by the UAS
receiving the request. Similarly, a header field labeled "not
applicable" for a response means that the UAS MUST NOT place the
header in the response, and the UAC MUST ignore the header in the
response.
A compact form of some common header fields is also defined for use
when overall message size is an issue.
The Contact, From and To header fields contain a URL. If the URL
contains a comma, question mark or semicolon, the URL MUST be
enclosed in angle brackets (< and >). Any URL parameters are
contained within these brackets. If the URL is not enclosed in angle
brackets, any semicolon-delimited parameters are header-parameters,
not URL parameters.
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Header field where proxy ACK BYE CAN INV OPT REG
____________________________________________________________
Accept R - o - m* o o
Accept 2xx - - - m* o o
Accept 415 - o - o o o
Accept-Encoding R - o - m* o o
Accept-Encoding 2xx - - - m* o o
Accept-Encoding 415 - o - o o o
Accept-Language R - o - m* o o
Accept-Language 2xx - - - m* o o
Accept-Language 415 - o - o o o
Alert-Info R am - - - o - -
Alert-Info 180 am - - - o - -
Allow R o o o o o o
Allow 2xx - o o m* m* o
Allow r - o o o o o
Allow 405 - m m m m m
Authentication-Info 2xx - o - o o o
Authorization R o o o o o o
Call-ID c r m m m m m m
Call-Info am - - - o o o
Contact R o - - m o o
Contact 1xx - - - o o -
Contact 2xx - - - m o o
Contact 3xx - o - o o o
Contact 485 - o - o o o
Content-Disposition o o - o o o
Content-Encoding o o - o o o
Content-Language o o - o o o
Content-Length r m* m* m* m* m* m*
Content-Type * * - * * *
CSeq c r m m m m m m
Date a o o o o o o
Error-Info 300-699 - o o o o o
Expires - - - o - o
From c r m m m m m m
In-Reply-To R - - - o - -
Max-Forwards R rm o o o o o o
MIME-Version o o o o o o
Organization am - - - o o o
Table 2: Summary of header fields, A--O
The Accept header follows the syntax defined in [H14.1]. The
semantics are also identical, with the exception that if no Accept
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Header field where proxy ACK BYE CAN INV OPT REG
___________________________________________________________________
Priority R a - - - o - -
Proxy-Authenticate 407 - m m m m m
Proxy-Authorization R r o o o o o o
Proxy-Require R r o o o o o o
Record-Route R amr o o o o o o
Record-Route 2xx,401,484 - o o o o o
Require g acr o o o o o o
Retry-After 404,413,480,486 - o o o o o
500,503 - o o o o o
600,603 - o o o o o
Route R r o o o o o o
Server r - o o o o o
Subject R - - - o - -
Supported - o o o o o
Timestamp o o o o o o
To gc(1) r m m m m m m
Unsupported 420 - o o o o o
User-Agent o o o o o o
Via c acmr m m m m m m
Warning r o o o o o o
WWW-Authenticate 401 - m m m m m
Table 3: Summary of header fields, P--Z; (1): copied with possible
addition of tag
header is present, the server SHOULD assume a default value of
application/sdp
Example:
Accept: application/sdp;level=1, application/x-private, text/html
22.2 Accept-Encoding
The Accept-Encoding header field is similar to Accept, but restricts
the content-codings [H3.5] that are acceptable in the response. See
[H14.3]. The syntax of this header is defined in [H14.3]. The
semantics in SIP are identical to those defined in [H14.3].
An empty Accept-Encoding header field is permissible, even though the
syntax in [H14.3] does not provide for it. It is equivalent to
Accept-Encoding: identity, i.e., only the identity encoding, meaning
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no encoding, is permissible. If this header is not present, the
default value is identity. This differs slightly from the HTTP
definition, which indicates that when not present, any encoding can
be used, but the identity encoding is preferred.
Example:
Accept-Encoding: gzip
22.3 Accept-Language
The Accept-Language header follows the syntax defined in [H14.4]. The
rules for ordering the languages based on the "q" parameter apply to
SIP as well.
The Accept-Language header is used in requests to indicate the
preferred languages for reason phrases, session descriptions or
status responses carried as message bodies in the response. If no
Accept-Language header field is present in a request, the server
assumes all languages are acceptable to the client.
Example:
Accept-Language: da, en-gb;q=0.8, en;q=0.7
22.4 Alert-Info
When present in an INVITE request, the Alert-Info header field
specifies an alternative ring tone to the UAS. When present in a 180
(Ringing) response, the Alert-Info header field specifies an
alternative ringback tone to the UAC. A typical usage is for a proxy
to insert this header to provide a distinctive ring feature.
The Alert-Info header can introduce security risks. These risks, and
the ways to handle them, are discussed in Section 22.9 which
discusses the Call-Info header, as the risks are identical.
In addition, a user SHOULD be able to disable this feature
selectively.
This helps prevent disruptions that could result from the
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use of this header by untrusted elements.
Example:
Alert-Info: <http://wwww.example.com/sounds/moo.wav>
The Allow header field lists the set of methods supported by the user
agent generating the message.
All methods, including ACK and CANCEL, understood by the UA MUST be
included in the list of methods in the Allow header, when present.
The absence of an Allow header MUST NOT be interpreted to mean that
the UA sending the message supports no methods. Rather, it implies
that the UA is not providing any information on what methods it
supports.
Supplying an Allow header in responses to methods other than OPTIONS
cuts down on the number of messages needed.
Example:
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE
22.6 Authentication-Info
The Authentication-Info header provides for mutual authentication
with HTTP Digest. A UAS MAY include this header in a 2xx response to
a request that was successfully authenticated using digest based on
the Authorization header.
Syntax and semantics follow those specified in RFC2617 [27].
Example:
Authentication-Info: nextnonce="47364c23432d2e131a5fb210812c"
22.7 Authorization
The Authorization header field contains authentication credentials of
a UA. Section 20.2.2 overviews the use of the Authorization header
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field, and Section 20.2.4 describes the syntax and semantics when
used with HTTP Basic and Digest authentication.
Note that this header field, along with Proxy-Authorization breaks
the general rules about multiple header fields. Although not a
comma-separated list, this header field may be present multiple
times, and MUST NOT be combined into a single header using the usual
rules described in Section 7.3.
Example:
Authorization: Digest username="Alice", realm="Bob's Friends",
nonce="84a4cc6f3082121f32b42a2187831a9e",
response="7587245234b3434cc3412213e5f113a5432"
The Call-ID header field uniquely identifies a particular invitation
or all registrations of a particular client. Note that a single
multimedia conference can give rise to several calls with different
Call-IDs, e.g., if a user invites a single individual several times
to the same (long-running) conference.Call-IDs are case- sensitive
and are simply compared byte-by-byte.
The compact form of the Call-IDheader field is i.
Examples:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@biloxi.com
i:f81d4fae-7dec-11d0-a765-00a0c91e6bf6@10.4.1.4
22.9 Call-Info
The Call-Info header field provides additional information about the
caller or callee, depending on whether it is found in a request or
response. The purpose of the URI is described by the "purpose"
parameter. "icon" designates an image suitable as an iconic
representation of the caller or callee; "info" describes the caller
or callee in general, e.g., through a web page; "card" provides a
business card (e.g., in vCard [33] or LDIF [34] formats). Additonal
tokens can be registered using IANA and the procedures in Section 27.
Usage of the Call-Info header can pose a security risk. If a callee
fetches the URLs provided by an malicious caller, the callee may be
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at risk for displaying inappropriate or offensive content, dangerous
or illegal content, and so on. Therefore, it is RECOMMENDED that a UA
only render the information in the Call-Info header if it can verify
the authenticity of the element which originated the header, and
trusts that element. This need not be the peer UA; a proxy can insert
this header into requests.
The use of this header is important in converged applications.
Example:
Call-Info: <http://wwww.example.com/alice/photo.jpg> ;purpose=icon,
<http://www.example.com/alice/> ;purpose=info
The Contact header field provides a URL whose meaning depends on the
the type of request or response it is in.
Parameters defined for Contact include "q" and "expires". Additional
parameters may be defined in other specifications.Even if the
"display-name" is empty, the "name-addr" form MUST be used if the
"addr-spec" contains a comma, semicolon or question mark. Note that
there may or may not be LWS between the display-name and the "<".
The Contact header field fulfills functionality similar to
the Location header field in HTTP. However, the HTTP header
only allows one address, unquoted. Since URIs can contain
commas and semicolons as reserved characters, they can be
mistaken for header or parameter delimiters, respectively.
The current syntax corresponds to that for the To and From
header, which also allows the use of display names.
The compact form of the Contact header field is m (for "moved").
Examples:
Contact: "Mr. Watson" <sip:watson@worcester.bell-telephone.com>
;q=0.7; expires=3600,
"Mr. Watson" <mailto:watson@bell-telephone.com> ;q=0.1
m: <sip:bob@10.5.1.5>
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22.11 Content-Disposition
The Content-Disposition header field describes how the message body
or, in the case of multipart messages, a message body part is to be
interpreted by the UAC or UAS. The SIP header extends the MIME
Content-Type (RFC 1806 [35]).
The value "session" indicates that the body part describes a session,
for either calls or early (pre-call) media. The value "render"
indicates that the body part should be displayed or otherwise
rendered to the user. For backward-compatibility, if the Content-
Disposition header is not missing, bodies of Content-Type
application/sdp imply the disposition "session", while other content
types imply "render".
The disposition type "icon" indicates that the body part contains an
image suitable as an iconic representation of the caller or callee.
The value "alert" indicates that the body part contains information,
such as an audio clip, that should be rendered instead of ring tone.
The handling parameter, handling-parm, describes how the UAS should
react if it receives a message body whose content type or disposition
type it does not understand. The parameter has defined values of
"optional" and "required". If the handling parameter is missing, the
value "required" is to be assumed. If this header field is missing,
the MIME type determines the default content disposition. If there is
none, "render" is assumed.
Example:
Content-Disposition: session
22.12 Content-Encoding
The Content-Encoding header field is used as a modifier to the
"media-type". When present, its value indicates what additional
content codings have been applied to the entity-body, and thus what
decoding mechanisms MUST be applied in order to obtain the media-type
referenced by the Content-Type header field. Content-Encoding is
primarily used to allow a body to be compressed without losing the
identity of its underlying media type.
If multiple encodings have been applied to an entity, the content
codings MUST be listed in the order in which they were applied.
All content-coding values are case-insensitive. The Internet Assigned
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Numbers Authority (IANA) acts as a registry for content-coding value
tokens. See [H3.5] for a definition of the syntax for content-coding.
Clients MAY apply content encodings to the body in requests. A server
MAY apply content encodings to the bodies in responses. The server
MUST only use encodings listed in the Accept-Encoding header in the
request.
The compact form of the Content-Encoding header field is e.
Examples:
Content-Encoding: gzip
e: tar
22.13 Content-Language
See [H14.12].
Example:
Content-Language: fr
22.14 Content-Length
The Content-Length header field indicates the size of the message-
body, in decimal number of octets, sent to the recipient.
Applications SHOULD use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
entity. (The size of the message-body does not include the CRLF
separating headers and body.) Any Content-Length greater than or
equal to zero is a valid value. If no body is present in a message,
then the Content-Length header field MUST be set to zero.
The ability to omit Content-Length simplifies the creation
of cgi-like scripts that dynamically generate responses.
The short form of the header is l.
Examples:
Content-Length: 349
l: 173
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22.15 Content-Type
The Content-Type header field indicates the media type of the
message-body sent to the recipient. The "media-type" element is
defined in [H3.7]. The Content-Type header MUST be present if the
body is not empty. If the body is empty, and a Content-Length header
is present, it indicates that the body of the specific type has zero
length (for example, if it is an emtpy audio file).
The short form of the header is c.
Examples:
Content-Type: application/sdp
c: text/html; charset=ISO-8859-4
A CSeq header field in a request contains a single decimal sequence
number and the request method. The sequence number MUST be
expressible as a 32-bit unsigned integer. The CSeq header serves to
order transactions within a dialog, and to provide a means to
uniquely identify transactions, and to differentiate between new
requests and request retransmissions.
Example:
CSeq: 4711 INVITE
The Date header field contains an RFC 1123 date (see [H14.18]). Note
that unlike HTTP/1.1, SIP only supports the most recent RFC 1123 [36]
formatting for dates. As in [H3.3], SIP restricts the timezone in
SIP-date to "GMT", while RFC 1123 allows any timezone.
The consistent use of GMT between Date, Expires and Retry-
After headers allows implementation of simple clients that
do not have a notion of absolute time. Note that rfc1123-
date is case-sensitive.
The Date header field reflects the time when the request or response
is first sent.
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The Date header field can be used by simple end systems
without a battery-backed clock to acquire a notion of
current time. However, in its GMT-form, it requires clients
to know their offset from GMT.
Example:
Date: Sat, 13 Nov 2001 23:29:00 GMT
22.18 Error-Info
The Error-Info header field provides a pointer to additional
information about the error status response.
SIP UACs have user interface capabilities ranging from pop
up windows and audio on PC softclients to audio-only on
"black" phones or endpoints connected via gateways. Rather
than forcing a server generating an error to choose between
sending an error status code with a detailed reason phrase
and playing an audio recording, the Error-Info header field
allows both to be sent. The UAC then has the choice of
which error indicator to render to the caller.
A UAC MAY treat a SIP URL in an Error-Info header field as if it were
a Contact in a redirect and generate a new INVITE, resulting an a
recorded announcement session being established. A non-SIP URL MAY be
rendered to the user.
Examples:
SIP/2.0 404 The number you have dialed is not in service
Error-Info: <sip:not-in-service-recording@atlanta.com>
The Expires header field gives the date and time after which the
message (or content) expires. The precise meaning of this is method
dependent.
Note that the expiration time in an INVITE does not affect the
duration of the actual session that may result from the invitation.
Session description protocols may offer the ability to express time
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limits on the session duration, however.
The value of this field can be either a date (see the Date header
field) or an integer number of seconds (in decimal), measured from
the receipt of the request. The latter approach is preferable for
short durations, as it does not depend on clients and servers sharing
a synchronized clock.
Examples:
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
The From header field indicates the initiator of the request. (Note
that this may be different from the initiator of the dialog. Requests
sent by the callee to the caller use the callee's address in the From
header field.)
The optional "display-name" is meant to be rendered by a human user
interface. A system SHOULD use the display name "Anonymous" if the
identity of the client is to remain hidden.
Even if the "display-name" is empty, the "name-addr" form MUST be
used if the "addr-spec" contains a comma, question mark, or
semicolon. Syntax issues are discussed in Section 7.3.1.
The short form of the header is f.
Examples:
From: "A. G. Bell" <sip:agb@bell-telephone.com> ;tag=a48s
From: sip:+12125551212@server.phone2net.com;tag=887s
f: Anonymous <sip:c8oqz84zk7z@privacy.org>;tag=hyh8
22.21 In-Reply-To
The In-Reply-To header field enumerates the Call-IDs that this call
references or returns. These Call-IDs may have been cached by the
client then included in this header in a return call.
