Internet Engineering Task Force MMUSIC WG
Internet Draft Handley/Schulzrinne/Schooler
draft-ietf-mmusic-sip-06.txt ISI/Columbia U./Caltech
June 13, 1998
Expires: November 1998
SIP: Session Initiation Protocol
STATUS OF THIS MEMO
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Distribution of this document is unlimited.
ABSTRACT
Many styles of multimedia conferencing are likely to co-
exist on the Internet, and many of them share the need to
invite users to participate. The Session Initiation
Protocol (SIP) is a simple protocol designed to enable
the invitation of users to participate in such multimedia
sessions. It is not tied to any specific conference
control scheme. In particular, it aims to enable user
mobility by relaying and redirecting invitations to a
user's current location.
This document is a product of the Multi-party Multimedia
Session Control (MMUSIC) working group of the Internet
Engineering Task Force. Comments are solicited and
should be addressed to the working group's mailing list
at confctrl@isi.edu and/or the authors.
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1 Introduction
1.1 Overview of SIP Functionality
The Session Initiation Protocol (SIP) is an application-layer
protocol that can establish, modify and terminate multimedia sessions
or calls. These multimedia sessions include multimedia conferences,
distance learning, Internet telephony and similar applications. SIP
can invite a person to both unicast and multicast sessions; the
initiator does not necessarily have to be a member of the session to
which it is inviting users. Media and participants can be added to an
existing session. SIP can be used to "call" both persons and
"robots", for example, to invite a media storage device to record an
ongoing conference or to invite a video-on-demand server to play a
video into a conference. (SIP does not directly control these
services, however; see RTSP [1].)
SIP can be used to initiate sessions as well as invite members to
sessions that have been advertised and established by other means.
(Sessions may be advertised using multicast protocols such as SAP
[2], electronic mail, news groups, web pages or directories (LDAP),
among others.)
SIP transparently supports name mapping and redirection services,
allowing the implementation of ISDN and Intelligent Network telephony
subscriber services. These facilities also enable personal mobility
services, this is defined as: "Personal mobility is the ability of
end users to originate and receive calls and access subscribed
telecommunication services on any terminal in any location, and the
ability of the network to identify end users as they move. Personal
mobility is based on the use of a unique personal identity (i.e.,
mobility complements terminal mobility, i.e., the ability to maintain
communications when moving a single end system from one network to
another.
SIP supports some or all of five facets of establishing and
terminating multimedia communications:
User location: determination of the end system to be used for
communication;
User capabilities: determination of the media and media parameters to
be used;
User availability: determination of the willingness of the called
party to engage in communications;
Call setup: "ringing", establishment of call parameters at both
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called and calling party;
Call handling: including transfer and termination of calls.
SIP can also initiate multi-party calls using a multipoint control
unit (MCU) or fully-meshed interconnection instead of multicast.
Internet telephony gateways that connect PSTN parties may also use
SIP to set up calls between them.
SIP is designed as part of the overall IETF multimedia data and
control architecture [4] currently incorporating protocols such as
RSVP (RFC 2205 [5]) for reserving network resources, the real-time
transport protocol (RTP) (RFC 1889 [6]) for transporting real-time
data and providing QOS feedback, the real-time streaming protocol
(RTSP) [1] for controlling delivery of streaming media, the session
announcement protocol (SAP) [2] for advertising multimedia sessions
via multicast and the session description protocol (SDP) (RFC 2327
[7]) for describing multimedia sessions, but the functionality and
operation of SIP does not depend on any of these protocols.
SIP may also be used in conjunction with other call setup and
signaling protocols. In that mode, an end system uses SIP protocol
exchanges to determine the appropriate end system address and
protocol from a given address that is protocol-independent. For
example, SIP may be used to determine that the party may be reached
via H.323, obtain the H.245 gateway and user address and then use
H.225.0 to establish the call [8]. In another example, it may be used
to determine that the callee is reachable via the public switched
telephone network (PSTN) and indicate the phone number to be called,
possibly suggesting an Internet-to-PSTN gateway to be used.
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 introduce conference control protocols. SIP
does not allocate multicast addresses.
SIP can invite users to sessions with and without resource
reservation. SIP does not reserve resources, but may convey to the
invited system the information necessary to do this. Quality-of-
service guarantees, if required, may depend on knowing the full
membership of the session; this information may or may not be known
to the agent performing session invitation.
1.2 Terminology
In this document, the key words "MUST", "MUST NOT", "REQUIRED",
"SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
and "OPTIONAL" are to be interpreted as described in RFC 2119 [9] and
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indicate requirement levels for compliant SIP implementations.
1.3 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 2068 [10]). The following terms have
special significance for SIP.
Call: A call consists of all participants in a conference invited by
a common source. A SIP call is identified by a globally unique
call-id (Section 6.12). Thus, if a user is, for example, invited
to the same multicast session by several people, each of these
invitations will be a unique call. A point-to-point Internet
telephony conversation maps into a single SIP call. In a MCU-
based call-in conference, each participant uses a separate call
to invite himself to the MCU.
Call leg: A call leg is identified by the combination of Call-ID, To
and From.
Client: An application program that establishes connections for the
purpose of sending requests. Clients may or may not interact
directly with a human user. User agents and proxies contain
clients (and servers).
Conference: A multimedia session (see below), identified by a common
session description. A conference may have zero or more members
and includes the cases of a multicast conference, a full-mesh
conference and a two-party "telephone call", as well as
combinations of these.
Downstream: Requests sent in the direction from the caller to the
callee.
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.
Initiator, calling party, caller: The party initiating a conference
invitation. Note that the calling party does not have to be the
same as the one creating the conference.
Invitation: A request sent to a user (or service) requesting
participation in a session. A successful SIP invitation consists
of two transactions: an INVITE request followed by an ACK
request.
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Invitee, invited user, called party, callee: The person or service
that the calling party is trying to invite to a conference.
Isomorphic request or response: Two requests or responses are defined
to be isomorphic for the purposes of this document if they have
the same values for the Call-ID, To, From and CSeq header
fields. In addition, requests have to have the same Request-
URI.
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). Location services are offered by location servers.
Location servers may be co-located with a SIP server, but the
manner in which a SIP server requests location services is
beyond the scope of this document.
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 program that acts as both a
server and a client for the purpose of making requests on behalf
of other clients. Requests are serviced internally or by passing
them on, possibly after translation, to other servers. A proxy
must interpret, and, if necessary, rewrite a request message
before forwarding it.
Redirect server: A redirect server is a server that accepts a SIP
request, maps the address into zero or more new addresses and
returns these addresses to the client. Unlike a proxy server ,
it does not initiate its own SIP request. Unlike a user agent
server , it does not accept calls.
Registrar: A registrar is server that accepts REGISTER requests. A
registrar is typically co-located with a proxy or redirect
server.
Ringback: Ringback is the signaling tone produced by the calling
client's application indicating that a called party is being
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alerted (ringing).
Server: A server is an application program that accepts requests in
order to service requests and sends back responses to those
requests. Servers are either proxy, redirect or user agent
servers.
Session: "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, [7]) (A session as defined for SDP may
comprise one or more RTP sessions.) As defined, a callee may 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. A transaction is identified
by the CSeq sequence number (Section 6.16) within a single call
leg The ACK request has the same CSeq number as the
corresponding INVITE request, but comprises a transaction on
its own.
Upstream: Responses sent in the direction from the called client to
the caller.
URL-encoded: A character string encoded according to RFC 1738,
Section 2.2 [11].
User agent client (UAC), calling user agent: A user agent client is a
client application that initiates the SIP request.
User agent server (UAS), called user agent: A user agent server is a
server application that contacts the user when a SIP request is
received and that returns a response on behalf of the user. The
response may accept, reject or redirect the call.
An application program may be capable of acting both as a client and
a server. For example, a typical multimedia conference control
application would act as a client to initiate calls or to invite
others to conferences and as a user agent server to accept
invitations. The properties of the different SIP server types are
summarized in Table 1.
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property redirect proxy user agent
server server server
_______________________________________________________
also acts as client no yes no
return 1xx status yes yes yes
return 2xx status no yes yes
return 3xx status yes yes yes
return 4xx status yes yes yes
return 5xx status yes yes yes
return 6xx status no yes yes
insert Via header no yes no
accept ACK no yes yes
Table 1: Properties of the different SIP server types
1.4 Summary of SIP Operation
This section explains the basic protocol functionality and operation.
Callers and callees are identified by SIP addresses, described in
Section 1.4.1. When making a SIP call, a caller first locates the
appropriate server (Section 1.4.2) and then sends a SIP request
(Section 1.4.3). The most common SIP operation is the invitation
(Section 1.4.4). Instead of directly reaching the intended callee, a
SIP request may be redirected or may trigger a chain of new SIP
requests by proxies (Section 1.4.5). Users can register their
location(s) with SIP servers (Section 4.2.6).
1.4.1 SIP Addressing
SIP addresses contain a user and host part. The user part is a user
name, a civil name or a telephone number. The host part is either a
domain name having a DNS SRV (RFC 2052 [12]), MX (RFC 974 [13], CNAME
or A record (RFC 1035 [14]), or a numeric network address.
A user's SIP address can be obtained out-of-band, can be learned via
existing media agents, can be included in some mailers' message
headers, or can be recorded during previous invitation interactions.
In many cases, the SIP address can be the same as a user's electronic
mail address, but this is not required. SIP can thus leverage off the
domain name system (DNS) to provide a first-stage location
mechanisms.
Examples include:
mjh@metro.isi.edu
watson@bell-telephone.com
root@[193.175.132.42]
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root@193.175.132.42
An address can designate an individual (possibly located at one of
several end systems), the first available person from a group of
individuals or a whole group. The form of the address, e.g.,
sales@example.com , is not sufficient, in general, to determine the
intent of the caller.
If a user or service chooses to be reachable at an address that is
guessable from the person's name and organizational affiliation, the
traditional method of ensuring privacy by having an unlisted "phone"
number is compromised. However, unlike traditional telephony, SIP
offers authentication and access control mechanisms and can avail
itself of lower-layer security mechanisms, so that client software
can reject unauthorized or undesired call attempts.
Since SIP requests and responses may also contain non-SIP addresses,
e.g., telephone numbers, SIP addresses are written as SIP URLs
(Section 2) when used within SIP headers. For example,
sip:info@ietf.org
1.4.2 Locating a SIP Server
A SIP client MUST follow the following steps to resolve the host part
of a callee address. If a client supports only TCP or UDP, but not
both, the client omits the respective address type. If the SIP
address contains a port number, that number is to be used, otherwise,
the default port number 5060 is to be used. The default port number
is the same for UDP and TCP. In all cases, the client first attempts
to contact the server using UDP, then TCP.
A client SHOULD rely on ICMP "Port Unreachable" messages rather than
time-outs to determine that a server is not reachable at a particular
address. (For socket-based programs: For TCP, connect() returns
ECONNREFUSED if there is no server at the designated address; for
UDP, the socket should be bound to the destination address using
connect() rather than sendto() or similar so that a second write()
fails with ECONNREFUSED. )
If the SIP address contains a numeric IP address, the client contacts
the SIP server at that address. Otherwise, the client follows the
steps below.
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1. If there is a SRV DNS resource record (RFC 2052 [12]) of
type sip.udp, contact the listed SIP servers in the order
of the preference values contained in those resource
records, using UDP as a transport protocol at the port
number given in the URL or, if none provided, the one
listed in the DNS resource record.
2. If there is a SRV DNS resource record (RFC 2052 [12]) of
type sip.tcp, contact the listed SIP servers in the order
of the preference value contained in those resource
records, using TCP as a transport protocol at the port
number given in the URL or, if none provided, the one
listed in the DNS resource record.
3. If there is a DNS MX record (RFC 974 [13]), contact the
hosts listed in their order of preference at the port
number listed in the URL or the default SIP port number if
none. For each host listed, first try to contact the SIP
server using UDP, then TCP.
4. Finally, check if there is a DNS CNAME or A record for the
given host and try to contact a SIP server at the one or
more addresses listed, again trying first UDP, then TCP.
If all of the above methods fail to locate a server, the caller MAY
contact an SMTP server at the user's host and use the SMTP EXPN
command to obtain an alternate address and repeat the steps above. As
a last resort, a client MAY choose to deliver the session description
to the callee using electronic mail.
A client MAY cache the result of the reachability steps for a
particular address and retry that host address for the next call. If
the client does not find a SIP server at the cached address, it MUST
start the search at the beginning of the sequence.
This sequence is modeled after that described for SMTP,
where MX records are to be checked before A records (RFC
1123 [15]).
1.4.3 SIP Transaction
Once the host part has been resolved to a SIP server, the client
sends one or more SIP requests to that server and receives one or
more responses from the server. A request (and its retransmissions)
together with the responses triggered by that request make up a SIP
transaction. The ACK request following an INVITE is not part of
the transaction since it may traverse a different set of hosts.
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If TCP is used, request and responses within a single SIP transaction
are carried over the same TCP connection (see Section 10). Several
SIP requests from the same client to the same server may use the same
TCP connection or may open a new connection for each request.
If the client sent the request via unicast UDP, the response is sent
to the address contained in the next Via header field (Section 6.40)
of the response. If the request is sent via multicast UDP, the
response is directed to the same multicast address and destination
port. For UDP, reliability is achieved using retransmission (Section
10).
The SIP message format and operation is independent of the transport
protocol.
1.4.4 SIP Invitation
A successful SIP invitation consists of two requests, INVITE
followed by ACK. The INVITE (Section 4.2.1) request asks the callee
to join a particular conference or establish a two-party
conversation. After the callee has agreed to participate in the call,
the caller confirms that it has received that response by sending an
ACK (Section 4.2.2) request. If the caller no longer wants to
participate in the call, it sends a BYE request instead of an ACK.
The INVITE request typically contains a session description, for
example written in SDP (RFC 2327, [7]) format, that provides the
called party with enough information to join the session. For
multicast sessions, the session description enumerates the media
types and formats that may be distributed to that session. For a
unicast session, the session description enumerates the media types
and formats that the caller is willing to receive and where it wishes
the media data to be sent. In either case, if the callee wishes to
accept the call, it responds to the invitation by returning a similar
description listing the media it wishes to receive. For a multicast
session, the callee should only return a session description if it is
unable to receive the media indicated in the caller's description or
wants to receive data via unicast.
The protocol exchanges for the INVITE method are shown in Fig. 1 for
a proxy server and in Fig. 2 for a redirect server. In Fig. 1, the
proxy server accepts the INVITE request (step 1), contacts the
location service with all or parts of the address (step 2) and
obtains a more precise location (step 3). The proxy server then
issues a SIP INVITE request to the address(es) returned by the
location service (step 4). The user agent server alerts the user
(step 5) and returns a success indication to the proxy server (step
6). The proxy server then returns the success result to the original
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caller (step 7). The receipt of this message is confirmed by the
caller using an ACK request, which is forwarded to the callee (steps
8 and 9). All requests and responses have the same Call-ID.