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This allows automatic call distribution systems to route
return calls to the originator of the first call and allows
callees to filter calls, so that only calls that return
calls they have originated will be accepted. This field is
not a substitute for request authentication.
Example:
In-Reply-To: 70710@saturn.bell-tel.com, 17320@saturn.bell-tel.com
22.22 Max-Forwards
The Max-Forwards header field may be used with any SIP method to
limit the number of proxies or gateways that can forward the request
to the next downstream server. This can also be useful when the
client is attempting to trace a request chain which appears to be
failing or looping in mid-chain.
The Max-Forwards value is a decimal integer indicating the remaining
number of times this request message is allowed to be forwarded. This
count is decremented by each server that forwards the request.
Example:
Max-Forwards: 6
22.23 MIME-Version
See [H19.4.1].
Example:
MIME-Version: 1.0
22.24 Organization
The Organization header field conveys the name of the organization to
which the entity issuing the request or response belongs.
The field MAY be used by client software to filter calls.
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Example:
Organization: Boxes by Bob
The Priority header field indicates the urgency of the request as
perceived by the client. Defined values include "non-urgent",
"normal", "urgent", and "emergency".
It is RECOMMENDED that the value of "emergency" only be used when
life, limb or property are in imminent danger. Otherwise, there are
no semantics defined for this header field.
These are the values of RFC 2076 [37], with the addition of
"emergency".
Examples:
Subject: A tornado is heading our way!
Priority: emergency
or
Subject: Weekend plans
Priority: non-urgent
22.26 Proxy-Authenticate
The Proxy-Authenticate header field consists of a challenge that
indicates the authentication scheme and parameters applicable to the
proxy for this Request-URI.
The syntax for this header and use is defined in [H14.33]. See 20.2.3
for further details on its usage. Example:
Proxy-Authenticate: Digest realm="Carrier SIP",
domain="sip:ss1.carrier.com",
nonce="f84f1cec41e6cbe5aea9c8e88d359",
opaque="", stale=FALSE, algorithm=MD5
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22.27 Proxy-Authorization
The Proxy-Authorization header field allows the client to identify
itself (or its user) to a proxy which requires authentication. The
Proxy-Authorization field value consists of credentials containing
the authentication information of the user agent for the proxy and/or
realm of the resource being requested.
See [H14.34] for a definition of the syntax, and section 20.2.3 for a
discussion of its usage.
Note that this header field, along with Authorization breaks the
general rules about multiple header fields. Although not a comma-
separated list, this header field may be present multiple times, and
MUST NOT be combined into a single header using the usual rules
described in Section 7.3.1.
Example:
Proxy-Authorization: Digest username="Alice", realm="Atlanta ISP",
nonce="c60f3082ee1212b402a21831ae",
response="245f23415f11432b3434341c022"
22.28 Proxy-Require
The Proxy-Require header field is used to indicate proxy-sensitive
features that must be supported by the proxy. See Section 22.30 for
more details on the mechanics of this message and a usage example.
Example:
Proxy-Require: foo
22.29 Record-Route
The Record-Route is inserted by proxies in a request to force future
requests in the session to route through the proxy.
Details of its use with the Route header field are described in
Section 16.4.
Example:
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Record-Route: <sip:bob@biloxi.com;maddr=10.1.1.1>,
<sip:bob@biloxi.com;maddr=10.2.1.1>
The Require header field is used by clients to tell user agent
servers about options that the client expects the server to support
in order to properly process the request. Although an optional
header, the Require MUST NOT be ignored if it is present.
This is to make sure that the client-server interaction
will proceed without delay when all options are understood
by both sides, and only slow down if options are not
understood (as in the example above). For a well-matched
client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation
mechanisms. In addition, it also removes ambiguity when the
client requires features that the server does not
understand. Some features, such as call handling fields,
are only of interest to end systems.
Example:
Require: com.example.billing
22.31 Retry-After
The Retry-After header field can be used with a 503 (Service
Unavailable) response to indicate how long the service is expected to
be unavailable to the requesting client and with a 404 (Not Found),
600 (Busy), or 603 (Decline) response to indicate when the called
party anticipates being available again. The value of this field can
be either an SIP-date or an integer number of seconds (in decimal)
after the time of the response.
An optional comment can be used to indicate additional information
about the time of callback. An optional "duration" parameter
indicates how long the called party will be reachable starting at the
initial time of availability. If no duration parameter is given, the
service is assumed to be available indefinitely.
Examples:
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Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Retry-After: Mon, 01 Jan 9999 00:00:00 GMT
(Dear John: Don't call me back, ever)
Retry-After: Fri, 26 Sep 1997 21:00:00 GMT;duration=3600
Retry-After: 120
In the third example, the callee is reachable for one hour starting
at 21:00 GMT. In the last example, the delay is 2 minutes.
The Route is used to force routing for a request through the listed
set of proxies. Details of its use with the Record-Route header field
are described in Section 13.
Example:
Route: <sip:bob@biloxi.com;maddr=10.1.1.1>, <sip:bob@10.4.1.4>
The Server header field contains information about the software used
by the user agent server to handle the request. The syntax for this
field is defined in [H14.38].
Example:
Server: HomeProxy v2
This header field provides a summary or indicates the nature of the
call, allowing call filtering without having to parse the session
description. (Note that the session description does not have to use
the same subject indication as the invitation.)
The short form of the header is s.
Example:
Subject: Need more boxes
s: Tech Support
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The Supported header field enumerates all the extensions upported by
the client or server. If empty, it means that no extensions are
supported.
Example:
Supported: foo, bar
The Timestamp header field describes when the client sent the request
to the server. The use of the Timestamp is covered in Section 13.
Example:
Timestamp: 54
The To header field specifies the logical recipient of the request.
The optional "display-name" is meant to be rendered by a human-user
interface. The "tag" parameter serves as a general mechanism to
distinguish multiple instances of a user identified by a single SIP
URL.
See Section 13 for details of the "tag" parameter.
Section 22.20 describes how To and From header fields are compared
for the purpose of matching requests to dialogs. Even if the
"display-name" is empty, the "name-addr" form MUST be used if the
"addr-spec" contains a comma, question mark, or semicolon. Note that
LWS is common, but not mandatory between the display-name and the
"<".
The short form of the header is t.
The following are examples of valid To headers:
To: The Operator <sip:operator@cs.columbia.edu>;tag=287447
t: sip:+12125551212@server.phone2net.com
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22.38 Unsupported
The Unsupported header field lists the features not supported by the
server. See Section 22.30 for a usage example and motivation.
Example:
Unsupported: foo
22.39 User-Agent
The User-Agent header field contains information about the client
user agent originating the request. The syntax and semantics are
defined in [H14.43].
Example:
User-Agent: Softphone Beta1.5
The Via field indicates the path taken by the request so far and
indicate the path that should be followed in routing responses.
The Via header field contains the transport protocol used to send the
message, the client's host name or network address and, if not the
default port number, the port number at which it wishes to receive
responses. The Via header field can also contains parameters such as
"maddr", "ttl", "received", and "branch"whose meaning and use are
described in other sections.
The short form of the header is v.
Example:
Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
Via: SIP/2.0/UDP 128.59.16.1:5060 ;received=128.59.19.3
In this example, the message originated from a multi-homed host with
two addresses, 128.59.16.1 and 128.59.19.3. The sender guessed wrong
as to which network interface would be used. Erlang.bell-
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telephone.com noticed the mismatch, and added a parameter to the
previous hop's Via header field, containing the address that the
packet actually came from.
Another example:
Via: SIP/2.0/UDP first.example.com:4000;ttl=16
;maddr=224.2.0.1 ;branch=a7c6a8dlze.1
The Warning header field is used to carry additional information
about the status of a response. Warning headers are sent with
responses and contain a three digit warning code, host name, and
warning text.
The "warn-text" should be in a natural language that is most likely
to be intelligible to the human user receiving the response. This
decision can be based on any available knowledge, such as the
location of the cache or user, the Accept-Language field in a
request, or the Content-Language field in a response. The default
language is i-default [38].
The first digit of warning codes beginning with "3" indicates
warnings specific to SIP.
This is a list of the currently-defined "warn-code"s, each with a
recommended warn-text in English, and a description of its meaning.
Note that these warnings describe failures induced by the session
description.
Warnings 300 through 329 are reserved for indicating problems with
keywords in the session description, 330 through 339 are warnings
related to basic network services requested in the session
description, 370 through 379 are warnings related to quantitative QoS
parameters requested in the session description, and 390 through 399
are miscellaneous warnings that do not fall into one of the above
categories.
300 Incompatible network protocol: One or more network protocols
contained in the session description are not available.
301 Incompatible network address formats: One or more network
address formats contained in the session description are
not available.
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302 Incompatible transport protocol: One or more transport
protocols described in the session description are not
available.
303 Incompatible bandwidth units: One or more bandwidth
measurement units contained in the session description were
not understood.
304 Media type not available: One or more media types contained
in the session description are not available.
305 Incompatible media format: One or more media formats
contained in the session description are not available.
306 Attribute not understood: One or more of the media
attributes in the session description are not supported.
307 Session description parameter not understood: A parameter
other than those listed above was not understood.
330 Multicast not available: The site where the user is located
does not support multicast.
331 Unicast not available: The site where the user is located
does not support unicast communication (usually due to the
presence of a firewall).
370 Insufficient bandwidth: The bandwidth specified in the
session description or defined by the media exceeds that
known to be available.
399 Miscellaneous warning: The warning text can include
arbitrary information to be presented to a human user, or
logged. A system receiving this warning MUST NOT take any
automated action.
1xx and 2xx have been taken by HTTP/1.1.
If the warning is caused by the session description, the status
response SHOULD include a session description similar to that
included in OPTIONS responses indicating the capabilities of the UAS.
Additional "warn-code"s, as in the example below, can be defined
through IANA.
Examples:
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Warning: 307 isi.edu "Session parameter 'foo' not understood"
Warning: 301 isi.edu "Incompatible network address type 'E.164'"
22.42 WWW-Authenticate
The WWW-Authenticate header field consists of a challenge that
indicates the authentication scheme and parameters applicable for
this Request-URI.
The syntax for this header and use is defined in [H14.47]. See 20.2.2
for further details on its usage.
Example:
WWW-Authenticate: Digest realm="Bob's Friends",
domain="sip:boxesbybob.com",
nonce="f84f1cec41e6cbe5aea9c8e88d359",
opaque="", stale=FALSE, algorithm=MD5
23 Response Codes
The response codes are consistent with, and extend, HTTP/1.1 response
codes. Not all HTTP/1.1 response codes are appropriate, and only
those that are appropriate are given here. Other HTTP/1.1 response
codes SHOULD NOT be used. Response codes not defined by HTTP/1.1 have
codes x80 upwards to avoid clashes with future HTTP response codes.
Also, SIP defines a new class, 6xx. The default behavior for unknown
response codes is given for each category of codes.
23.1 Provisional 1xx
Provisional responses indicate that the server or proxy contacted is
performing some further action and does not yet have a definitive
response. A server typically sends a 1xx response if it expects to
takemore than 200 ms to obtain a final response. Note that 1xx
responses are not transmitted reliably, that is, they do not cause
the client to send an ACK.
Provisional (1xx) responses MAY contain message bodies, including
session descriptions.
Provisional responses are also known as informational responses.
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This response indicates that the request has been received by the
next hop server and that some unspecified action is being taken on
behalf of this call (e.g., a database is being consulted). This
response stops retransmissions of an INVITE by a UAC.
The user agent receiving the INVITE is trying to alert the user. This
response MAY be used to initiate local ringback.
23.1.3 181 Call Is Being Forwarded
A proxy server MAY use this status code to indicate that the call is
being forwarded to a different set of destinations.
The called party is temporarily unavailable, but the callee has
decided to queue the call rather than reject it. When the callee
becomes available, it will return the appropriate final status
response. The reason phrase MAY give further details about the status
of the call, e.g., "5 calls queued; expected waiting time is 15
minutes". The server MAY issue several 182 responses to update the
caller about the status of the queued call.
23.1.5 183 Session Progress
The 183 (Session Progress) response is used to convey information
about the progress of the call which is not otherwise classified. The
Reason-Phrase, header fields, or message body MAY be used to convey
more details about the call progress.
23.2 Successful 2xx
The request was successful.
The request has succeeded. The information returned with the response
depends on the method used in the request.
23.3 Redirection 3xx
3xx responses give information about the user's new location, or
about alternative services that might be able to satisfy the call.
23.3.1 300 Multiple Choices
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The address in the request resolved to several choices, each with its
own specific location, and the user (or user agent) can select a
preferred communication end point and redirect its request to that
location.
The response MAY include a message body containing a list of resource
characteristics and location(s) from which the user or user agent can
choose the one most appropriate, if allowed by the Accept request
header.
The choices SHOULD also be listed as Contact fields (Section 22.10).
Unlike HTTP, the SIP response MAY contain several Contact fields or a
list of addresses in a Contact field. User agents MAY use the Contact
header field value for automatic redirection or MAY ask the user to
confirm a choice. However, this specification does not define any
standard for such automatic selection.
This status response is appropriate if the callee can be
reached at several different locations and the server
cannot or prefers not to proxy the request.
23.3.2 301 Moved Permanently
The user can no longer be found at the address in the Request-URI and
the requesting client SHOULD retry at the new address given by the
Contact header field (Section 22.10). The caller SHOULD update any
local directories, address books and user location caches with this
new value and redirect future requests to the address(es) listed.
23.3.3 302 Moved Temporarily
The requesting client SHOULD retry the request at the new address(es)
given by the Contact header field (Section 22.10). The Request-URI
of the new request uses the value of the Contact header in the
response. The new request can take two different forms. In the first
approach, the To, From, Call-ID, and CSeq header fields in the new
request are the same as in the original request, with a new branch
identifier in the Via header field. Proxies MUST follow this behavior
and UACs MAY. In the second approach, UAs MAY also use the Contact
information for the To header field, as well as a new Call-ID value.
The duration of the redirection can be indicated through an Expires
(Section 22.19) header. If there is no explicit expiration time, the
address is only valid for this call and MUST NOT be cached for future
calls.
23.3.4 305 Use Proxy
Various Authors [Page 153]
Internet Draft SIP October 26, 2001
The requested resource MUST be accessed through the proxy given by
the Contact field. The Contact field gives the URI of the proxy. The
recipient is expected to repeat this single request via the proxy.
305 responses MUST only be generated by user agent servers.
23.3.5 380 Alternative Service
The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the
response. Formats for such bodies are not defined here, and may be
the subject of future standardization.
23.4 Request Failure 4xx
4xx responses are definite failure responses from a particular
server. The client SHOULD NOT retry the same request without
modification (e.g., adding appropriate authorization). However, the
same request to a different server might be successful.
23.4.1 400 Bad Request
The request could not be understood due to malformed syntax. The
Reason-Phrase SHOULD identify the syntax problem in more detail,
e.g., "Missing Call-ID header".
23.4.2 401 Unauthorized
The request requires user authentication. This response is issued by
user agent servers and registrars, while 407 (Proxy Authentication
Required) is used by proxy servers.