+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning | :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | 4: INVITE 5: ring :
: cz@cs.tu-berlin.de ========================>(~~~~~~)=========>(~~~~~~) :
: <........................( )<.........( ) :
: : 7: 200 OK : ( )6: 200 OK ( ) :
: : : ( tune ) ( play ) :
: : 8: ACK : ( )9: ACK ( ) :
: ========================>(~~~~~~)=========>(~~~~~~) :
+.....................+ +...............................+
====> SIP request
....> SIP response
----> non-SIP protocols
Figure 1: Example of SIP proxy server
The redirect server shown in Fig. 2 accepts the INVITE request (step
1), contacts the location service as before (steps 2 and 3) and,
instead of contacting the newly found address itself, returns the
address to the caller (step 4), which is then acknowledged via an
ACK request (step 5). The caller issues a new request, with the same
call-ID but a higher CSeq, to the address returned by the first
server (step 6). In the example, the call succeeds (step 7). The
caller and callee complete the handshake with an ACK (step 8).
The next section discusses what happens if the location service
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+....... cs.columbia.edu .......+
: :
: (~~~~~~~~~~) :
: ( location ) :
: ( service ) :
: (~~~~~~~~~~) :
: ^ | :
: | hgs@play :
: 2| 3| :
: | | :
: henning| :
+.. cs.tu-berlin.de ..+ 1: INVITE : | | :
: : henning@cs.col: | | :
: cz@cs.tu-berlin.de =======================>(~~~~~~) :
: | ^ | <.......................( ) :
: | . | : 4: 302 Moved : ( ) :
: | . | : hgs@play : ( tune ) :
: | . | : : ( ) :
: | . | : 5: ACK : ( ) :
: | . | =======================>(~~~~~~) :
: | . | : : :
+.......|...|.........+ : :
| . | : :
| . | : :
| . | : :
| . | : :
| . | 6: INVITE hgs@play.cs.columbia.edu (~~~~~~) :
| . ==================================================> ( ) :
| ..................................................... ( ) :
| 7: 200 OK : ( play ) :
| : ( ) :
| 8: ACK : ( ) :
======================================================> (~~~~~~) :
+...............................+
====> SIP request
....> SIP response
----> non-SIP protocols
Figure 2: Example of SIP redirect server
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returns more than one possible alternative.
1.4.5 Locating a User
A callee may move between a number of different end systems over
time. These locations can be dynamically registered with the SIP
server (Sections 1.4.7, 4.2.6). A location server may also use one or
more other protocols, such as finger (RFC 1288 [16]), rwhois (RFC
2167 [17]), LDAP (RFC 1777 [18]), multicast-based protocols or
operating-system dependent mechanism to actively determine the end
system where a user might be reachable. A location server may return
several locations because the user is logged in at several hosts
simultaneously or because the location server has (temporarily)
inaccurate information. The SIP server combines the results to yield
a list of a zero or more locations. It is recommended that each
location server sorts results according to the likelihood of success.
The action taken on receiving a list of locations varies with the
type of SIP server. A SIP redirect server returns the list to the
client sending the request as Location headers (Section 6.22). A SIP
proxy server can sequentially or in parallel try the addresses until
the call is successful (2xx response) or the callee has declined the
call (6xx response). With sequential attempts, a proxy server can
implement an "anycast" service.
If a proxy server forwards a SIP request, it MUST add itself to the
end of the list of forwarders noted in the Via (Section 6.40)
headers. The Via trace ensures that replies can take the same path
back, ensuring correct operation through compliant firewalls and
avoiding request loops. On the response path, each host MUST remove
its Via, so that routing internal information is hidden from the
callee and outside networks. When a multicast request is made, first
the host making the request, then the multicast address itself are
added to the path. A proxy server MUST check that it does not
generate a request to a host listed in the Via list. (Note: If a
host has several names or network addresses, this may not always
work. Thus, each host also checks if it is part of the Via list.)
A SIP invitation may traverse more than one SIP proxy server. If one
of these "forks" the request, i.e., issues more than one request in
response to receiving the invitation request, it is possible that a
client is reached, independently, by more than one copy of the
invitation request. Each of these copies bears the same Call-ID. The
user agent MUST return the appropriate status response. Duplicate
requests are not an error, so there is no need to alert the user.
1.4.6 Changing an Existing Session
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In some circumstances, it may be necessary to change the parameters
of an existing session. For example, two parties may have been
conversing and then want to add a third party, switching to multicast
for efficiency. One of the participants invites the third party with
the new multicast address and simultaneously sends an INVITE to the
second party, with the new multicast session description, but with
the old call identifier.
1.4.7 Registration Services
The REGISTER request allows a client to let a proxy or redirect
server know which address(es) it may be reached under. A client may
also use it to install call handling features at the server.
1.5 Protocol Properties
1.5.1 Minimal State
A single conference session or call may involve one or more SIP
request-response transactions. Proxy servers do not have to keep
state for a particular call, however, they MAY maintain state for a
single SIP transaction, as discussed in Section 11.
For efficiency, a server may cache the results of location service
requests.
1.5.2 Lower-Layer-Protocol Neutral
SIP makes minimal assumptions about the underlying transport and
network-layer protocols. The lower-layer may provide either a packet
or a byte stream service, with reliable or unreliable service.
In an Internet context, SIP is able to utilize both UDP and TCP as
transport protocols, among others. UDP allows the application to more
carefully control the timing of messages and their retransmission, to
perform parallel searches without requiring TCP connection state for
each outstanding request, and to use multicast. Routers can more
readily snoop SIP UDP packets. TCP allows easier passage through
existing firewalls, and given the similar protocol design, allows
common servers for SIP, HTTP and the Real Time Streaming Protocol
(RTSP) [1].
When TCP is used, SIP can use one or more connections to attempt to
contact a user or to modify parameters of an existing conference.
Different SIP requests for the same SIP call may use different TCP
connections or a single persistent connection, as appropriate.
User agents SHOULD implement both UDP and TCP transport, proxy and
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redirect servers MUST.
For concreteness, this document will only refer to Internet
protocols. However, SIP may also be used directly with protocols
such as ATM AAL5, IPX, frame relay or X.25. The necessary naming
conventions are beyond the scope of this document.
1.5.3 Text-Based
SIP is text-based, using ISO 10646 in UTF-8 encoding throughout. This
allows easy implementation in languages such as Java, Tcl and Perl,
allows easy debugging, and most importantly, makes SIP flexible and
extensible. As SIP is used for initiating multimedia conferences
rather than delivering media data, it is believed that the additional
overhead of using a text-based protocol is not significant.
2 SIP Uniform Resource Locators
SIP URLs are used within SIP messages to indicate the originator,
current destination and final recipient of a SIP request, and to
specify redirection addresses. A SIP URL can also be embedded in web
pages or other hyperlinks to indicate that a user or service may be
called.
Because interaction with some resources may require message headers
or message bodies to be specified as well as the SIP address, the SIP
URL scheme is defined to allow setting SIP request-header fields and
the SIP message-body. (This is similar to the mailto: URL [19].)
A SIP URL follows the guidelines of RFC 1630, as revised, [20,21] and
takes the following form:
SIP-URL = "sip:" [ userinfo "@" ] hostport
url-parameters [ headers ]
url-parameters = *( ";" url-parameter )
url-parameter = transport-param | user-param
| ttl-param | maddr-param | tag-param | other-param
transport-param = "transport=" ( "udp" | "tcp" )
ttl-param = "ttl=" ttl
ttl = 1*3DIGIT ; 0 to 255
maddr-param = "maddr=" maddr
maddr = IPv4address ; multicast address
user-param = "user=" ( "phone" )
tag-param = "tag=" UUID
other-param = *uric
headers = "?" header *( "&" header )
header = hname "=" hvalue
hname = *uric
hvalue = *uric
digits = 1*DIGIT
Internet Draft SIP June 13, 1998
Note that all URL reserved characters MUST be encoded. The special
hname "body" indicates that the associated hvalue is the message-
body of the SIP INVITE request. Within sip URLs, the characters
"?", "=", "&" are reserved.