23.4.3 402 Payment Required
Reserved for future use.
23.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request SHOULD NOT be repeated.
23.4.5 404 Not Found
The server has definitive information that the user does not exist at
the domain specified in the Request-URI. This status is also returned
if the domain in the Request-URI does not match any of the domains
handled by the recipient of the request.
23.4.6 405 Method Not Allowed
Various Authors [Page 154]
Internet Draft SIP October 26, 2001
The method specified in the Request-Line is not allowed for the
address identified by the Request-URI. The response MUST include an
Allow header field containing a list of valid methods for the
indicated address.
23.4.7 406 Not Acceptable
The resource identified by the request is only capable of generating
response entities which have content characteristics not acceptable
according to the accept headers sent in the request.
23.4.8 407 Proxy Authentication Required
This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. SIP access
authentication is explained in section 20 and 20.2.3.
This status code can be used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the
callee requires authentication.
23.4.9 408 Request Timeout
The server could not produce a response within a suitable amount of
time, for example, if it could not determine the location of the user
in time. The client MAY repeat the request without modifications at
any later time.
The requested resource is no longer available at the server and no
forwarding address is known. This condition is expected to be
considered permanent. If the server does not know, or has no facility
to determine, whether or not the condition is permanent, the status
code 404 (Not Found) SHOULD be used instead.
23.4.11 413 Request Entity Too Large
The server is refusing to process a request because the request
entity is larger than the server is willing or able to process. The
server MAY close the connection to prevent the client from continuing
the request.
If the condition is temporary, the server SHOULD include a Retry-
After header field to indicate that it is temporary and after what
time the client MAY try again.
23.4.12 414 Request-URI Too Long
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The server is refusing to service the request because the Request-URI
is longer than the server is willing to interpret.
23.4.13 415 Unsupported Media Type
The server is refusing to service the request because the message
body of the request is in a format not supported by the server for
the requested method. The server SHOULD return a list of acceptable
formats using the Accept, Accept-Encoding and Accept-Language header
fields. UAC processing of this response is described in Section
8.1.3.4.
23.4.14 420 Bad Extension
The server did not understand the protocol extension specified in a
Proxy-Require (Section 22.28) or Require (Section 22.30) header
field. The server SHOULD include a list of the unsupported
extensions in an Unsupported header in the response. UAC processing
of this response is described in Section 8.1.3.4.
23.4.15 421 Extension Required
The UAS needs a particular extension to process the request, but this
extension is not listed in a Supported header in the request.
Responses with this status code MUST contain a Require header listing
the required extensions.
In general, a UAS SHOULD NOT use this response when it wishes to
apply an extension to a request. The end result will often be no
service at all, and a break in interoperability. Rather, servers
SHOULD process the request using baseline SIP capabilities and any
extensions supported by the client.
23.4.16 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in, logged in in such a
manner as to preclude communication with the callee or activated the
"do not disturb" feature). The response MAY indicate a better time to
call in the Retry-After header. The user could also be available
elsewhere (unbeknownst to this host). The reason phrase SHOULD
indicate a more precise cause as to why the callee is unavailable.
This value SHOULD be setable by the user agent. Status 486 (Busy
Here) MAY be used to more precisely indicate a particular reason for
the call failure.
This status is also returned by a redirect server that recognizes the
user identified by the Request-URI, but does not currently have a
Various Authors [Page 156]
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valid forwarding location for that user.
23.4.17 481 Call/Transaction Does Not Exist
This status indicates that the UAS received a request that does not
match any existing dialog or transaction.
23.4.18 482 Loop Detected
The server has detected a loop (Section 3).
23.4.19 483 Too Many Hops
The server received a request that contains a Max-Forwards (Section
22.22) header with the value zero.
23.4.20 484 Address Incomplete
The server received a request with a Request-URI that was incomplete.
Additional information SHOULD be provided.
This status code allows overlapped dialing. With overlapped
dialing, the client does not know the length of the dialing
string. It sends strings of increasing lengths, prompting
the user for more input, until it no longer receives a 484
status response.
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Contact headers.
Revealing alternatives can infringe on privacy concerns of the user
or the organization. It MUST be possible to configure a server to
respond with status 404 (Not Found) or to suppress the listing of
possible choices if the request address was ambiguous.
Example response to a request with the URL lee@example.com :
485 Ambiguous SIP/2.0
Contact: Carol Lee <sip:carol.lee@example.com>
Contact: Ping Lee <sip:p.lee@example.com>
Contact: Lee M. Foote <sip:lee.foote@example.com>
Various Authors [Page 157]
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Some email and voice mail systems provide this
functionality. A status code separate from 3xx is used
since the semantics are different: for 300, it is assumed
that the same person or service will be reached by the
choices provided. While an automated choice or sequential
search makes sense for a 3xx response, user intervention is
required for a 485 response.
The callee's end system was contacted successfully but the callee is
currently not willing or able to take additional calls at this end
system. The response MAY indicate a better time to call in the
Retry-After header. The user could also be available elsewhere, such
as through a voice mail service. Status 600 (Busy Everywhere) SHOULD
be used if the client knows that no other end system will be able to
accept this call.
23.4.23 487 Request Terminated
The request was terminated by a BYE or CANCEL request. This response
is never returned for a CANCEL request itself.
23.4.24 488 Not Acceptable Here
The response has the same meaning as 606 (Not Acceptable), but only
applies to the specific entity addressed by the Request-URI and the
request may succeed elsewhere.
23.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has
erred.
23.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from
fulfilling the request. The client MAY display the specific error
condition, and MAY retry the request after several seconds.
If the condition is temporary, the server MAY indicate when the
client may retry the request using the Retry-After header.
23.5.2 501 Not Implemented
The server does not support the functionality required to fulfill the
request. This is the appropriate response when a UAS does not
recognize the request method and is not capable of supporting it for
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any user. (Proxies forward all requests regardless of method.)
23.5.3 502 Bad Gateway
The server, while acting as a gateway or proxy, received an invalid
response from the downstream server it accessed in attempting to
fulfill the request.
23.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading (i.e., congestion) or maintenance of the
server. The implication is that this is a temporary condition which
will be alleviated after some delay. If known, the length of the
delay MAY be indicated in a Retry-After header. If no Retry-After is
given, the client MUST handle the response as it would for a 500
response.
A client (proxy or UAC) receiving a 503 SHOULD attempt to forward the
request to an alternate server. It SHOULD NOT forward any other
requests to that server for the duration specified in the Retry-After
header, if present.
Note: The existence of the 503 status code does not imply that a
server has to use it when becoming overloaded. Some servers MAY wish
to simply refuse the connection.
23.5.5 504 Server Time-out
The server did not receive a timely response from the server (e.g., a
location server) it accessed in attempting to process the request.
Note that 408 (Request Timeout) should be used if there was no
response within the period specified in the Expires header field from
the upstream server.
23.5.6 505 Version Not Supported
The server does not support, or refuses to support, the SIP protocol
version that was used in the request message. The server is
indicating that it is unable or unwilling to complete the request
using the same major version as the client, other than with this
error message. The response MAY contain an entity describing why that
version is not supported and what other protocols are supported by
that server. The format for such an entity is not defined here and
may be the subject of future standardization.
23.5.7 513 Message Too Large
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The server was unable to process the request since the message length
exceeded its capabilities.
23.6 Global Failures 6xx
6xx responses indicate that a server has definitive information about
a particular user, not just the particular instance indicated in the
Request-URI.
23.6.1 600 Busy Everywhere
The callee's end system was contacted successfully but the callee is
busy and does not wish to take the call at this time. The response
MAY indicate a better time to call in the Retry-After header. If the
callee does not wish to reveal the reason for declining the call, the
callee uses status code 603 (Decline) instead. This status response
is returned only if the client knows that no other end point (such as
a voice mail system) will answer the request. Otherwise, 486 (Busy
Here) should be returned.
The callee's machine was successfully contacted but the user
explicitly does not wish to or cannot participate. The response MAY
indicate a better time to call in the Retry-After header.
23.6.3 604 Does Not Exist Anywhere
The server has authoritative information that the user indicated in
the Request-URI does not exist anywhere.
23.6.4 606 Not Acceptable
The user's agent was contacted successfully but some aspects of the
session description such as the requested media, bandwidth, or
addressing style were not acceptable.
A 606 (Not Acceptable) response means that the user wishes to
communicate, but cannot adequately support the session described. The
606 (Not Acceptable) response MAY contain a list of reasons in a
Warning header field describing why the session described cannot be
supported. Reasons are listed in Section 22.41. It is hoped that
negotiation will not frequently be needed, and when a new user is
being invited to join an already existing conference, negotiation may
not be possible. It is up to the invitation initiator to decide
whether or not to act on a 606 (Not Acceptable) response.
24 Locating a SIP Server
Various Authors [Page 160]
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NOTE: Usage of SRV records is still under discussion with
IESG, and therefore this section is likely to change in
subsequent versions of bis.
The SIP URI provides a way to identify a communications resource. For
this URI to be useful in a SIP element, a mechanism is necessary to
take this URI and determine the IP address, port, and transport of
one or more servers that message destined for this URI should be sent
to. We refer to the combination of an IP address, port, and transport
as a next hop next hop can be configured to be the same for all URIs.
In this case, the next hop is referred to as a outbound proxy
commonly used in a user agent which is required to send all requests
to a specific server for policy processing or firewall traversal, for
example. The outbound proxy can be configured by any mechanism,
including DHCP [39].
When the next hop is not configured, a mechanism is needed to
determine one or more next hops from the URI. Section 24.1 provides
an algorithm which can be used to determine an ordered list of next
hops. Typically, the URI that is used is from the Request-URI of a
request, in order to determine where to send that request. However,
in certain circumstances (which are documented in Section 19.2.2), a
URI may have been extracted from a response in order to determine
where to send the response.
Once the ordered list of next hops is computed, they are used
according to the procedures of Section 24.2.
24.1 Computing the List of Next Hops
The algorithm for computing the list of next hops begins by setting
three variables. The first variable is called the target address
maddr parameter of the URI, if present. If not present, it MUST be
set to the host element of the URI. The next variable is called the
target port set to the port element of the URI if present, else the
target port MUST remain empty. The target transport MUST be set to
the headertransport element of the URI if present, else the target
transport MUST remain empty.
The algorithm begins by examining the target address. If it contains
a numeric IP address, the procedures of Section 24.1.1 MUST be
followed. Otherwise, the target transport is examined. If it is
empty, and the target port is either empty or contains a value of
5060, the procedures of Section 24.1.2 MUST be followed. If the
target transport is not empty, and the target port is empty, the
procedures of Section 24.1.2 MUST be followed if the target transport
is UDP. If the target transport and target port are not empty, but
the target port contains the default port for the target transport
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(5060 for UDP, TCP, and SCTP, 5061 for TLS), the procedures of
Section 24.1.2 MUST also be followed. Otherwise, the procedures of
Section 24.1.3 MUST be followed. Effectively, this case occurs when
the target port and target transport don't "match", taking into
account their defaults if empty.
24.1.1 Numeric Destination Address
The addresses of the next hops are all the same, and MUST be equal to
the value of the target address.
If the target transport is specified, and the element supports that
transport, there is only a single next hop, using the target
transport. If the target transport is not specified, the number of
next hops is equal to the number of transports the element supports.
The first next hop MUST be UDP, and the ordering of the remaining
transports is at the discretion of the element.
For each next hop, the port number is equal to the target port, if
specified, otherwise the default port for that transport of that next
hop.
For example, consider the SIP URI sip:joe@1.2.3.4 present in the
Request-URI of a request. A UAC wishes to use this URI to determine
the set of next hops. The UAC supports UDP and TLS. It applies the
algorithm in this section, and ends up with the following ordered
list of IP address, port, transport:
{1.2.3.4, 5060, UDP}
{1.2.3.4, 5061, TLS}
24.1.2 SRV Resolution of Host Name
DNS SRV records are retrieved according to RFC 2782 [40]. The service
identifier for DNS SRV records is "_sip". If the target transport is
not empty, only records for that transport are retrieved. (If the
element does not support the transport specified, the lookup fails.)
If the target transport is empty, the element retrieves records for
all transport protocols it supports. The results of all queries are
merged and then sorted according to priority, independent of the
transport protocol. If this list is empty, follow the procedure in
Section 24.1.3.
Note that the behavior above differs slightly from that described in
RFC 2782. There, A records are consulted if the query for one
Various Authors [Page 162]
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transport protocol fails; here, we only abandon the SRV lookup if
none of the transport protocols supported by the client yield an
answer.
Clients MUST NOT cache query results except according to the rules in
RFC 1035 [41].
24.1.3 Address Record Resolution of Host Name
When the target address is not a numeric IP, and there is a target
port which does not match the default port for the target transport,
SRV records are not used. This is because SRV will normally provide
ports, so if one is provided that is not a default, this would seem
to imply the the URL is trying to explicitly identify the
destination, rather than using SRV.
In this case, the client queries the DNS server for address records
for the destination address. Address records include A RR's, AAAA
RR's, or other similar records, chosen according to the client's
network protocol capabilities.
The DNS address records are kept sorted in the order returned by the
DNS server. For each address, the port is set to the target port. For
each address, the transport is set to the target transport if not
empty, otherwise, the target transport MUST be UDP for the first
address, and is at the discretion of the implementation for the
others.
OPEN ISSUE #221: Selection of transports for the case when
multiple A records are returned requires more work.
Clients MUST NOT cache query results except according to the rules in
RFC 1035 [41].
24.2 Contacting the Next Hops
The algorithms of the previous section will result in an ordered list
of next hops. This section describes how that list is used.
If the ordered list was obtained through SRV, servers are contacted
as specified in the "Usage rules" section of RFC 2782 [40], which
describes procedures for using the weight field to randomly select
servers amongst those of equal priority.
The SIP element takes the ordered list, and it tries to contact each
next hop in turn, until a server responds. If contacting a next hop
results in a failure, as defined in the next paragraph, the element
Various Authors [Page 163]
Internet Draft SIP October 26, 2001
moves to the next next hop in the list, until the list is exhausted.
If the list is exhausted, then the element gives up.
Failures SHOULD be detected through network failure indications or
timeouts. If the element sending the message is a client sending a
request using a client transaction, the client transaction will
report any transport layer failures. If the element sending the
message is a client sending a request directly to the transport
layer, the transport layer will report any failures (See Section
19.4). In either case, the client SHOULD try the next address. This
will involve creating a new client transaction for it in the former
case. The new request MUST have a new branch ID in the Via header.
Note also that the new destination might be with a different
transport, which might require a change in other parts of the Via
header.
Response failures are handled by the transport layer itself, which
may retry the response to the next next hop. See Section 19.2.2.
Failures can be detected through timeouts only if the element is a
client sending a request through the client transaction. In that
case, if a timeout is reported by the client transaction, the client
SHOULD try the next next hop in the list.
OPEN ISSUE #219: It might be easier to encapsulate the SRV
processing in one place, at the transport layer, rather
than the behavior being dependent on client v. server. This
can only be done if merging of srv records across
transports is deprecated, along with failures based on
timeouts.