The mailto: URL and RFC 822 email addresses require that numeric
host addresses ("host numbers") are enclosed in square brackets
(presumably, since host names might be numeric), while host numbers
without brackets are used for all other URLs. The SIP URL requires
the latter form.
The elements userinfo, uric, hostport, IPv4address are defined in
[21].
The SIP scheme MAY use the format "user:password" in the userinfo
field. The use of passwords in the userinfo is NOT RECOMMENDED,
because the passing of authentication information in clear text (such
as URI) has proven to be a security risk in almost every case where
it has been used.
If the host is an Internet telephony gateway, the userinfo field can
also encode a telephone number using the notation of telephone-
subscriber defined in [22]. The telephone number is a special case of
a user name and cannot be distinguished by a BNF. Thus, a URL
parameter, user, is added to distinguish telephone numbers from user
names. The phone identifier is to be used when connecting to a
telephony gateway. Even without this parameter, recipients of SIP
URLs MAY interpret the pre-@ part as a phone number if local
restrictions on the name space for user name allow to make this
determination.
The tag parameter allows to distinguish several instances of a user
that share the same host and port values, for example, where these
designate a firewall. The tag value is a version-1 (time-based) or
version-4 (random) UUID [31]. The tag value is designed to be
globally unique within each Call-ID and only to be used within the
same Call-ID. It SHOULD NOT be included in long-lived SIP URLs,
e.g., those found on web pages or user databases. A single user
maintains the same tag throughout the call identified by the Call-ID.
If a server handles SIP addresses for another domain, it MUST URL-
encode the "@" character (%40).
SIP URLs can define specific parameters of the request, including the
transport mechanism (UDP or TCP) and the use of multicast to make a
request. These parameters are added after the host and are separated
by semi-colons. For example, to specify to call j.doe@big.com using
multicast to 239.255.255.1 with a ttl of 15, the following URL would
be used:
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The transport protocol UDP is to be assumed when a multicast address
is given.
Examples of SIP URLs are:
sip:j.doe@big.com
sip:j.doe:secret@big.com;transport=tcp
sip:j.doe@big.com?subject=project
sip:+1-212-555-1212:1234@gateway.com;user=phone
sip:1212@gateway.com
sip:alice@10.1.2.3
sip:alice@example.com;tag=f81d4fae-7dec-11d0-a765-00a0c91e6bf6
sip:alice
Within a SIP message, URLs are used to indicate the source and
intended destination of a request, redirection addresses and the
current destination of a request. Normally all these fields will
contain SIP URLs.
SIP URLs are case-insensitive, so that sip:j.doe@example.com and
SIP:J.Doe@Example.com are equivalent. All URL parameters are included
when comparing SIP URLs for equality.
In some circumstances a non-SIP URL may be used in a SIP message. An
example might be making a call from a telephone which is relayed by a
gateway onto the internet as a SIP request. In such a case, the
source of the call is really the telephone number of the caller, and
so a SIP URL is inappropriate and a phone URL might be used instead.
To allow for this flexibility, SIP headers that specify user
addresses allow these addresses to be SIP and non-SIP URLs.
Clearly not all URLs are appropriate to be used in a SIP message as a
user address. The correct behavior when an unknown scheme is
encountered by a SIP server is defined in the context of each of the
header fields that use a SIP URL.
3 SIP Message Overview
SIP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [23]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by
themselves as line terminators.
Except for the above difference in character sets, much of the
message syntax is identical to HTTP/1.1, rather than repeating it
here we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1
specification (RFC 2068 [10]). In addition, we describe SIP in both
prose and an augmented Backus-Naur form (BNF) [H2.1] described in
detail in RFC 2234 [24].
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transactions can be carried in a single TCP connection or UDP
datagram. UDP datagrams, including all headers, should not normally
be larger than the path maximum transmission unit (MTU) if the MTU is
known, or 1400 bytes if the MTU is unknown.
The 1400 bytes accommodates lower-layer packet headers
within the "typical" MTU of around 1500 bytes. Recent
studies [25] 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
[26]). Thus, another reasonable value would be a message
size of 950 bytes, to accommodate packet headers within the
SLIP MTU without fragmentation.
A SIP message is either a request from a client to a server, or a
response from a server to a client.
SIP-message ___ Request | Response
Both Request (section 4) and Response (section 5) messages use the
generic-message format of RFC 822 [27] for transferring entities (the
body of the message). Both types of message consist of a start-line,
one or more header fields (also known as "headers"), an empty line
(i.e., a line with nothing preceding the carriage-return line-feed (
CRLF)) indicating the end of the header fields, and an optional
message-body. To avoid confusion with similar-named headers in HTTP,
we refer to the header describing the message body as entity headers.
These components are described in detail in the upcoming sections.
generic-message = start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line | Section 4.1
Status-Line Section 5.1
message-header = *( general-header
| request-header
| response-header
| entity-header )
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general-header = Call-ID ; Section 6.12
| CSeq ; Section 6.16
| Date ; Section 6.17
| Encryption ; Section 6.18
| Expires ; Section 6.19
| From ; Section 6.20
| Record-Route ; Section 6.30
| Timestamp ; Section 6.36
| To ; Section 6.37
| Via ; Section 6.40
entity-header = Content-Encoding ; Section 6.13
| Content-Length ; Section 6.14
| Content-Type ; Section 6.15
request-header = Accept ; Section 6.7
| Accept-Encoding ; Section 6.8
| Accept-Language ; Section 6.9
| Authorization ; Section 6.11
| Hide ; Section 6.21
| Location ; Section 6.22
| Max-Forwards ; Section 6.23
| Organization ; Section 6.24
| Priority ; Section 6.25
| Proxy-Authorization ; Section 6.27
| Proxy-Require ; Section 6.28
| Route ; Section 6.32
| Require ; Section 6.29
| Response-Key ; Section 6.31
| Subject ; Section 6.35
| User-Agent ; Section 6.39
response-header = Allow ; Section 6.10
| Location ; Section 6.22
| Proxy-Authenticate ; Section 6.26
| Retry-After ; Section 6.33
| Server ; Section 6.34
| Unsupported ; Section 6.38
| Warning ; Section 6.41
| WWW-Authenticate ; Section 6.42
Table 2: SIP headers
In the interest of robustness, any leading empty line(s) MUST be
ignored. In other words, if the Request or Response message begins
with a CRLF, the CRLF should be ignored.
4 Request
The Request message format is shown below:
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Request = Request-Line ; Section 4.1
*( general-header
| request-header
| entity-header )
CRLF
[ message-body ] ; Section 8
4.1 Request-Line
The 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 final CRLF sequence.
Request-Line ___ Method SP Request-URI SP SIP-Version CRLF
4.2 Methods
The methods are defined below. Methods that are not supported by a
proxy or redirect server are treated by that server as if they were
an INVITE method and forwarded accordingly.
Methods that are not supported by a user agent server cause a 501
(Not Implemented) response to be returned (Section 7).
Method = "ACK" | "BYE" | "CANCEL" | "INVITE"
| "OPTIONS" | "REGISTER"
4.2.1 INVITE
The INVITE method indicates that the user or service is being
invited to participate in a session. The message body contains a
description of the session to which the callee is being invited. For
two-party calls, the caller indicates the type of media it is able to
receive as well as their parameters such as network destination. If
the session description format allows this, it may also indicate
"send-only" media. A success response indicates in its message body
which media the callee wishes to receive.
A server MAY automatically respond to an invitation for a conference
the user is already participating in, identified either by the SIP
Call-ID or a globally unique identifier within the session
description, with a 200 (OK) response.
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same Call-ID, 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. It MUST also
inspect any other header fields for changes and act accordingly. 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 may 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.)
This method MUST be supported by a SIP server and client.
4.2.2 ACK
The ACK request confirms that the client has received a final
response to an INVITE request. ( ACK is used only with INVITE
requests.) 2xx responses are acknowledged by client user agents, all
other final responses by the first proxy or client user agent to
receive the response. The Via is always initialized to the host that
originates the ACK request, i.e., the client user agent after a 2xx
response or the first proxy to receive a non-2xx final response. The
ACK request is forwarded as the corresponding INVITE request, based
on its Request-URI. See Section 10 for details. This method MUST be
supported by a SIP server and client.
The ACK request MAY contain a message body with the final session
description to be used by the callee. If the ACK message body is
empty, the callee uses the session description in the INVITE
request.
4.2.3 OPTIONS
The client is being queried as to its capabilities. A server that
believes it can contact the user, such as a user agent where the user
is logged in and has been recently active, MAY respond to this
request with a capability set. Support of this method is REQUIRED.
A called user agent MAY return a status reflecting how it would have
responded to an invitation, e.g., 600 (Busy).
4.2.4 BYE
The user agent client uses BYE to indicate to the server that it
wishes to abort the call. A BYE request is forwarded like an INVITE
request. It terminates any on-going searches for the named call. A
caller SHOULD issue a BYE request before aborting a call ("hanging
up"). Note that a BYE request may also be issued by the callee.
If the INVITE request contained a Location header, the callee sends
the BYE request to that address rather than the From address.
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supported by all other SIP server types.
4.2.5 CANCEL
The CANCEL request cancels any pending searches, but does not
terminate an accepted call at a particular user agent. (A call is
considered accepted if the callee has returned a 200 (OK) status
response.) Any client MAY issue a CANCEL request at any time. A
proxy, in particular, MAY choose to send a CANCEL to destinations
that have not yet returned a final response after it has received a
2xx or 6xx response for one or more of the parallel-search requests.
A proxy that receives a CANCEL request forwards the request to all
destinations with pending requests triggered by an INVITE. The
Call-ID, To and From in the CANCEL request are identical to those
contained in the INVITE request, but the Via header field is
initialized to the proxy issuing the CANCEL request.
Once a user agent server has received a CANCEL, it MUST NOT issue a
2xx response for the cancelled invitation.
A redirect server or user agent server returns 200 (OK) if the Call-
ID exists and 481 (Invalid Call-ID) if not, but takes no further
action. In particular, any existing call is unaffected.
The BYE request cannot be used to cancel branches of a
parallel search, since several branches may, through
intermediate proxies, find the same user agent server and
then terminate the call.
This method MUST be supported by proxy servers and SHOULD be
supported by all other SIP server types.
4.2.6 REGISTER
A client uses the REGISTER method to register the address listed in
the To header to a SIP server.
A user agent SHOULD register with a local server on startup by
sending a REGISTER request to the well-known "all SIP servers"
multicast address, 224.0.1.75, with a time-to-live value of 1.
SIP user agents on the same subnet MAY listen to that address and use
it to become aware of the location of other local users [28];
however, they do not respond to the request.
The REGISTER request interprets header fields as follows. We define
"address-of-record" as the SIP address that the registry knows the
registrand under, typically of the form "user@domain" rather than
"user@host". In third-party registration, the entity issuing the
request is different from the entity being registered.
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registration is to be created or updated.
From: The From header field contains the address-of-record of the
person responsible for the registration. For first-party
registration, it is identical to the To header field value.
Request-URI: The Request-URI names the destination of the
registration request, i.e., the domain of the registrar. The
user name MUST be empty. Generally, the domains in the
Request-URI and the To header have the same value; however, it
is possible to register as a "visitor", while maintaining one's
name. For example, a traveller sip:alice@acme.com may register
under sip:@atlanta.ayh.org , with the former as the To field and
the latter as the Request-URI. The request is no longer
forwarded once it reached the server whose authoritative domain
is the one listed in the Request-URI.
Location: If the request contains a Location header field, requests
for the Request-URI will also be directed to the address(es)
given. It is recommended that user agents include both SIP UDP
and TCP addresses in their registration. Registrations are
additive.
We cannot require that registration and requests use the
same transport protocol, as multicast registrations may be
quite useful.
Otherwise, future call control requests will be directed to the
network source address of the REGISTER request, using the To
address in the REGISTER request as the Request-URI. If the
registration is changed while a user agent or proxy server processes
an invitation, the new information should be used.
This allows a service known as "directed pick-up".
After registration, the server MAY forward incoming SIP requests to
the network source address and port that originated the registration
request. A server SHOULD silently drop the registration after one
hour, unless refreshed by the client. A client may request a lower or
higher refresh interval through the Expires header (Section 6.19).
Based on this request and its configuration, the server chooses the
expiration interval and indicates it through the Expires header in
the response. A single address (if host-independent) may be
registered from several different clients.
A client cancels an existing registration by sending a REGISTER
request with an expiration time ( Expires) of zero seconds for a
particular Location or the wildcard Location designated by a "*"
for all registrations.
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Beyond its use as a simple location service, this method is
needed if there are several SIP servers on a single host,
so that some cannot use the default port number. Each such
server would register with a server for the administrative
domain. Since a client may not have easy access to the host
address or port number, using the source address and port
from the request itself seems simpler.
Support of this method is RECOMMENDED.
4.3 Request-URI
The Request-URI field is a SIP URL as described in Section 2 or a
general URI. It indicates the user or service to which this request
is being addressed. Unlike the To field, the Request-URI field may
be re-written by proxies. For example, a proxy may perform a lookup
on the contents of the To field to resolve a username from a mail
alias, and then use this username as part of the Request-URI field
of requests it generates.
The host part of the Request-URI typically agrees with one of the
host names of the server. If it does not, the server SHOULD proxy the
request to the address indicated or return a 404 (Not Found) response
if it is unwilling or unable to do so. The case where the Request-URI
and server host name disagrees occurs, for example, for a firewall
proxy that handles outgoing calls. It is similar to the operation of
HTTP proxies.
If a SIP server receives a request with a URI indicating a scheme
other than SIP which that server does not understand, the server MUST
return a 400 (Bad Request) response. It MUST do this even if the To
field contains a scheme it does understand.
4.3.1 SIP Version
Both request and response messages include the version of SIP in use,
and basically follow [H3.1], with HTTP replaced by SIP. To be
conditionally compliant with this specification, applications sending
SIP messages MUST include a SIP-Version of "SIP/2.0".
4.4 Option Tags
Option tags are unique identifiers used to designate new options in
SIP. These tags are used in Require (Section 6.29) and Unsupported
(Section 6.38) fields.
Syntax:
option-tag ___ 1*urlc
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a reverse domain name or register the new option with the Internet
Assigned Numbers Authority (IANA). For example,
"com.foo.mynewfeature" is an apt name for a feature whose inventor
can be reached at "foo.com". Options registered with IANA have the
prefix "org.ietf.sip.", options described in RFCs have the prefix
"org.ietf.rfc.N", where N is the RFC number. Option tags are case-
insensitive.
4.4.1 Registering New Option Tags with IANA
When registering a new SIP option, the following information should
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 NOT contain any spaces, control characters or periods.
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 For proprietary options, contact information (postal and email
address);
Borrowed from RTSP and the RTP AVP.