Once a next hop is successfully contacted, that same next hop address
MUST be used for all subsequent messages that share the same Call-ID.
More specifically, once a request is delivered successfully to a
particular next hop, all subsequent requests with the same Call-ID
MUST be delivered to that next hop. Once a response is delivered
successfully to a particular next hop, all subsequent responses with
the same Call-ID MUST be delivered to that next hop. However, if that
next hop fails, the selection algorithms MUST be re-run for the top.
This is a change from RFC2543, which only used the same
address for requests within a transaction. Broadening the
scope to Call-ID helps, for example, ensure that requests
with credentials after a challenge are delivered to the
same server that issued the challenge.
Various Authors [Page 164]
Internet Draft SIP October 26, 2001
A stateless proxy can accomplish this, for example, by using the
modulo N of a hash of the Call-ID value as the uniform random number
described in the weighting algorithm of RFC 2782 [40]. Here, N is the
sum of weights within the priority class.
OPEN ISSUE #220: This stateless selection algorithm doesn't
work if there are failures.
25 Examples
In the following examples, we often omit the message body and the
corresponding Content-Length and Content-Type headers for brevity.
25.1 Registration
Bob registers on start-up. The message flow is shown in Figure 9.
biloxi.com Bob's
registrar softphone
| |
| REGISTER F1 |
|<---------------|
| 200 OK F2 |
|--------------->|
Figure 9: SIP Registration Example
F1 REGISTER Bob -> Registrar
REGISTER sip:registrar.biloxi.com
Via: SIP/2.0/UDP 10.4.1.4:5060
To: Bob <sip:bob@biloxi.com>
From: Bob <sip:bob@biloxi.com>;tag=456248
Call-ID: 843817637684230@phone21.boxesbybob.com
CSeq: 1826 REGISTER
Contact: <sip:bob@10.4.1.4>
Expires: 7200
Contact-Length: 0
Various Authors [Page 165]
Internet Draft SIP October 26, 2001
The registration expires after two hours. The registrar responds with
a 200 OK:
F2 200 OK Registrar -> Bob
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.1.4:5060
To: Bob <sip:bob@biloxi.com>
From: Bob <sip:bob@biloxi.com>;tag=456248
Call-ID: 843817637684230@phone21.boxesbybob.com
CSeq: 1826 REGISTER
Contact: <sip:bob@10.4.1.4>
Expires: 7200
Contact-Length: 0
25.2 Session Setup
This example contains the full details of the example session setup
in Section 4. The message flow is shown in Figure 1.
F1 INVITE Alice -> atlanta.com proxy
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:alice@10.1.3.3>
Content-Type: application/sdp
Contact-Length: 142
(Alice's SDP not shown)
F2 100 Trying atlanta.com proxy -> Alice
SIP/2.0 100 Trying
Various Authors [Page 166]
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Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact-Length: 0
F3 INVITE atlanta.com proxy -> biloxi.com proxy
INVITE sip:bob@biloxi.com SIP/2.0
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:alice@10.1.3.3>
Content-Type: application/sdp
Contact-Length: 142
(Alice's SDP not shown)
F4 100 Trying biloxi.com proxy -> atlanta.com proxy
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact-Length: 0
F5 INVITE biloxi.com proxy -> Bob
INVITE sip:bob@10.4.1.4 SIP/2.0
Various Authors [Page 167]
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Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:alice@10.1.3.3>
Content-Type: application/sdp
Contact-Length: 142
(Alice's SDP not shown)
F6 180 Ringing Bob -> biloxi.com proxy
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact-Length: 0
F7 180 Ringing biloxi.com proxy -> atlanta.com proxy
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact-Length: 0
Various Authors [Page 168]
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F8 180 Ringing atlanta.com proxy -> Alice
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact-Length: 0
F9 200 OK Bob -> biloxi.com proxy
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.1.1:5060;branch=4b43c2ff8.1
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:bob@10.4.1.4>
Content-Type: application/sdp
Contact-Length: 131
(Bob's SDP not shown)
F10 200 OK biloxi.com proxy -> atlanta.com proxy
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.1:5060;branch=77ef4c2312983.1
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:bob@10.4.1.4>
Content-Type: application/sdp
Contact-Length: 131
(Bob's SDP not shown)
Various Authors [Page 169]
Internet Draft SIP October 26, 2001
F11 200 OK atlanta.com proxy -> Alice
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 INVITE
Contact: <sip:bob@10.4.1.4>
Content-Type: application/sdp
Contact-Length: 131
(Bob's SDP not shown)
F12 ACK Alice -> Bob
ACK sip:bob@10.4.1.4 SIP/2.0
Via: SIP/2.0/UDP 10.1.3.3:5060
To: Bob <sip:bob@biloxi.com>;tag=a6c85cf
From: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 314159 ACK
Contact-Length: 0
The media session between Alice and Bob is now established.
Bob hangs up first. Note that Bob's SIP phone maintains its own CSeq
numbering space, which, in this example, begins with 231. Also not
that since Bob is making the request, the To and From URLs and tags
have been swapped.
F13 BYE Bob -> Alice
BYE sip:alice@10.1.3.3 SIP/2.0
Via: SIP/2.0/UDP 10.4.1.4:5060
From: Bob <sip:bob@biloxi.com>;tag=a6c85cf
To: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 231 BYE
Contact-Length: 0
Various Authors [Page 170]
Internet Draft SIP October 26, 2001
F14 200 OK Alice -> Bob
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.4.1.4:5060
From: Bob <sip:bob@biloxi.com>;tag=a6c85cf
To: Alice <sip:alice@atlanta.com>;tag=1928301774
Call-ID: a84b4c76e66710@10.1.3.3
CSeq: 231 BYE
Contact-Length: 0
The SIP Call Flows document [42] contains further examples of SIP
messages.
;; This buffer is for notes you don't want to save, and for Lisp
evaluation. ;; If you want to create a file, first visit that file
with C-x C-f, ;; then enter the text in that file's own buffer.
26 Augmented BNF for the SIP Protocol
All of the mechanisms specified in this document are described in
both prose and an augmented Backus-Naur Form (BNF) similar to that
used by RFC 822 [12] and RFC 2234 [43]. Implementors will need to be
familiar with the notation in order to understand this specification.
The augmented BNF includes the following constructs:
name = definition
The name of a rule is simply the name itself (without any enclosing
"<" and ">") and is separated from its definition by the equal "="
character. White space is only significant in that indentation of
continuation lines is used to indicate a rule definition that spans
more than one line. Certain basic rules are in uppercase, such as SP,
LWS, HT, CRLF, DIGIT, ALPHA, etc. Angle brackets are used within
definitions whenever their presence will facilitate discerning the
use of rule names.
"literal"
Quotation marks surround literal text. Unless stated otherwise, the
text is case-insensitive.
Various Authors [Page 171]
Internet Draft SIP October 26, 2001
rule1 | rule2
Elements separated by a bar ("|") are alternatives, e.g., "yes | no"
will accept yes or no.
(rule1 rule2)
Elements enclosed in parentheses are treated as a single element.
Thus, "(elem (foo | bar) elem)" allows the token sequences "elem foo
elem" and "elem bar elem".
*rule
The character "*" preceding an element indicates repetition. The full
form is "<n>*<m>element" indicating at least <n> and at most <m>
occurrences of element. Default values are 0 and infinity so that
"*(element)" allows any number, including zero; "1*element" requires
at least one; and "1*2element" allows one or two.
[rule]
Square brackets enclose optional elements; "[foo bar]" is equivalent
to "*1(foo bar)".
N rule
Specific repetition: "<n>(element)" is equivalent to
"<n>*<n>(element)"; that is, exactly <n> occurrences of (element).
Thus 2DIGIT is a 2-digit number, and 3ALPHA is a string of three
alphabetic characters.
#rule
Various Authors [Page 172]
Internet Draft SIP October 26, 2001
A construct "#" is defined, similar to "*", for defining lists of
elements. The full form is "<n>#<m> element" indicating at least <n>
and at most <m> elements, each separated by one or more commas (",")
and OPTIONAL linear white space (LWS). This makes the usual form of
lists very easy; a rule such as
( *LWS element *( *LWS "," *LWS element ))
can be shown as 1# element. Wherever this construct is used, null
elements are allowed, but do not contribute to the count of elements
present. That is, "(element), , (element)" is permitted, but counts
as only two elements. Therefore, where at least one element is
required, at least one non-null element MUST be present. Default
values are 0 and infinity so that "#element" allows any number,
including zero; "1#element" requires at least one; and "1#2element"
allows one or two.
; comment
A semi-colon, set off some distance to the right of rule text, starts
a comment that continues to the end of line. This is a simple way of
including useful notes in parallel with the specifications.
26.1 Basic Rules
The following rules are used throughout this specification to
describe basic parsing constructs. The US-ASCII coded character set
is defined by ANSI X3.4-1986.
OCTET = %x00-ff ; any 8-bit sequence of data
CHAR = %x00-7f ; any US-ASCII character (octets 0 - 127)
upalpha = "A" | "B" | "C" | "D" | "E" | "F" | "G" | "H" | "I" |
"J" | "K" | "L" | "M" | "N" | "O" | "P" | "Q" | "R" |
"S" | "T" | "U" | "V" | "W" | "X" | "Y" | "Z"
lowalpha = "a" | "b" | "c" | "d" | "e" | "f" | "g" | "h" | "i" |
"j" | "k" | "l" | "m" | "n" | "o" | "p" | "q" | "r" |
"s" | "t" | "u" | "v" | "w" | "x" | "y" | "z"
alpha = lowalpha | upalpha
DIGIT = "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" |
"8" | "9"
Various Authors [Page 173]
Internet Draft SIP October 26, 2001
alphanum = alpha | DIGIT
CTL = %x00-1f | %x7f ; (octets 0 -- 31) and DEL (127)
CR = %d13 ; US-ASCII CR, carriage return character
LF = %d10 ; US-ASCII LF, line feed character
SP = %d32 ; US-ASCII SP, space character
HT = %d09 ; US-ASCII HT, horizontal tab character
CRLF = CR LF ; typically the end of a line
The following are defined in RFC 2396 [9] for the SIP URI:
unreserved = alphanum | mark
mark = "-" | "_" | "." | "!" | "~" | "*" | "'"
| "(" | ")"
escaped = "%" hex hex
SIP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP 1.1 as described in
RFC2615 [8].
LWS = *( SP | HT ) [CRLF] 1*( SP | HT ) ; linear whitespace
To separate the header name from the rest of value, a colon is used,
which, by the above rule allows whitespace before, but no line break,
and whitespace after, including a linebreak. The HCOLON defines this
construct.
HCOLON = *( SP | HT ) ":" LWS
The TEXT-UTF8 rule is only used for descriptive field contents and
values that are not intended to be interpreted by the message parser.
Words of *TEXT-UTF8 contain characters from the UTF-8 character set
(RFC 2279 [11]). The TEXT-UTF8-TRIM rule is used for descriptive
field contents that are not quoted strings, where leading and
trailing LWS is not meaningful. In this regard, SIP differs from
HTTP, which uses the ISO 8859-1 character set.
Various Authors [Page 174]
Internet Draft SIP October 26, 2001
TEXT-UTF8 = *(TEXT-UTF8char | LWS)
TEXT-UTF8-TRIM = *TEXT-UTF8char *(*LWS TEXT-UTF8char)
TEXT-UTF8char = %x21-7e
| UTF8-NONASCII
UTF8-NONASCII = %xc0-df 1UTF8-CONT
| %xe0-ef 2UTF8-CONT
| %xf0-f7 3UTF8-CONT
| %xf8-fb 4UTF8-CONT
| %xfc-fd 5UTF8-CONT
UTF8-CONT = %x80-bf
A CRLF is allowed in the definition of TEXT-UTF8 only as part of a
header field continuation. It is expected that the folding LWS will
be replaced with a single SP before interpretation of the TEXT-UTF8
value.
Hexadecimal numeric characters are used in several protocol elements.
Some elements (authentication) force hex alphas to be lower case.
LHEX = digit | "a" | "b" | "c" | "d" | "e" | "f"
Others allow mixed upped and lower case
hex = LHEX | "A" | "B" | "C" | "D" | "E" | "F"
Many SIP header field values consist of words separated by LWS or
special characters. Unless otherwise stated, tokens are case-
insensitive. These special characters MUST be in a quoted string to
be used within a parameter value.
token = 1*(alphanum | "-" | "." | "!" | "%" | "*"
| "_" | "+" | "`" | "'" | "~" )
separators = "(" | ")" | "<" | ">" | "@" |
"," | ";" | ":" | "\" | <"> |
"/" | "[" | "]" | "?" | "=" |
"{" | "}" | SP | HT
When tokens are used or separators are used between elements,
whitespace is often allowed before or after these characters:
Various Authors [Page 175]
Internet Draft SIP October 26, 2001
MINUS = LWS "-" LWS ; minus
DOT = LWS "." LWS ; period
PERCENT = LWS "%" LWS ; percent
BANG = LWS "!" LWS ; exclamation
PLUS = LWS "+" LWS ; plus
STAR = LWS "*" LWS ; askerisk
TILDE = LWS " LWS ; tilde
EQUAL = LWS "=" LWS ; equal
LPAREN = LWS "(" LWS ; left parenthesis
RPAREN = LWS ")" LWS ; right parenthesis
LANGLE = LWS "<" LWS ; left angle bracket
RAQUOT = ">" LWS ; right angle quote
LAQUOT = LWS "<"; left angle quote
RANGLE = LWS ">" LWS ; right angle bracket
BAR = LWS "|" LWS ; vertical bar
ATSIGN = LWS "@" LWS ; atsign
COMMA = LWS "," LWS ; comma
SEMI = LWS ";" LWS ; semicolon
COLON = LWS ":" LWS ; colon
DQUOT = LWS <"> LWS ; double quotation mark
LDQUOT = LWS <">; open double quotation mark
RDQUOT = <"> LWS ; close double quotation mark
LBRACK = LWS "{" LWS ; left square bracket
RBRACK = LWS "}" LWS ; right square bracket
Comments can be included in some SIP header fields by surrounding the
comment text with parentheses. Comments are only allowed in fields
containing "comment" as part of their field value definition. In all
other fields, parentheses are considered part of the field value.
comment = LPAREN *(ctext | quoted-pair | comment) RPAREN
ctext = < any TEXT-UTF8 excluding "(" and ")">
A string of text is parsed as a single word if it is quoted using
double-quote marks. In quoted strings, quotation marks (") and
backslashes (\) need to be escaped.
quoted-string = ( LWS <"> *(qdtext | quoted-pair ) <"> )
qdtext = LWS | %x21 | %x23-5b | %x5d-7e
| UTF8-NONASCII
Various Authors [Page 176]
Internet Draft SIP October 26, 2001
The backslash character ("\") MAY be used as a single-character
quoting mechanism only within quoted-string and comment constructs.