5 Response
After receiving and interpreting a request message, the recipient
responds with a SIP response message. The response message format is
shown below:
Response = Status-Line ; Section 5.1
*( general-header
| response-header
| entity-header )
CRLF
[ message-body ] ; Section 8
[H6] applies except that HTTP-Version is replaced by SIP-Version.
Also, SIP defines additional response codes and does not use some
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5.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version (Section 4.3.1) 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.
Status-Line ___ SIP-version SP Status-Code SP Reason-Phrase CRLF
5.1.1 Status Codes and Reason Phrases
The Status-Code is a 3-digit integer result code that indicates the
outcome of the attempt to understand and satisfy the 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. The client
is not required to examine or display the Reason-Phrase.
We provide an overview of the Status-Code below, and provide full
definitions in Section 7. The first digit of the Status-Code defines
the class of response. The last two digits do not have any
categorization role. SIP/2.0 allows 6 values for the first digit:
1xx: Informational -- request received, continuing process;
2xx: Success -- the action was successfully received, understood, and
accepted;
3xx: Redirection -- further action must 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 is invalid at any server.
Presented below are the individual values of the numeric response
codes, and an example set of corresponding reason phrases for
SIP/2.0. These reason phrases are only recommended; they may be
replaced by local equivalents without affecting the protocol. Note
that SIP adopts many HTTP/1.1 response codes. SIP/2.0 adds response
codes in the range starting at x80 to avoid conflicts with newly
defined HTTP response codes, and extends these response codes in the
6xx range.
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Status-Code = Informational Fig. 3
| Success Fig. 3
| Redirection Fig. 4
| Client-Error Fig. 5
| Server-Error Fig. 6
| Global-Failure Fig. 7
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
Informational = "100" ; Trying
| "180" ; Ringing
| "181" ; Call Is Being Forwarded
| "182" ; Queued
Success = "200" ; OK
Figure 3: Informational and success status codes
Redirection = "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "305" ; Use Proxy
| "380" ; Alternative Service
Figure 4: Redirection status codes
SIP response codes are extensible. SIP applications are not required
to understand the meaning of all registered response codes, though
such understanding is obviously desirable. However, applications MUST
understand the class of any response code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 response code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if a client
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. In
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Client-Error = "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "407" ; Proxy Authentication Required
| "408" ; Request Timeout
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "413" ; Request Message Body Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "420" ; Bad Extension
| "480" ; Temporarily not available
| "481" ; Invalid Call-ID
| "482" ; Loop Detected
| "483" ; Too Many Hops
Figure 5: Client error status codes
Server-Error = "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Timeout
| "505" ; SIP Version not supported
Figure 6: Server error status codes
include human-readable information which will explain the unusual
status.
6 Header Field Definitions
SIP header fields are similar to HTTP header fields in both syntax
and semantics [H4.2, H14]. In general the ordering of the header
fields is not of importance (with the exception of Via fields, see
below), but proxies MUST NOT reorder or otherwise modify header
fields other than by adding a new Via or other hop-by-hop field.
Proxies MUST NOT, for example, change how header fields are broken
across lines. This allows an authentication field to be added after
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Global-Failure | "600" ; Busy
| "603" ; Decline
| "604" ; Does not exist anywhere
| "606" ; Not Acceptable
Figure 7: Global failure status Codes
The header fields required, optional and not applicable for each
method are listed in Table 3. The table uses "o" to indicate
optional, "m" mandatory and "-" for not applicable. A "*" indicates
that the header fields are needed only if message body is not empty:
The Content-Type and Content-Length headers are required when there
is a valid message body (of non-zero length) associated with the
message (Section 8).
The "type" column describes the request and response types the header
field may be used for. A numeric value indicates the status code for
a response, while "R" refers to any request header, "r" to any
response header. "g" and "e" designate general (Section 6.1) and
entity header (Section 6.2) fields, respectively.
The "enc." column describes whether this message header may be
encrypted end-to-end. A "n" designates fields that MUST NOT be
encrypted, while "c" designates fields that SHOULD be encrypted if
encryption is used.
The "e-e" column has a value of "e" for end-to-end and a value of "h"
for hop-by-hop headers.
Other headers may be added as required; a server MAY ignore optional
headers that it does not understand. A compact form of these header
fields is also defined in Section 9 for use over UDP when the request
has to fit into a single packet and size is an issue.
Table 4 in Appendix A indicates which system components should be
capable of parsing which header fields.
6.1 General Header Fields
There are a few header fields that have general applicability for
both request and response messages. These header fields apply only to
the message being transmitted.
General-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields may be given the semantics of general
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type enc. e-e ACK BYE CAN INV OPT REG
________________________________________________________________________________
Accept R e o o o o o o
Accept-Encoding R e o o o o o o
Accept-Language R n e o o o o o o
Allow 405 e o o o o o o
Authorization R e o o o o o o
Call-ID g n e m m m m o -
Content-Encoding e e * - - * * *
Content-Length e e m - - m m m
Content-Type e e * - - * * *
CSeq g n e m m m m m o
Date g e o o o o o o
Encryption g n e o o o o o o
Expires g e - - - o o o
From g e m m m m m m
Hide R n h o o o o o o
Location R e - - - - - o
Location 3xx e - - o o o o
Location 2xx e - - o o o -
Max-Forwards R n e o o o o o o
Organization R c e - - - o o o
Proxy-Authenticate 407 n h o o o o o o
Proxy-Authorization R n h o o o o o o
Proxy-Require R n h o o o o o o
Priority R c e - - - o - -
Require R n e o o o o o o
Retry-After R c e - - - - - o
Retry-After 600,603 c e - - - o - -
Response-Key R c e - o o o o o
Record-Route R h o o o o o o
Record-Route 2xx h o o o o o o
Route R h - o o o o o
Server r c e o o o o o o
Subject R c e - - - o - -
Timestamp g e o o o o o o
To g n e m m m m m m
Unsupported 420 e o o o o o o
User-Agent R c e o o o o o o
Via g n e m m m m m m
Warning r e o o o o o o
WWW-Authenticate 401 c e o o o o o o
Table 3: Summary of header fields
6.2 Entity Header Fields
Entity-header fields define meta-information about the message-body
or, if no body is present, about the resource identified by the
request. The term "entity header" is an HTTP 1.1 term where the
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the same terminology for header fields but usually refer to the
"message body" rather then the entity as the two are the same in SIP.
6.3 Request Header Fields
The request-header fields allow the client to pass additional
information about the request, and about the client itself, to the
server. These fields act as request modifiers, with semantics
equivalent to the parameters on a programming language method
invocation.
Request-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of request-
header fields if all parties in the communication recognize them to
be request-header fields. Unrecognized header fields are treated as
entity-header fields.
6.4 Response Header Fields
The response-header fields allow the server to pass additional
information about the response which cannot be placed in the Status-
Line. These header fields give information about the server and about
further access to the resource identified by the Request-URI.
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
6.5 End-to-end and Hop-by-hop Headers
End-to-end headers must be transmitted unmodified across all proxies,
while hop-by-hop headers may be modified or added by proxies.
6.6 Header Field Format
Header fields ( general-header, request-header, response-header, and
entity-header) follow the same generic header format as that given in
Section 3.1 of RFC 822 [27,29].
Each header field consists of a name followed by a colon (":") and
the field value. Field names are case-insensitive. The field value
may be preceded by any amount of leading white space (LWS), though a
single space (SP) is preferred. Header fields can be extended over
multiple lines by preceding each extra line with at least one SP or
horizontal tab (HT). Applications SHOULD follow HTTP "common form"
when generating these constructs, since there might exist some
implementations that fail to accept anything beyond the common forms.
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message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = < the OCTETs making up the field-value
and consisting of either *TEXT or combinations
of token, tspecials, and quoted-string>
The order in which header fields are received is not significant if
the header fields have different field names. 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. The
order in which header fields with the same field-name are received is
therefore significant to the interpretation of the combined field
value, and thus a proxy MUST NOT change the order of these field
values when a message is forwarded.
Field names are not case-sensitive, although their values may be.
6.7 Accept
See [H14.1] for syntax. This request-header field is used only with
the OPTIONS and INVITE request methods to indicate what media types
are acceptable in the response.
Example:
Accept: application/sdp;level=1, application/x-private, text/html
6.8 Accept-Encoding
The Accept-Encoding request-header field is similar to Accept, but
restricts the content-codings [H3.4.1] that are acceptable in the
response. See [H14.3].
6.9 Accept-Language
See [H14.4] for syntax. The Accept-Language request header can be
used to allow the client to indicate to the server in which language
it would prefer to receive reason phrases, session descriptions or
status responses carried as message bodies. This may also be used as
a hint by the proxy to which destination to connect the call to
(e.g., for selecting a human operator).
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Accept-Language: da, en-gb;q=0.8, en;q=0.7
6.10 Allow
See [H14.7]. The Allow entity-header field lists the set of methods
supported by the resource identified by the Request-URI. The purpose
of this field is strictly to inform the recipient of valid methods
associated with the resource. An Allow header field MUST be present
in a 405 (Method Not Allowed) response.
6.11 Authorization
See [H14.8] and [30]. A user agent that wishes to authenticate itself
with a server -- usually, but not necessarily, after receiving a 401
response -- MAY do so by including an Authorization request-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.
6.12 Call-ID
The Call-ID general header uniquely identifies a particular
invitation. Note that a single multimedia conference may 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.
For an INVITE request, a callee client application alerts the user
only if the user has not responded previously to the Call-ID in the
INVITE request. If the user is already a member of the conference and
the conference parameters contained in the session description have
not changed, a callee client application MAY silently accept the
call, regardless of the Call-ID. An invitation for an existing
Call-ID or session may change the parameters of the conference. A
client application MAY decide to simply indicate to the user that the
conference parameters have been changed and accept the invitation
automatically or it MAY require user confirmation.
A user may be invited to the same conference or call using several
different Call-IDs. If desired, the client 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.
The Call-ID may be any string consisting of the unreserved URI
characters that can be guaranteed to be globally unique for the
duration of the request. Call-IDs are case-sensitive and are not
URL-encoded.
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reason to deal with the complexity of URL-encoding and
case-ignoring string comparison.
The form UUID@host is recommended, where host is either the fully
qualified domain name or a globally routable IP address. The UUID is
a version-4 (random) UUID [31].
Using cryptographically random identifiers provides some
protection against session hijacking.
Call-ID = ( "Call-ID" | "i" ) ":" UUID "@" host
Example:
Call-ID: f81d4fae-7dec-11d0-a765-00a0c91e6bf6@foo.bar.com
6.13 Content-Encoding
The Content-Encoding entity-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 document to be compressed without losing
the identity of its underlying media type. See [H14.11].
6.14 Content-Length
The Content-Length entity-header field indicates the size of the
message-body, in decimal number of octets, sent to the recipient.
Content-Length = "Content-Length" ":" 1*DIGIT
An example is
Content-Length: 3495
Applications MUST use this field to indicate the size of the
message-body to be transferred, regardless of the media type of the
entity. 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
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how to determine the length of the message body.
6.15 Content-Type
The Content-Type entity-header field indicates the media type of the
message-body sent to the recipient.
Content-Type = "Content-Type" ":" media-type
Example of this header field are
Content-Type: application/sdp
Content-Type: text/html; charset=ISO-8859-4
6.16 CSeq
Clients MUST add the CSeq (command sequence) general-header field to
every request. A CSeq request header field contains a single decimal
sequence number chosen by the requesting client, unique within a
single value of Call-ID or, for requests without Call-ID, within the
request type. The sequence number MUST be expressible as a 32-bit
unsigned integer. The initial value of the sequence number is
arbitrary, but a value of zero is RECOMMENDED. Consecutive requests
that differ in request method, headers or body, but have the same
Call-ID MUST contain strictly monotonically increasing and contiguous
sequence numbers; sequence numbers do not wrap around.
Retransmissions of the same request carry the same sequence number,
but an INVITE with a different message body or different header
fields (a "re-invitation") acquires a new, higher sequence number. A
server responding to a request containing a CSeq header MUST echo
the value in the response. If the Method value is missing, the
server fills it it appropriately.
The ACK and CANCEL requests MUST contain the same CSeq value as
the INVITE request that it refers to, while a BYE request
cancelling an invitation MUST have a higher sequence number.
A user agent server MUST remember the highest sequence number for any
INVITE request with the same Call-ID value. The server MUST respond
to, but ignore any INVITE request with a lower sequence number.
All requests spawned in a parallel search have the same CSeq value
as the request triggering the parallel search.
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Strictly speaking, CSeq header fields are needed for any
SIP request that can be cancelled by a BYE or CANCEL
request or where a client can issue several requests for
the same Call-ID in close succession. Without a sequence
number, the response to an INVITE could be mistaken for
the response to the cancellation ( BYE or CANCEL). Also,
if the network duplicates packets or if an ACK is delayed
until the server has sent an additional response, the
client could interpret an old response as the response to a
re-invitation issued shortly thereafter. Using CSeq also
makes it easy for the server to distinguish different
versions of an invitation, without comparing the message
body.
The Method value allows the client to distinguish the response to an
INVITE request from that of a CANCEL response. CANCEL requests can
be generated by proxies; if they were to increase the sequence
number, it might conflict with a later request issued by the user
agent for the same call.
With a length of 32 bits, a server could generate, within a single
call, one request a second for about 136 years before needing to wrap
around.
Forked requests must have the same CSeq as there would be ambiguity
otherwise between these forked requests and later BYE issued by the
client user agent.
Example:
CSeq: 4711 INVITE
6.17 Date
General header field. See [H14.19].
The Date header field is useful for simple devices without
their own clock.
6.18 Encryption
The Encryption general-header field specifies that the content has
been encrypted. Section 12 describes the overall SIP security
architecture and algorithms. It is intended for end-to-end encryption
of requests and responses. Requests are encrypted with a public key
belonging to the entity named in the To header field. Responses are
encrypted with the public key conveyed in the Response-Key header
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SIP chose not to adopt HTTP's Content-Transfer-Encoding
header because the encrypted body may contain additional
SIP header fields as well as the body of the message.
For any encrypted message, at least the message body and possibly
other message header fields are encrypted. An application receiving a
request or response containing an Encryption header field decrypts
the body and then concatenates the plaintext to the request line and
headers of the original message. Message headers in the decrypted
part completely replace those with the same field name in the
plaintext part. (Note: If only the body of the message is to be
encrypted, the body has to be prefixed with CRLF to allow proper
concatenation.) Note that the request method and Request-URI cannot
be encrypted.
Encryption only provides privacy; the recipient has no
guarantee that the request or response came from the party
listed in the From message header, only that the sender
used the recipients public key. However, proxies will not
be able to modify the request or response.
Encryption = "Encryption" ":" encryption-scheme 1*SP
#encryption-params
encryption-scheme = token
encryption-params = token "=" ( token | quoted-string )
The token indicates the form of encryption used; it is
described in section 12.
The following example for a message encrypted with ASCII-armored PGP
was generated by applying "pgp -ea" to the payload to be encrypted.