Unlike HTTP/1.1, the characters CR and LF cannot be escaped by this
mechanism to avoid conflict with line folding and header separation.
quoted-pair = "\" (%x00 - %x09 | %x0b | %x0c | %x0e - %x7f)
SIP-URL = "sip:" [ userinfo "@" ] hostport
url-parameters [ headers ]
userinfo = [ user | telephone-subscriber [ ":" password ]]
user = *( unreserved | escaped | user-unreserved )
user-unreserved = " " | "=" | "+" | "$" | "," | ";" | "?" | "/"
password = *( unreserved | escaped |
" " | "=" | "+" | "$" | "," )
hostport = host [ ":" port ]
host = hostname | IPv4address | IPv6reference
hostname = *( domainlabel "." ) toplabel [ "." ]
domainlabel = alphanum
| alphanum *( alphanum | "-" ) alphanum
toplabel = alpha | alpha *( alphanum | "-" ) alphanum
IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6reference = "[" IPv6address "]"
IPv6address = hexpart [ ":" IPv4address ]
hexpart = hexseq | hexseq "::" [ hexseq ] | "::" [ hexseq ]
hexseq = hex4 *( ":" hex4)
hex4 = 1*4HEX
port = 1*DIGIT
url-parameters = *( ";" url-parameter)
url-parameter = transport-param | user-param | method-param
|ttl-param | maddr-param | other-param
transport-param = "transport="
( "udp" | "tcp" | "sctp" | "tls"
| other-transport)
other-transport = token
user-param = "user=" ( "phone" | "ip" | other-user)
Various Authors [Page 177]
Internet Draft SIP October 26, 2001
other-user = token
method-param = "method=" Method
ttl-param = "ttl=" ttl
maddr-param = "maddr=" host
other-param = pname [ "=" pvalue ]
pname = 1*paramchar
pvalue = 1*paramchar
paramchar = param-unreserved | unreserved | escaped
param-unreserved = "[" | "]" | "/" | ":" | " " | "+" | "$"
headers = "?" header *( " " header )
header = hname "=" hvalue
hname = 1*( hnv-unreserved | unreserved | escaped )
hvalue = *( hnv-unreserved | unreserved | escaped )
hnv-unreserved = "[" | "]" | "/" | "?" | ":" | "+" | "$"
SIP-message = Request | Response
Request = Request-Line
*( message-header )
CRLF
[ message-body ]
Request-Line = Method SP Request-URI SP SIP-Version CRLF
Request-URI = SIP-URL | absoluteURI
SIP-Version = "SIP/2.0"
message-header
= Accept
| Accept-Encoding
| Accept-Language
| Alert-Info
| Allow
| Authentication-Info
| Authorization
| Call-ID
| Call-Info
| Contact
| Content-Disposition
| Content-Encoding
| Content-Language
Various Authors [Page 178]
Internet Draft SIP October 26, 2001
| Content-Length
| Content-Type
| CSeq
| Date
| Error-Info
| Expires
| From
| In-Reply-To
| Max-Forwards
| MIME-Version
| Organization
| Priority
| Proxy-Authenticate
| Proxy-Authorization
| Proxy-Require
| Record-Route
| Require
| Retry-After
| Route
| Server
| Subject
| Supported
| Timestamp
| To
| Unsupported
| User-Agent
| Via
| Warning
| WWW-Authenticate
Method = "INVITE" | "ACK" | "OPTIONS" | "BYE"
| "CANCEL" | "REGISTER" | extension-method
extension-method = token
option-tag = token
Response
= Status-Line
*( message-header )
CRLF
[ message-body ]
Status-Line = SIP-version SP Status-Code SP Reason-Phrase CRLF
Status-Code
Various Authors [Page 179]
Internet Draft SIP October 26, 2001
= Informational
| Redirection
| Success
| Client-Error
| Server-Error
| Global-Failure
| extension-code
extension-code = 3DIGIT
Reason-Phrase
= *<TEXT-UTF8, excluding CR, LF>
Informational
= "100" ; Trying
| "180" ; Ringing
| "181" ; Call Is Being Forwarded
| "182" ; Queued
| "183" ; Session Progress
Success = "200" ; OK
Redirection = "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "305" ; Use Proxy
| "380" ; Alternative Service
Client-Error = "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required
| "408" ; Request Timeout
| "409" ; Conflict
Various Authors [Page 180]
Internet Draft SIP October 26, 2001
| "410" ; Gone
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "420" ; Bad Extension
| "480" ; Temporarily not available
| "481" ; Call Leg/Transaction Does Not Exist
| "482" ; Loop Detected
| "483" ; Too Many Hops
| "484" ; Address Incomplete
| "485" ; Ambiguous
| "486" ; Busy Here
| "487" ; Request Terminated
| "488" ; Not Acceptable Here
Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Server Time-out
| "505" ; SIP Version not supported
Global-Failure = "600" ; Busy Everywhere
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
Accept = "Accept" HCOLON
#( media-range [ accept-params ] )
media-range = ( "*/*"
| ( type LWS "/" "*" LWS )
| ( type SLASH subtype )
) *( SEMI parameter )
accept-params = SEMI "q" EQUAL qvalue *( accept-extension )
accept-extension = SEMI token [ EQUAL ( token | quoted-string ) ]
Various Authors [Page 181]
Internet Draft SIP October 26, 2001
Accept-Encoding = "Accept-Encoding" HCOLON
1#( codings [ SEMI "q" EQUAL qvalue ] LWS )
codings = ( content-coding | "*" )
content-coding = token
qvalue = ( "0" [ "." 0*3DIGIT ] )
| ( "1" [ "." 0*3("0") ] )
Accept-Language = "Accept-Language" HCOLON
1#( language-range [ SEMI "q" EQUAL qvalue ] )
language-range = ( ( 1*8ALPHA *( MINUS 1*8ALPHA ) ) | "*" )
Alert-Info = "Alert-Info" HCOLON #
( LAQUOT URI RAQUOT *( COLON generic-param ))
generic-param = token [ EQUAL ( token | host |
quoted-string ) ]
Allow = "Allow" HCOLON 1#Method
Authorization = "Authorization" HCOLON credentials
credentials = LWS "Digest" digest-response
digest-response = 1#( username | realm | nonce | digest-uri
| dresponse | [ algorithm ] | [cnonce]
| [opaque] | [message-qop]
| [nonce-count] | [auth-param] )
username = "username" EQUAL username-value
username-value = quoted-string
digest-uri = "uri" EQUAL digest-uri-value
digest-uri-value = request-uri ; As specified by HTTP/1.1
message-qop = "qop" EQUAL qop-value
cnonce = "cnonce" EQUAL cnonce-value
cnonce-value = nonce-value
nonce-count = "nc" EQUAL nc-value
dresponse = "response" EQUAL request-digest
request-digest = LDQUOT 32LHEX RDQUOT
Various Authors [Page 182]
Internet Draft SIP October 26, 2001
AuthenticationInfo = "Authentication-info" HCOLON 1#( digest | nextnonce )
nextnonce = "nextnonce" EQUAL nonce-value
callid = token [ ATSIGN token ]
Call-ID = ( "Call-ID" | "i" ) HCOLON callid
Call-Info = "Call-Info" HCOLON # ( LAQUOT URI RAQUOT
*( SEMI info-param) )
info-param = "purpose" EQUAL ( "icon" | "info"
| "card" | token ) | generic-param
Contact = ( "Contact" | "m" ) HCOLON
(STAR | (1# (( name-addr | addr-spec )
*( SEMI contact-params ))))
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = SIP-URL | URI
display-name = LWS (*token | quoted-string)
contact-params = "q" EQUAL qvalue
| "action" EQUAL "proxy" | "redirect"
| "expires" EQUAL delta-seconds |
LDQUOT SIP-date RDQUOT
| contact-extension
contact-extension = generic-param
qvalue = ( "0" [ "." 0*3DIGIT ] )
| ( "1" [ "." 0*3("0") ] )
delta-seconds = 1*DIGIT
Content-Disposition = "Content-Disposition" HCOLON
disposition-type *( SEMI disposition-param )
disposition-type = "render" | "session" | "icon" | "alert"
| disp-extension-token
disposition-param = "handling" EQUAL
Various Authors [Page 183]
Internet Draft SIP October 26, 2001
( "optional" | "required" |
other-handling ) | generic-param
other-handling = token
disp-extension-token = token
Content-Encoding = ( "Content-Encoding" | "e" ) HCOLON
1#content-coding
Content-Language = "Content-Language" HCOLON 1#language-tag
language-tag = primary-tag *( MINUS subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Content-Length = ( "Content-Length" | "l" ) HCOLON 1*DIGIT
Content-Type = ( "Content-Type" | "c" ) HCOLON media-type
CSeq = "CSeq" HCOLON 1*DIGIT Method
Date = "Date" HCOLON SIP-date
SIP-date = rfc1123-date
rfc1123-date = wkday COMMA SP date1 SP time SP "GMT"
date1 = 2DIGIT SP month SP 4DIGIT
; day month year (e.g., 02 Jun 1982)
time = 2DIGIT ":" 2DIGIT ":" 2DIGIT
; 00:00:00 - 23:59:59
wkday = "Mon" | "Tue" | "Wed"
| "Thu" | "Fri" | "Sat" | "Sun"
month = "Jan" | "Feb" | "Mar" | "Apr"
| "May" | "Jun" | "Jul" | "Aug"
Various Authors [Page 184]
Internet Draft SIP October 26, 2001
| "Sep" | "Oct" | "Nov" | "Dec"
Error-Info = "Error-Info" HCOLON #
( LAQUOT URI RAQUOT
*( SEMI generic-param ))
Expires = "Expires" HCOLON ( SIP-date | delta-seconds )
From = ( "From" | "f" ) HCOLON
( name-addr | addr-spec )
*( SEMI from-param )
from-param = tag-param | generic-param
tag-param = "tag" EQUAL token
In-Reply-To = "In-Reply-To" HCOLON 1# callid
Max-Forwards = "Max-Forwards" HCOLON 1*DIGIT
MIME-Version = "MIME-Version" HCOLON 1*DIGIT "." 1*DIGIT
Organization = "Organization" HCOLON TEXT-UTF8-TRIM
Priority = "Priority" HCOLON priority-value
priority-value = "emergency" | "urgent" | "normal"
| "non-urgent" | other-priority
other-priority = token
Various Authors [Page 185]
Internet Draft SIP October 26, 2001
Proxy-Authenticate = "Proxy-Authenticate" HCOLON 1#challenge
challenge = LWS "Digest" digest-challenge
digest-challenge = 1#( realm | [ domain ] | nonce |
[ opaque ] | [ stale ] | [ algorithm ] |
[ qop-options ] | [auth-param] )
realm = "realm" EQUALS realm-value
realm-value = quoted-string
domain = "domain" EQUAL LDQUOT URI
( 1*SP URI ) RDQUOT
URI = absoluteURI | abs_path
nonce = "nonce" EQUAL nonce-value
nonce-value = quoted-string
opaque = "opaque" EQUAL quoted-string
stale = "stale" EQUAL ( "true" | "false" )
algorithm = "algorithm" EQUAL ( "MD5" | "MD5-sess" |
token )
qop-options = "qop" EQUAL LDQUOT 1#qop-value RDQUOT
qop-value = "auth" | "auth-int" | token
Proxy-Authorization = "Proxy-Authorization" HCOLON credentials
Proxy-Require = "Proxy-Require" HCOLON 1#option-tag
Record-Route = "Record-Route" HCOLON 1#
( name-addr *( SEMI rr-param ))
rr-param = generic-param
Require = "Require" HCOLON 1#option-tag
Retry-After = "Retry-After" HCOLON
( SIP-date | delta-seconds )
[ comment ] *( SEMI retry-param )
retry-param = "duration" EQUAL delta-seconds |
generic-param
Various Authors [Page 186]
Internet Draft SIP October 26, 2001
Route = "Route" HCOLON 1# ( name-addr
*( SEMI rr-param ))
Server = "Server" HCOLON 1*( product | comment )
product = token [SLASH product-version]
product-version = token
Subject = ( "Subject" | "s" ) HCOLON TEXT-UTF8-TRIM
Supported = ( "Supported" | "k" ) HCOLON 0#option-tag
Timestamp = "Timestamp" HCOLON *(DIGIT)
[ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
To = ( "To" | "t" ) HCOLON ( name-addr |
addr-spec ) *( SEMI to-param )
to-param = tag-param | generic-param
Unsupported = "Unsupported" HCOLON 1#option-tag
User-Agent = "User-Agent" HCOLON 1*( product | comment )
Via = ( "Via" | "v" ) HCOLON
Various Authors [Page 187]
Internet Draft SIP October 26, 2001
1#( sent-protocol sent-by
*( SEMI via-params ) [ comment ] )
via-params = via-hidden | via-ttl | via-maddr
| via-received | via-branch
| via-extension
via-hidden = "hidden"
via-ttl = "ttl" EQUAL ttl
via-maddr = "maddr" EQUAL host
via-received = "received" EQUAL host
via-branch = "branch" EQUAL token
via-extension = generic-param
sent-protocol = protocol-name SLASH protocol-version
SLASH transport
protocol-name = "SIP" | token
protocol-version = token
transport = "UDP" | "TCP" | "TLS" | "SCTP"
| other-transport
sent-by = host [ COLON port ]
ttl = 1*3DIGIT ; 0 to 255
Warning = "Warning" HCOLON 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 3DIGIT
warn-agent = ( host [ COLON port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
warn-text = quoted-string
pseudonym = token
WWW-Authenticate = "WWW-Authenticate" HCOLON challenge
27 IANA Considerations
All new or experimental method names, header field names, and status
codes used in SIP applications SHOULD be registered with IANA in
order to prevent potential naming conflicts. It is RECOMMENDED that
new "option- tag"s and "warn-code"s also be registered. Before IANA
registration, new protcol elements SHOULD be characterized in an
Internet- Draft or, preferably, an RFC.
For Internet-Drafts, IANA is requested to make the draft available as
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part of the registration database.
By the time an RFC is published, colliding names may have
already been implemented.
When a registration for either a new header field, new method or new
status code is created based on an Internet-Draft, and that
Internet-Draft becomes an RFC, the person that performed the
registration MUST notify IANA to change the registration to point to
the RFC instead of the Internet-Draft.
Registrations should be sent to iana@iana.org
27.1 Option Tags
Option tags are used in headers such as Require, Supported, Proxy-
Require and Unsupported in support of SIP compatibility mechanisms
for extensions. For more on the use of option tags in these headers
see Section 21.2. The option tag itself is a string that is
associated with a particular SIP option (e.g. an extension) in order
to identify the option in signaling between SIP endpoints.
When registering a new SIP option with IANA, the following
information MUST be provided:
o Name and description of option. The name MAY be of any length,
but SHOULD be no more than twenty characters long. The name
MUST consist of alphanum (See Section 26) characters only
o A listing of any new SIP header fields, header parameter
fields or parameter values defined by this option. A SIP
option MUST NOT redefine header fields or parameters defined
in either RFC 2543, any standards-track extensions to RFC
2543, or other extensions registered through IANA
o Indication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies)
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a
computer manual
o Contact information (postal and email address)
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This procedure has been borrowed from RTSP [4] and the RTP
AVP [44].
27.2 Warn-Codes
Warning codes provide information supplemental to the status code in
SIP response messages when the failure of the transaction results
from a Session Description Protocol (SDP, [6]). New "warn-code"
values can be registered with IANA as they arise.
The "warn-code" consists of three digits. A first digit of "3"
indicates warnings specific to SIP.
Warnings 300 through 329 are reserved for indicating problems with
keywords in the session description, 330 through 339 are warnings
related to basic network services requested in the session
description, 370 through 379 are warnings related to quantitative QoS
parameters requested in the session description, and 390 through 399
are miscellaneous warnings that do not fall into one of the above
categories.
1xx and 2xx have been taken by HTTP/1.1.
27.3 Header Field Names
Header field names do not require working group or working group
chair review prior to IANA registration, but SHOULD be documented in
an RFC or Internet- Draft before IANA is consulted.
The following information needs to be provided to IANA in order to
register a new header field name:
o The name and email address of the individual performing the
registration.
o The name of the header field being registered.
o A compact form version for that header field, if one is
defined.
o The name of the draft or RFC where the header field is
defined.
o A copy of the draft or RFC where the header field is defined.
Header fields SHOULD NOT use the X prefix notation and MUST NOT
duplicate the names of header fields used by SMTP or HTTP unless the
syntax is a compatible superset and the semantics are similar. Some
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common and widely used header fields MAY be assigned one-letter
compact forms (Section 7.3.3). Compact forms can only be assigned
after SIP working group review. In the absence of this working group,
a designated expert reviews the request.
27.4 Method and Response Codes
Because the status code space is limited, they do require working
group or working group chair review, and MUST be documented in an RFC
or Internet draft. The same procedures apply to new method names.
The following information needs to be provided to IANA in order to
register a new response code or method:
o The name and email address of the individual performing the
registration.
o The number of the response code or name of the method being
registered.
o The default reason phrase for that status code, if applicable.
o The name of the draft or RFC where the method or status code
is defined.
o A copy of the draft or RFC where the method or status code is
defined.
28 Changes Made in Version 00
o Indicated that UAC should send both CANCEL and BYE after a
retransmission fails.
o Added semicolon and question mark to the list of unreserved
characters for the user part of SIP URLs to handle tel: URLs
properly.
o Uniform handling of if hop count Max-Forwards: return 483.
Note that this differs from HTTP/1.1 behavior, where only
OPTIONS and TRACE allow this header, but respond as the final
recipient when the value reaches zero.
o Clarified that a forking proxy sends ACKs only for INVITE
requests.
o Clarified wording of DNS caching. Added paragraph on "negative
caching", i.e., what to do if one of the hosts failed. It is
probably not a good idea to simply drop this host from the
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list if the DNS ttl value is more than a few minutes, since
that would mean that load balancing may not work for quite a
while after a server is brought back on line. This will be
true in particular if a server group receives a large number
of requests from a small number of upstream servers, as is
likely to be the case for calls between major consumer ISPs.
However, without getting into arbitrary and complicated retry
rules, it seems hard to specify any general algorithm. Might
it be worthwhile to simply limit the "black list" interval to
a few minutes?
o Added optional Call-Info and Alert-Info header fields that
describe the caller and information to be used in alerting.
(Currently, avoided use of "purpose" qualification since it is
not yet clear whether rendering content without understanding
its meaning is always appropriate. For example, if a UAS does
not understand that this header is to replace ringing, it
would mix both local ring tone and the indicated sound URL.)
TBD!
o SDP "s=" lines can't be empty, unfortunately.
o Noted that maddr could also contain a unicast address, but
SHOULD contain the multicast address if the request is sent
via multicast (Section 22.40.
o Clarified that responses are sent to port in Via sent-by
value.
o Added "other-*" to the user URL parameter and the Hide and
Content-Disposition headers.
o Clarified generation of timeout (408) responses in forking
proxies and mention the Expires header.
o Clarified that CANCEL and INVITE are separate transactions
(Fig. 7). Thus, the INVITE request generates a 487 (Request
Terminated) if a CANCEL or BYE arrives.
o Clarified that Record-Route SHOULD be inserted in every
request, but that the route, once established, persists. This
provides robustness if the called UAS crashes.
o Emphasized that proxy, redirect, registrar and location
servers are logical, not physical entities and that UAC and
UAS roles are defined on a request-by-request basis. (Section
6)
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o In Section 22.40, noted that the maddr and received parameters
also need to be encrypted when doing Via hiding.
o Simplified Fig. 7 to only show INVITE transaction.
o Added definition of the use of Contact (Section 22.10) for
OPTIONS.
o Added HTTP/RFC822 headers Content-Language and MIME-Version.
o Added note in minimal section indicating that UAs need to
support UDP.
o Added explanation explaining what a UA should do when
receiving an initial INVITE with a tag.
o Clarified UA and proxy behavior for 302 responses.
o Added details on what a UAS should do when receiving a tagged
INVITE request for an unknown call leg. This could occur if
the UAS had crashed and the UAC sends a re-INVITE or if the
BYE got lost and the UAC still believes to be in the call.
o Added definition of Contact in 4xx, 5xx and 6xx to "redirect"
to more error details.
o Added note to forking proxy description to gather *-
Authenticate from responses. This allows several branches to
be authenticated simultaneously.
o Changed URI syntax to use URL escaping instead of quotation
marks.
o Changed SIP URL definition to reference RFC 2806 for
telephone-subscriber part.
o Clarified that the To URI should basically be ignored by the
receiving UAS except for matching requests to call legs. In
particular, To headers with a scheme or name unknown to the
callee should be accepted.
o Clarified that maddr is to be added by any client, either
proxy or UAC.
o Added response code 488 to indicate that there was no common
media at the particular destination. (606 indicates such
failure globally.)
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o In Section 22.19, noted that registration updates can shorten
the validity period.
o Added note to enclose the URI for digest in quotation marks.
The BNF in RFC 2617 is in error.
o Clarified that registrars use Authorization and WWW-
Authenticate, not proxy authentication.
o Added note in Section 22.10 that "headers" are copied from
Contact into the new request.
o Changed URL syntax so that port specifications have to have at
least one digit, in line with other URL formats such as
"http". Previously, an empty port number was permissible.
o In SDP section, added a section on how to add and delete
streams in re-INVITEs.
o IETF-blessed extensions now have short names, without
org.ietf. prefix.
o Cseq is unique within a call leg, not just within a call
(Section 22.16).
o Added IPv6 literal addresses to the SIP URL definition,
according to RFC 2732 [45]. Modified the IPv4 address to
limit segments to at most three digits.
o modify registration procedure so that it explicitly references
the URL comparison. Updates with shorter expiration time are
now allowed.
o For send-only media, SDP still must indicate the address and
port, since these are needed as destinations for RTCP
messages.
o Changed references regarding DNS SRV records from RFC 2052 to
RFC 2782, which is now a Proposed Standard. Integrated SRV
into the search procedure and removed the SRV appendix. The
only visible change is that protocol and service names are now
prefixed by an underscore. Added wording that incorporates the
precedence of maddr.
o Allow parameters in Record-Route and Route headers.
o In Table 1, list udp as the default value for the transport
parameter in SIP URI.
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o Removed sentence that From can be encrypted. It cannot, since
the header is needed for call-leg identification.
o Added note that a UAC only copies a To tag into subsequent
transactions if it arrives in a 200 OK to an INVITE. This
avoids the problem that occurs when requests get resubmitted
after receiving, say, a 407 (or possibly 500, 503, 504, 305,
400, 411, 413, maybe even 408). Under the old rules, these
requests would have a tag, which would force the called UAS to
reject the request, since it doesn't have an entry for this
tag.
o Loop detection has been modified to take the request-URI into
account. This allows the same request to visit the server
twice, but with different request URIs ("spiral").
o Elaborated on URL comparison and comparison of From/To fields.
o Added np-queried user parameter.
o Changed tag syntax from UUID to token, since there's no reason
to restrict it to hex.
o Added Content-Disposition header based on earlier discussions
about labeling what to do with a message body (part).
o Clarification: proxies must insert To tags for locally
generated responses.
o Clarification: multicast may be used for subsequent
registrations.
o Feature: Added Supported header. Needed if client wants to
indicate things the server can usefully return in the
response.
o Bug: The From, To, and Via headers were missing extension
parameters. The Encryption and Response-Key header fields now
"officially" allow parameters consisting only of a token,
rather than just "token = value".
o Bug: Allow was listed as optional in 405 responses in Table 2.
It is mandatory.
o Added: "A BYE request from either called or calling party
terminates any pending INVITE, but the INVITE request
transaction MUST be completed with a final response."
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o Clarified: "If an INVITE request for an existing session
fails, the session description agreed upon in the last
successful INVITE transaction remains in force."
o Clarified what happens if two INVITE requests meet each other
on the wire, either traveling the same or in opposite
directions:
A UAC MUST NOT issue another INVITE request for the
same call leg before the previous transaction has
completed. A UAS that receives an INVITE before it
sent the final response to an INVITE with a lower CSeq
number MUST return a 400 (Bad Request) response and
MUST include a Retry-After header field with a
randomly chosen value of between 0 and 10 seconds. A
UA that receives an INVITE while it has an INVITE
transaction pending, returns a 500 (Internal Server
Error) and also includes a Retry-After header field.
o Expires header clarified: limits only duration of INVITE
transaction, not the actual session. SDP does the latter.
o The In-Reply-To header was added.
o There were two incompatible BNFs for WWW-Authenticate. One
defined for PGP, and the other borrowed from HTTP. For basic
or digest:
WWW-Authenticate: basic realm="Wallyworld"
and for pgp:
WWW-Authenticate: pgp; realm="Wallyworld"
The latter is incorrect and the semicolon has been removed.
o Added rules for Route construction from called to calling UA.
o We now allow Accept and Accept-Encoding in BYE and CANCEL
requests. There is no particular reason not to allow them, as
both requests could theoretically return responses,
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particularly when interworking with other signaling systems.
o PGP "pgp-pubalgorithm" allows server to request the desired
public-key algorithm.
o ABNF rules now describe tokens explicitly rather than by
subtraction; explicit character enumeration for CTL, etc.
o Registrars should be careful to check the Date header as the
expiration time may well be in the past, as seen by the
client.
o Content-Length is mandatory; Table 2 erroneously marked it as
optional.
o User-Agent was classified in a syntax definition as a request
header rather than a general header.
o Clarified ordering of items to be signed and include realm in
list.
o Allow Record-Route in 401 and 484 responses.
o Hop-by-hop headers need to precede end-to-end headers only if
authentication is used.
o 1xx message bodies MAY now contain session descriptions.
o Changed references to HTTP/1.1 and authentication to point to
the latest RFCs.
o Added 487 (Request terminated) status response. It is issued
if the original request was terminated via CANCEL or BYE.
o The spec was not clear on the identification of a call leg.
Section 1.3 says it's the combination of To, From, and Call-
ID. However, requests from the callee to the caller have the
To and From reversed, so this definition is not quite
accurate. Additionally, the "tag" field should be included in
the definition of call leg. The spec now says that a call leg
is defined as the combination of local-address, remote-
address, and call-id, where these addresses include tags.
Text was added to Section 6.21 to emphasize that the From and
To headers designate the originator of the request, not that
of the call leg.
o All URI parameters, except method, are allowed in a Request-
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URI. Consequently, also updated the description of which
parameters are copied from 3xx responses in Sec. 22.10.
o The use of CRLF, CR,or LF to terminate lines was confusing.
Basically, each header line can be terminated by a CR, LF, or
CRLF. Furthermore, the end of the headers is signified by a
"double return". Simplified to require sending of CRLF, but
require senders to receive CR and LF as well and only allow CR
CR, LF LF in addition to double CRLF as a header-body
separator.
o Round brackets in Contact header were part of the HTTP legacy,
and very hard to implement. They are also not that useful and
were removed.
o The spec said that a proxy is a back-to-back UAS/UAC. This is
almost, but not quite, true. For example, a UAS should insert
a tag into a provisional response, but a proxy should not.
This was clarified.
o Section 6.13 in the RFC begins mid-paragraph after the BNF.
The following text was misplaced in the conversion to ASCII:
Even if the "display-name" is empty, the "name-addr"
form MUST be used if the "addr-spec" contains a comma,
semicolon or question mark.
29 Changes Made in Version 01
o Uniform syntax specification for semicolon parameters:
Foo = "Foo" ":" something *( ";" foo-param )
foo-param = "bar" "=" token
| generic-param
o Removed np-queried user parameter since this is now part of a
tel URL extension parameter.
o In SDP section, noted that if the capabilities intersection is
empty, a dummy format list still has to be returned due to SDP
syntax constraints. Previously, the text had required that no
formats be listed. (Brian Rosen)
o Reorganized tables 2 and 3 to show proxy interaction with
headers rather than "end-to-end" or "hop-by-hop".
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30 Changes Made in Version 02
o Added "or UAS" in description of received headers in Section
22.40. This makes the response algorithm work even if the last
IP address in the Via is incorrect.
o Tentatively removed restriction that CANCEL requests cannot
have Route headers. (Billy Biggs)
o Tentatively added Also header for BYE requests, as it is
widely implemented and a simple means to implement
unsupervised call transfer. Subject to removal if there is
protest. (Billy Biggs)
o If a proxy sends a request by UDP (TCP), the spec did not
disallow placing TCP (UDP) in the transport parameter of the
Via field, which it should. Added a note that the transport
protocol actually used is included.
o No default value for the q parameter in Contact is defined.
This is not strictly needed, but is useful for consistent
behaviors at recursive proxies and at UAC's. Now 0.5.
o Clarified that To and From tag values should be different to
simplify request matching when calling oneself.
o Removed ability to carry multiple requests in a single UDP
packet (Section 22.14).
o Added note that Allow MAY be included in requests, to indicate
requestor capabilities for the same call ID.
o Added note to Section 22.17 indicating that registrars MUST
include the Date header to accomodate UAs that do not have a
notion of absolute time.
o Added note emphasizing that non-SIP URIs are permissible in
REGISTER.
o Rewrote the server lookup section to be more precise and more
like pseudo-code, with nesting instead of "gotos".
o Removed note
Note that the two URLs example.com and
example.com:5060, while considered equal, may not lead
to the same server, as the former causes a DNS SRV
lookup, while the latter only uses the A record.
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since that is no longer the case.
o Emphasized that proxies have to forward requests with unknown
methods.
o Aligned definition of call leg with URI comparison rules.
o Required that second branch parameter be globally unique, so
that a proxy can distinguish different branches in spiral
scenarios similar to the following, with record-routing in
place:
B ---> P1 -------> P2 ------------> P1 ----------------> A
BYE B B/1 P1/2,B/1 P2/3,P1/2,B/1 P1/4,P2/3,P1/2,B/1
Here, A/1 denotes the Via entry with host A and branch
parameter 1. Also, this requires updating the definition of
isomorphic requests, since the Request-URI is the same for all
BYE that are record-routed.
o Removed Via hiding from spec, for the following reasons:
- complexity, particularly hidden "gotchas" that surface at
various points (as in this instance);
- interference with loop detection and debugging;
- Unlike HTTP, where via-hiding makes sense since all data is
contained in the request or response, Via-hiding in SIP by
itself does nothing to hide the caller or callee, as address
information is revealed in a number of places:
- Contact;
- Route/Record-Route;
- SDP, including the o= and c= lines;
- possibly accidental leakage in User-Agent header and
Call-ID headers.
- Unless this is implemented everywhere, the feature is not
likely to be very useful, without the sender having any
recourse such as "don't route this request unless you can
hide". It appears that almost all existing proxies simply
ignore the Hide header.
Various Authors [Page 200]
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o Added Error-Info header field.
31 Changes Made in Version 03
o Description of Route and Record-Route moved to separate
section, which is new. All UAs must now support this
mechanism.
o Removed status code 411, since it cannot occur (Jonathan
Rosenberg, James Jack).
o Rewrote Record-Route section to reflect new mechanism. In
particular, requests from callee to caller now use the same
path as in the opposite direction, without substituting the
From header field values. The maddr parameter is now optional.
o Disallowed SIP URLs that only have a password, without a user
name. The prototype from RFC 1738 also doesn't allow this.
o Allow registrar to set the expiration time.
o CSeq (Section 22.16) is counted within a call leg, not a call.
o Removed wording that connection closing is equivalent to
CANCEL or 500. This does not work for connections that are
used for multiple transactions and has other problems.
o Cleaned up CSeq section. Removed text about inserting CSeq
method when it is absent. Clarified that CSeq increments for
all requests, not just invite. Clarified that all out of order
requests, not just out of order INVITE, are rejected with a
400 class response. Clarified the meaning of "initial"
sequence number. Clarified that after a request forks, each
200 OK is a separate call leg, and thus, separate CSeq space.
Clarified that CSeq numbers are independent for each direction
of a call leg.
o Massive reorganization and cleanup of the SDP section.
Introduced the concept of the offer-answer model. Clarified
that set of codecs in m line are usable all at the same time.
Inserted size restriction on representation of values in o
line. Explicitly describe forked media. New media lines for
adding streams appear at the bottom of the SDP (used to say
append).
o Removed Also.
o Added text to Require and Proxy-Require sections, making it a
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SHOULD to retry the request without the unsupported extension.
o Added text to section on 415, saying that UAC SHOULD retry the
request without the unsupported body.
o Added text to section on CANCEL and ACK, clarifying much of
the behavior.
o Modified Content-Type to indicate that it can be present even
if the body is empty.
o From tags mandatory
o Old text said that if you hang up before sending an ACK, you
need not send the ACK. That is wrong. Text fixed so that an
ACK is always sent.
o Old text said that if you never got a response to an INVITE,
the UAC should send both an INVITE and CANCEL. This doesn't
make sense. Rahter, it should do nothing and consider the call
terminated.
o Added text that says pending requests are responded to with a
487 if a BYE is received.
o Updated section 2.2, so that its clear that Contact is not
used with BYE.
o Clarified Via processing rules. Added text on handling loops
when proxies route on headers besides the request URI. Added
text on handling case when sent-by contains a domain name.
Added text to 6.47 on opening TCP connections to send
responses upstream.
o Clarified that a 1xx with an unknown xx is not the same as the
100 response.
o Removed usage of Retry-After in REGISTER.
o Clarified usage of persistent connections.
o Clarified that servers supporting HTTP basic or digest in
rfc2617 MUST be backwards compatible with RFC 2069.
o Clarified that ACK contains the same branch ID as the request
its acknowledging.
o Added definitions for spiral, B2BUA.
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o Rephrased definitions for UAC, UAS, Call, call-leg, caller,
callee, making them more concrete.
o URL comparison ignores parameters not present in both URLs
only for unknown parameters.
o Clarified that * in Contact is used only in REGISTER with
Expires header zero. Mentioned * case in section on Contact
syntax.
o Removed text that says a UA can insert a Contact in 2xx that
indicates the address of a proxy. Not likely to work in
general.
o Removed SDP text about aligning media streams within a media
type to handle certain crash and restart cases.
o Receiving a 481 to a mid-call request terminates that call
leg. Agreed upon at IETF 49.
o Introduced definition of regular transaction - non-INVITE
excepting ACK and CANCEL.
o Clarified rules for overlapping transactions.
o Forking proxies MUST be stateful (used to say SHOULD). Proxies
that send requests on multicast MUST be stateful (used to say
nothing)
o Text added recommending that registrars authorize that entity
in From field can register address-of-record in the To field.
o Forwarding of non-100 provisionals upstream in a proxy changed
from SHOULD to MUST.
o Removed PGP.
32 Changes Made in Version 04
o Removed Unsupported as a request header from Table 3.
o Clarified SDP procedures for changing IP address and port.
Specifically, spelled out the duration for which a UA needs to
received media on the old port and address.
o Added text in the SDP session which recommends that the
answerer use the same ordering of codecs as used on the offer,
in order to help ensure symmetric codec operation under normal
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conditions.
o Fixed bug in the example in the SDP section, where the new
media line was listed at the top. Should have been the bottom.
o Authorization credentials are cached based on the URL of the
To header, not the entire To header as 10.48 implied.
o Section 10.31, on Proxy-Authenticate, indicated that a server
responds with a 401 if the client guessed wrong. This is
incorrect. It should be 407.
o Section 10.14, removed motivational text about Contact
allowing an INVITE to be routed directly between end systems,
since its confusing. Some have interpreted to mean that
Record-Route is ignored when Contact is present.
o Added reference to SCTP RFC.
o Updated 2.2 to allow non-SIP URLs in OPTIONS and 2xx to
OPTIONS.
o Fixed example in 20.5. Added ACK for 487, and added To tag to
487 response.
o Clarified further URL comparisons. Its only URL parameters
without defaults that are ignored if not present in both URLs.
o Section 1.5.2, UDP mandatory for all. TCP is a SHOULD for UA,
MUST for proxy, registrar, redirect servers.
o Brought syntax for Contact, Via, and the SIP URL into
alignment between the text and postscript versions.
o Updated the text in section 6 which said that the ordering of
header fields follows HTTP, with the exception of Via, where
order matters. However, the HTTP spec says that order matters,
so this sentence is redundant and confusing. The sentence was
removed.
o Added e lines to SDP examples in the Examples section.
o Rewrote Allow discussion, more formally defining its semantics
and usage cases.
o Updated text on 604 status, to indicate that its based on the
Request-URI, not the To.
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o Added response registrations to IANA considerations. Provided
more details on registration process.
o Clarified that only a UAS rejects a request because the To tag
doesn't match a local value.
o Clarified that stateless proxies need to route based on static
criteria only.
o Proxy and UAC CANCEL generation upon 2xx, 6xx if it forked is
now a SHOULD; used to be a MAY.
o Added text saying that a UAS SHOULD send a BYE if it never
gets an ACK for a 2xx establishing a call leg.
o Added text saying that a UAS SHOULD send a re-INVITE if it
never gets an ACK for a 2xx to a re-INVITE.
o Added text on 503 processing, indicating that a client should
try a different server when receiving a 503, and that a proxy
shouldn't forward a 503 upstream unless it can't service any
other requests.
o Removed motivational text in Section 10.43 on Via headers
since its not consistent with the text before it.
o Changed IPSec reference to RFC2401, from RFC1825.
o Updated retransmission defininition in 17.3.4 to be consistent
with the rest of the spec.
o Softened the language for insertion of the transport param in
the record-route. Specifically, it can be inserted in private
networks where it is known apriori that the specific transport
is supported.
o Updated definition of B2BUA.
o Added text to section on 420 processing, which mandates that
the client retry the request without extensions listed in the
Unsupported header in the response.
o Allow Authentication-Info header to be used for HTTP digest.
33 Changes Made in Version 05
o Updated Table 2 to reflect that Error-Info is a response
header in 3xx-6xx responses (it was previously listed as a
Various Authors [Page 205]
Internet Draft SIP October 26, 2001
request header).
o Removed WWW-Authenticate as a request header from Table 3.
Authentication of responses is now done according to RFC2617.
o Updated the Accept, Accept-Encoding and Accept-Language
sections. More details on precise semantics for the various
requests and responses is now provided. Presence of these
headers is now a SHOULD for INVITE and 2xx to INVITE when a
non-default value is present. Extra emphasis is placed on
including the Accept-Language in INVITE and 2xx in order to
support internationalization. Usage of these three headers in
CANCEL has been removed since it makes no sense.
o Generalized local outbound processing rules in Section 16.4.1
to cover the case where the UAS is using a local outbound
proxy which was not in the initial call setup path.
o Updated record-routing section, so that a proxy can insert a
transport param if it knows that the proxy on one side
supports the specific transport (the previous text required
the proxy to know whether the proxies on both sides supported
the specific transport).
o Added Authentication-Info to Section 10.
o Clarified the meaning of Table 2 for responses.
o Updated Table 1 to reflect that maddr is no longer mandatory
in Record-Route.
o Updated Table 3 so that header fields in responses to ACK are
never listed as optional, mandatory, etc. - only not
applicable. This is because responses to ACK are not allowed.
Also improved wording in Section 5.1.1 to clarify that there
MUST NOT be responses to ACK.
o Updated SRV procedures. Old text said to treat a failure to
contact a server as a 4xx, which would stop the SRV
processing. But, this is not so. Sentence was stricken.
o Updated 12.1 to clarify that 2xx INVITE responses MUST contain
session descriptions.
o Changed User-Agent to a request header in Table 3.
o Updated SDP section, so that a UA cannot change the SDP when
it gets a re-INVITE with no SDP.
Various Authors [Page 206]
Internet Draft SIP October 26, 2001
o Clarified Appendix B that a unicast offer MUST have a unicast
response.
o Clarified that any request can be record-routed, but it may
not be used by the UA, depending on the method.
o non-2xx responses to INVITE no longer retransmitted over TCP.
o Removed lower bound on T1 and T2 in private networks, which
can use lower values. Furthermore, T1 can be smaller on the
public Internet if proper RTT estimation is used.
o UAS Cannot send a BYE for a call leg until it receives ACK, in
order to eliminate a race condition between BYE and 200 OK.
o Support of CR or LF alone as line terminators, as opposed to
CRLF, is no longer required.
o Client behavior on receipt of a 3xx to re-INVITE is now
specified, and it is no longer forbidden to generate a 3xx.
This is needed to maintain the idempotency of INVITE, as a
proxy might redirect without knowing its a 3xx.
o CANCEL cannot be sent before a 1xx is received, in order to
eliminate race condition between request and CANCEL.
o Termination of the client and server transactions is now based
entirely on timeouts, rather than retransmission counters, in
order to unify TCP and UDP behavior. Timeout values scale as a
function of the RTT estimate, defined as T1. For reliable
transports, many of these timers are now set to zero. Many
timeouts differ than in bis-04.
o Added a working RTT estimation algorithm using the Timestamp
header, and specified it to be compliant to RFC 2988.
o UAS accepting requests with unknown schemes in the URI in the
To field is now a RECOMMENDED instead of SHOULD. This reflects
the fact that processing a request when the To field doesn't
match is a matter of policy.
o Bodies are now allowed in any request and response, including
CANCEL, although there may not be any semantics associated
with that.
o Supporting of INVITE without SDP is now a MUST (no strength
was previously specified).
Various Authors [Page 207]
Internet Draft SIP October 26, 2001
o Registration procedures for visiting, which had a few
sentences in bis-04, have been removed. Roaming is a complex
issue, and should be treated elsewhere.
o Bis-04 mandated that a 2xx response to REGISTER contain
expires Contact parameters indicating the expiration time of a
contact. This behavior has now been made consistent with
requests, so that the expiration time of a contact is the same
in either case: the expires param is used first if present,
then the Expires header if present, else one hour for SIP
URLs.
o Action parameter in contact registrations is deprecated.
o 2xx to REGISTER MUST contain current contacts. This was just a
SHOULD in bis-04.
o Multicast operation radically changed. Now, the treatment is
no different than unicast. That is, only the first non-1xx
response to a multicast request will be used. This is a
natural consequence of the layering now applied to the
protocol. This still enables anycast types of functions,
mirroring the real usage of registrar discovery.
o To completely separate transport rules from transaction rules,
the rule in bis-04 that said a UAC SHOULD keep a connection
opened until a response is received, has been turned into a
timer recommendation. Specifically, the spec now says that it
is RECOMMENDED that connections be kept opened for a minimum
interval of sufficient duration to guarantee, with high
probability, that responses are sent over the same connections
as a request.
o Re-use of existing connections for new requests to the same
address and port is now RECOMMENDED, it was only a MAY in
bis-04.
o Modification of headers below the Authorization header by
proxies is no longer disallowed, since the only mechanism that
used Authorization in that way, PGP, has been deprecated
previously.
o Authentication of registrations now RECOMMENDED; no strength
was defined previously.
o Registering of new headers with IANA is now SHOULD; no
strength was defined previously.
Various Authors [Page 208]
Internet Draft SIP October 26, 2001
o Proxy aggregation of challenges now a SHOULD; no strength was
defined previously.
o Server support of basic authentication downgraded from SHOULD
to MAY.
o UAC resubmitting requests with credentials after a challenge
upgraded from MAY to SHOULD.
o TLS is now RECOMMENDED as the transport layer security for SIP
signaling.
o UA recursion on a redirect is now SHOULD; no strength was
assigned previously.
o UA reuse of headers in a recursed request is now SHOULD; no
strength was assigned previously.
o Security considerations added for Call-Info and Alert-Info.
o Proxies no longer forward a 6xx immediately on receiving it.
Instead, they CANCEL pending branches immediately. This avoids
a potential race condition that would result in a UAC getting
a 6xx followed by a 2xx. In all cases except this race
condition, the result will be the same - the 6xx is forwarded
upstream.
o The term call-leg has been eliminated from the spec; a more
generic term, dialog, is used in its place.
o For SRV processing, subsequent requests with the same Call-ID
(as opposed to the same transaction in bis-04) are sent to the
same server.
o SRV processing generalized to deal with the fact that the
default port is transport dependent.
o Per IESG request, draft-ietf-sip-serverfeatures has been
integrated into bis.
o Per IESG request, draft-ietf-sip-100rel will be integrated
into bis. This is marked with a placeholder in this draft.
o The BNF has been converted from implicit LWS to explicit LWS.
o Caching of responses in a proxy to avoid redoing location
server lookups used to be a SHOULD. Caching behavior for
responses is now fully encapsulated in the transaction
Various Authors [Page 209]
Internet Draft SIP October 26, 2001
processing.
o Proxy usage of SRV in processing Route headers upgraded from
SHOULD to MUST.
34 Acknowledgments
We wish to thank the members of the IETF MMUSIC and SIP WGs for their
comments and suggestions. Detailed comments were provided by Brian
Bidulock, Jim Buller, Neil Deason, Dave Devanathan, Cédric Fluckiger,
Yaron Goland, Bernie Höneisen, Phil Hoffer, Christian Huitema, Jean
Jervis, Gadi Karmi, Peter Kjellerstedt, Anders Kristensen, Jonathan
Lennox, Gethin Liddell, Keith Moore, Vern Paxson, Moshe J. Sambol,
Chip Sharp, Igor Slepchin, Robert Sparks, Eric Tremblay., and Rick
Workman.
Brian Rosen provided the compiled BNF.
This work is based, inter alia, on [46,47].
35 Authors' Addresses
Authors addresses are listed alphabetically for the editors, the
writers, and then the original authors of RFC 2543.
Jonathan Rosenberg
dynamicsoft
72 Eagle Rock Ave
East Hanover, NJ 07936
USA
electronic mail: jdrosen@dynamicsoft.com
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
electronic mail: Gonzalo.Camarillo@ericsson.com
Alan Johnston
Various Authors [Page 210]
Internet Draft SIP October 26, 2001
WorldCom
100 South 4th Street
St. Louis, MO 63102
USA
electronic mail: alan.johnston@wcom.com
Jon Peterson
NeuStar, Inc
1800 Sutter Street, Suite 570
Concord, CA 94520
USA
electronic mail: jon.peterson@neustar.com
Robert Sparks
dynamicsoft, Inc.
5100 Tennyson Parkway
Suite 1200
Plano, Texas 75024
USA
electronic mail: rsparks@dynamicsoft.com
Mark Handley
ACIRI
electronic mail: mjh@aciri.org
Eve Schooler
Computer Science Department 256-80
California Institute of Technology
Pasadena, CA 91125
USA
electronic mail: schooler@cs.caltech.edu
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synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of
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Internet Draft SIP October 26, 2001
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Full Copyright Statement
Copyright (c) The Internet Society (2001). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
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the copyright notice or references to the Internet Society or other
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copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
Various Authors [Page 215]
Internet Draft SIP October 26, 2001
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Table of Contents
1 Introduction ........................................ 1
2 Overview of SIP Functionality ....................... 2
3 Terminology ......................................... 3
4 Overview of Operation ............................... 3
5 Structure of the Protocol ........................... 11
6 Definitions ......................................... 13
7 SIP Messages ........................................ 18
7.1 Requests ............................................ 18
7.2 Responses ........................................... 19
7.3 Header Fields ....................................... 20
7.3.1 Header Field Format ................................. 20
7.3.2 Header Field Classification ......................... 23
7.3.3 Compact Form ........................................ 23
7.4 Bodies .............................................. 23
7.4.1 Message Body Type ................................... 23
7.4.2 Message Body Length ................................. 24
7.5 Framing SIP messages ................................ 24
8 General User Agent Behavior ......................... 24
8.1 UAC Behavior ........................................ 25
8.1.1 Generating the Request .............................. 25
8.1.1.1 To .................................................. 25
8.1.1.2 From ................................................ 26
8.1.1.3 Call-ID ............................................. 27
8.1.1.4 CSeq ................................................ 27
8.1.1.5 Via ................................................. 28
8.1.1.6 Contact ............................................. 28
8.1.1.7 Request-URI ......................................... 28
8.1.1.8 Supported and Require ............................... 29
8.1.1.9 Additional Message Components ....................... 29
8.1.2 Sending the Request ................................. 29
8.1.3 Processing Responses ................................ 30
8.1.3.1 Unrecognized Responses .............................. 30
8.1.3.2 Vias ................................................ 30
8.1.3.3 Processing 3xx responses ............................ 30
Various Authors [Page 216]
Internet Draft SIP October 26, 2001
8.1.3.4 Processing 4xx responses ............................ 30
8.2 UAS Behavior ........................................ 31
8.2.1 Authentication/Authorization ........................ 31
8.2.2 Method Inspection ................................... 31
8.2.3 Header Inspection ................................... 32
8.2.3.1 To and Request-URI .................................. 32
8.2.3.2 Require ............................................. 33
8.2.4 Content Processing .................................. 33
8.2.5 Applying Extensions ................................. 34
8.2.6 Processing the Request .............................. 34
8.2.7 Generating the Response ............................. 34
8.3 Redirect Servers .................................... 35
9 Canceling a Request ................................. 36
9.1 Client Behavior ..................................... 37
9.2 Server Behavior ..................................... 38
10 Registrations ....................................... 38
10.1 Overview of Usage ................................... 38
10.2 Construction of the REGISTER request ................ 40
10.2.1 Adding Bindings with REGISTER ....................... 41
10.2.1.1 Setting the Expiration Interval of Contact
Addresses ...................................................... 43
10.2.1.2 Setting Preference among Contact Addresses .......... 43
10.2.2 Removing Bindings with REGISTER ..................... 43
10.2.3 Fetching Bindings with REGISTER ..................... 44
10.2.4 Refreshing Registrations ............................ 44
10.2.5 Discovering a Registrar ............................. 44
10.3 Processing of REGISTER at the Registrar ............. 45
11 Querying for Capabilities ........................... 48
11.1 Construction of OPTIONS Request ..................... 48
11.2 Processing of OPTIONS Request ....................... 49
12 Dialogs ............................................. 50
12.1 Creation of a Dialog ................................ 51
12.1.1 UAS ................................................. 51
12.1.2 UAC ................................................. 53
12.2 Requests within a Dialog ............................ 54
12.2.1 UAC Behavior ........................................ 54
12.2.1.1 Generating the Request .............................. 54
12.2.1.2 Processing the Responses ............................ 55
12.2.2 UAS behavior ........................................ 56
12.3 Termination of a Dialog ............................. 57
13 Initiating a Session ................................ 57
13.1 Overview ............................................ 57
13.2 Caller Processing ................................... 58
13.2.1 Creating the Initial INVITE ......................... 58
13.2.2 Processing INVITE Responses ......................... 60
13.2.2.1 1xx responses ....................................... 60
13.2.2.2 3xx responses ....................................... 60
13.2.2.3 4xx, 5xx and 6xx responses .......................... 61
Various Authors [Page 217]
Internet Draft SIP October 26, 2001
13.2.2.4 2xx responses ....................................... 61
13.3 Callee Processing ................................... 62
13.3.1 Processing of the INVITE ............................ 62
13.3.1.1 Progess ............................................. 64
13.3.1.2 The INVITE is redirected ............................ 64
13.3.1.3 The INVITE is rejected .............................. 64
13.3.1.4 The INVITE is accepted .............................. 64
14 Modifying an Existing Session ....................... 65
14.1 UAC Behavior ........................................ 66
14.2 UAS Behavior ........................................ 66
15 Terminating a Session ............................... 67
15.1 Terminating a Dialog with a BYE ..................... 68
15.1.1 UAC Behavior ........................................ 68
15.1.2 UAS Behavior ........................................ 69
16 Proxy Behavior ...................................... 69
16.1 Overview ............................................ 69
16.2 Stateful Proxy ...................................... 70
16.3 Request Validation .................................. 72
16.4 Making a Routing Decision ........................... 74
16.5 Request Processing .................................. 76
16.6 Response Processing ................................. 82
16.7 Handling transport errors ........................... 87
16.8 CANCEL Processing ................................... 88
16.9 Stateless proxy ..................................... 88
17 Transactions ........................................ 89
17.1 Client transaction .................................. 92
17.1.1 INVITE Client Transaction ........................... 92
17.1.1.1 Overview of INVITE Transaction ...................... 92
17.1.1.2 Formal Description .................................. 93
17.1.1.3 Construction of the ACK Request ..................... 96
17.1.2 non-INVITE Client Transaction ....................... 97
17.1.2.1 Overview of the non-INVITE Transaction .............. 97
17.1.2.2 Formal Description .................................. 97
17.1.3 Matching Responses to Client Transactions ........... 98
17.1.4 Handling Transport Errors ........................... 100
17.2 Server Transaction .................................. 100
17.2.1 INVITE Server Transaction ........................... 100
17.2.2 non-INVITE Server Transaction ....................... 103
17.2.3 Matching Requests to Server Transactions ............ 104
17.3 RTT Estimation ...................................... 104
18 Reliability of Provisional Responses ................ 106
19 Transport ........................................... 106
19.1 Clients ............................................. 107
19.1.1 Sending Requests .................................... 107
19.1.2 Receiving Responses ................................. 108
19.2 Servers ............................................. 108
19.2.1 Receiving Requests .................................. 108
19.2.2 Sending Responses ................................... 109
Various Authors [Page 218]
Internet Draft SIP October 26, 2001
19.3 Framing ............................................. 110
19.4 Error Handling ...................................... 110
20 Security Considerations ............................. 111
20.1 Transport and Network Layer Security ................ 112
20.2 SIP Authentication .................................. 113
20.2.1 Framework ........................................... 113
20.2.2 User to User Authentication ......................... 114
20.2.3 Proxy to User Authentication ........................ 115
20.2.4 Authentication Schemes .............................. 117
20.2.4.1 HTTP Basic .......................................... 117
20.2.4.2 HTTP Digest ......................................... 117
20.3 SIP Encryption ...................................... 118
20.4 Denial of Service ................................... 119
21 Common Message Components ........................... 121
21.1 SIP Uniform Resource Locators ....................... 121
21.1.1 SIP URL components .................................. 121
21.1.2 Character escaping requirements ..................... 124
21.1.3 Example SIP URLs .................................... 125
21.1.4 SIP URL Comparison .................................. 126
21.2 Option Tags ......................................... 128
21.3 Tags ................................................ 128
22 Header Fields ....................................... 129
22.1 Accept .............................................. 131
22.2 Accept-Encoding ..................................... 132
22.3 Accept-Language ..................................... 133
22.4 Alert-Info .......................................... 133
22.5 Allow ............................................... 134
22.6 Authentication-Info ................................. 134
22.7 Authorization ....................................... 134
22.8 Call-ID ............................................. 135
22.9 Call-Info ........................................... 135
22.10 Contact ............................................. 136
22.11 Content-Disposition ................................. 137
22.12 Content-Encoding .................................... 137
22.13 Content-Language .................................... 138
22.14 Content-Length ...................................... 138
22.15 Content-Type ........................................ 139
22.16 CSeq ................................................ 139
22.17 Date ................................................ 139
22.18 Error-Info .......................................... 140
22.19 Expires ............................................. 140
22.20 From ................................................ 141
22.21 In-Reply-To ......................................... 141
22.22 Max-Forwards ........................................ 142
22.23 MIME-Version ........................................ 142
22.24 Organization ........................................ 142
22.25 Priority ............................................ 143
22.26 Proxy-Authenticate .................................. 143
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22.27 Proxy-Authorization ................................. 144
22.28 Proxy-Require ....................................... 144
22.29 Record-Route ........................................ 144
22.30 Require ............................................. 145
22.31 Retry-After ......................................... 145
22.32 Route ............................................... 146
22.33 Server .............................................. 146
22.34 Subject ............................................. 146
22.35 Supported ........................................... 147
22.36 Timestamp ........................................... 147
22.37 To .................................................. 147
22.38 Unsupported ......................................... 148
22.39 User-Agent .......................................... 148
22.40 Via ................................................. 148
22.41 Warning ............................................. 149
22.42 WWW-Authenticate .................................... 151
23 Response Codes ...................................... 151
23.1 Provisional 1xx ..................................... 151
23.1.1 100 Trying .......................................... 151
23.1.2 180 Ringing ......................................... 152
23.1.3 181 Call Is Being Forwarded ......................... 152
23.1.4 182 Queued .......................................... 152
23.1.5 183 Session Progress ................................ 152
23.2 Successful 2xx ...................................... 152
23.2.1 200 OK .............................................. 152
23.3 Redirection 3xx ..................................... 152
23.3.1 300 Multiple Choices ................................ 152
23.3.2 301 Moved Permanently ............................... 153
23.3.3 302 Moved Temporarily ............................... 153
23.3.4 305 Use Proxy ....................................... 153
23.3.5 380 Alternative Service ............................. 154
23.4 Request Failure 4xx ................................. 154
23.4.1 400 Bad Request ..................................... 154
23.4.2 401 Unauthorized .................................... 154
23.4.3 402 Payment Required ................................ 154
23.4.4 403 Forbidden ....................................... 154
23.4.5 404 Not Found ....................................... 154
23.4.6 405 Method Not Allowed .............................. 154
23.4.7 406 Not Acceptable .................................. 155
23.4.8 407 Proxy Authentication Required ................... 155
23.4.9 408 Request Timeout ................................. 155
23.4.10 410 Gone ............................................ 155
23.4.11 413 Request Entity Too Large ........................ 155
23.4.12 414 Request-URI Too Long ............................ 155
23.4.13 415 Unsupported Media Type .......................... 156
23.4.14 420 Bad Extension ................................... 156
23.4.15 421 Extension Required .............................. 156
23.4.16 480 Temporarily Unavailable ......................... 156
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23.4.17 481 Call/Transaction Does Not Exist ................. 157
23.4.18 482 Loop Detected ................................... 157
23.4.19 483 Too Many Hops ................................... 157
23.4.20 484 Address Incomplete .............................. 157
23.4.21 485 Ambiguous ....................................... 157
23.4.22 486 Busy Here ....................................... 158
23.4.23 487 Request Terminated .............................. 158
23.4.24 488 Not Acceptable Here ............................. 158
23.5 Server Failure 5xx .................................. 158
23.5.1 500 Server Internal Error ........................... 158
23.5.2 501 Not Implemented ................................. 158
23.5.3 502 Bad Gateway ..................................... 159
23.5.4 503 Service Unavailable ............................. 159
23.5.5 504 Server Time-out ................................. 159
23.5.6 505 Version Not Supported ........................... 159
23.5.7 513 Message Too Large ............................... 159
23.6 Global Failures 6xx ................................. 160
23.6.1 600 Busy Everywhere ................................. 160
23.6.2 603 Decline ......................................... 160
23.6.3 604 Does Not Exist Anywhere ......................... 160
23.6.4 606 Not Acceptable .................................. 160
24 Locating a SIP Server ............................... 160
24.1 Computing the List of Next Hops ..................... 161
24.1.1 Numeric Destination Address ......................... 162
24.1.2 SRV Resolution of Host Name ......................... 162
24.1.3 Address Record Resolution of Host Name .............. 163
24.2 Contacting the Next Hops ............................ 163
25 Examples ............................................ 165
25.1 Registration ........................................ 165
25.2 Session Setup ....................................... 166
26 Augmented BNF for the SIP Protocol ................. 171
26.1 Basic Rules ......................................... 173
27 IANA Considerations ................................. 188
27.1 Option Tags ......................................... 189
27.2 Warn-Codes .......................................... 190
27.3 Header Field Names .................................. 190
27.4 Method and Response Codes ........................... 191
28 Changes Made in Version 00 .......................... 191
29 Changes Made in Version 01 .......................... 198
30 Changes Made in Version 02 .......................... 199
31 Changes Made in Version 03 .......................... 201
32 Changes Made in Version 04 .......................... 203
33 Changes Made in Version 05 .......................... 205
34 Acknowledgments ..................................... 210
35 Authors' Addresses .................................. 210
36 Bibliography ........................................ 211
Various Authors [Page 221]