INVITE sip:watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From: <sip:a.g.bell@bell-telephone.com>
To: T. A. Watson <sip:watson@bell-telephone.com>
Call-ID: 187602141351@worcester.bell-telephone.com
Content-Length: 885
Encryption: PGP,version=2.6.2,encoding=ascii
hQEMAxkp5GPd+j5xAQf/ZDIfGD/PDOM1wayvwdQAKgGgjmZWe+MTy9NEX8O25Red
h0/pyrd/+DV5C2BYs7yzSOSXaj1C/tTK/4do6rtjhP8QA3vbDdVdaFciwEVAcuXs
ODxlNAVqyDi1RqFC28BJIvQ5KfEkPuACKTK7WlRSBc7vNPEA3nyqZGBTwhxRSbIR
RuFEsHSVojdCam4htcqxGnFwD9sksqs6LIyCFaiTAhWtwcCaN437G7mUYzy2KLcA
zPVGq1VQg83b99zPzIxRdlZ+K7+bAnu8Rtu+ohOCMLV3TPXbyp+err1YiThCZHIu
X9dOVj3CMjCP66RSHa/ea0wYTRRNYA/G+kdP8DSUcqYAAAE/hZPX6nFIqk7AVnf6
IpWHUPTelNUJpzUp5Ou+q/5P7ZAsn+cSAuF2YWtVjCf+SQmBR13p2EYYWHoxlA2/
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wbWvia6cAIfsvlZ9JGocmQYF7pcuz5pnczqP+/yvRqFJtDGD/v3s++G2R+ViVYJO
z/lxGUZaM4IWBCf+4DUjNanZM0oxAE28NjaIZ0rrldDQmO8V9FtPKdHxkqA5iJP+
6vGOFti1Ak4kmEz0vM/Nsv7kkubTFhRl05OiJIGr9S1UhenlZv9l6RuXsOY/EwH2
z8X9N4MhMyXEVuC9rt8/AUhmVQ==
=bOW+
Since proxies may base their forwarding decision on any combination
of SIP header fields, there is no guarantee that an encrypted request
"hiding" header fields will reach the same destination as an
otherwise identical un-encrypted request.
6.19 Expires
The Expires entity-header field gives the date and time after which
the message content expires.
This header field is currently defined only for the REGISTER and
INVITE methods. For REGISTER, it is a request and response-header
field and allows the client to indicate how long the registration is
to be valid; the server uses it to indicate when the client has to
re-register. The server's choice overrides that of the client. The
server MAY choose a shorter time interval than that requested by the
client, but SHOULD not choose a longer one.
For INVITE, it is a request and response-header field. In a request,
the callee can limit the validity of an invitation. (For example, if
a client wants to limit how long a search should take at most or when
a conference invitation is time-limited. A user interface may take
this is as a hint to leave the invitation window on the screen even
if the user is not currently at the workstation.) This also limits
the duration of a search. If the request expires before the search
completes, the proxy returns a 408 (Request Timeout) status. In a 302
(Moved Temporarily) response, a server can advise the client of the
maximal duration of the redirection.
The value of this field can be either an HTTP-date 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.
Expires = "Expires" ":" ( HTTP-date | delta-seconds )
Two example of its use are
Expires: Thu, 01 Dec 1994 16:00:00 GMT
Expires: 5
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6.20 From
Requests and responses MUST contain a From general-header field,
indicating the initiator of the request. The server copies the To and
From header fields from the request to the response. The optional
display-name is meant to be rendered by a human user interface.
From = ( "From" | "f" ) ":" ( name-addr | addr-spec )
name-addr = [ display-name ] "<" addr-spec ">"
addr-spec = SIP-URL | URI
display-name = *token | quoted-string
Examples:
From: A. G. Bell <sip:agb@bell-telephone.com>
From: sip:+12125551212@server.phone2net.com
From: Anonymous <sip:c8oqz84zk7z@privacy.org>
Call-ID, To and From are needed to identify a call leg
matters in calls with third-party control. The format is
similar to the equivalent RFC 822 header, but with a URI
instead of just an email address.
6.21 Hide
The Hide request header field indicates that the path comprised of
the Via header fields (Section 6.40) should be hidden from
subsequent proxies and user agents. It can take two forms: Hide:
route and Hide:hop. Hide header fields are typically added by the
client user agent, but MAY be added by any proxy along the path.
If a request contains the " Hide: route" header field, all following
proxies SHOULD hide their previous hop. If a request contains the "
Hide: hop" header field, only the next proxy SHOULD hide the previous
hop and then remove the Hide option unless it also wants to remain
anonymous.
A server hides the previous hop by encrypting the host and port
parts of the top-most Via header with an algorithm of its choice.
Servers SHOULD add additional "salt" to the host and port
information prior to encryption to prevent malicious downstream
proxies from guessing earlier parts of the path based on seeing
identical encrypted Via headers. Hidden Via fields are marked with
the hidden Via option, as described in Section 6.40.
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decrypt all Via headers marked as "hidden" to perform loop
detection. Servers that are not capable of hiding can ignore hidden
Via fields in their loop detection algorithm.
If hidden headers were not marked, a proxy would have to
decrypt all headers to detect loops, just in case one was
encrypted, as the Hide: Hop option may have been removed
along the way.
A host MUST NOT add such a " Hide:hop" header field unless it can
guarantee it will only send a request for this destination to the
same next hop. The reason for this is that it is possible that the
request will loop back through this same hop from a downstream proxy.
The loop will be detected by the next hop if the choice of next hop
is fixed, but could loop an arbitrary number of times otherwise.
A client requesting " Hide: route" can only rely on keeping the
request path private if it sends the request to a trusted proxy.
Hiding the route of a SIP request may be of limited value if the
request results in data packets being exchanged directly between the
calling and called user agent.
The use of Hide header fields is discouraged unless path privacy is
truly needed; Hide fields impose extra processing costs and
restrictions for proxies and can cause requests to generate 482 (Loop
Detected) responses that could otherwise be avoided.
The encryption of Via header fields is described in more detail in
Section 12.
The Hide header field has the following syntax:
Hide = "Hide" ":" ( "route" | "hop" )
6.22 Location
The Location general-header field can appear in requests, 2xx
responses and 3xx responses.
REGISTER requests: REGISTER requests MAY contain Location header
fields. They indicate under which locations the user may be
reachable. The REGISTER request defines a wildcard Location
field, "*". that is only used with Expires: 0 to remove all
registrations for a particular user.
INVITE and ACK requests: INVITE and ACK requests MAY contain
Location headers indicating the location the request is
originating from. If the SIP address does not refer to the user
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several locations within the same domain served by the proxy.)
This allows the callee to send a BYE directly to the
caller instead of through a series of proxies. The Via
header is not sufficient since the desired address may be
that of a proxy.
INVITE 2xx responses: A user agent server sending a definitive,
positive response (2xx), MAY insert a Location response header
indicating the SIP address under which it is reachable most
directly for future SIP requests, such as ACK. This may be the
address of the server itself or that of a proxy, e.g., if the
host is behind a firewall. If the SIP address does not refer to
the user agent server, the SIP URL MUST contain a tag parameter
uniquely identifying the user agent. (The same person may be
logged on at several locations within the same domain served by
the proxy.) The value of this Location header is copied into
the Request-URI of subsequent ACK and BYE requests for this
call.
REGISTER 2xx responses: Similarly, a REGISTER response SHOULD return
all locations that a user is currently reachable under.
3xx responses: The Location response-header field can be used with a
3xx response codes to indicate one or more addresses to try. It
can appear in responses to INVITE and OPTIONS methods. The
Location header field contains URIs giving the new locations or
user names to try, or may simply specify additional transport
parameters. A 301 (Moved Permanently) or 302 (Moved Temporarily)
response SHOULD contain a Location field containing URIs of new
addressed to be tried. A 301 or 302 response may also give the
same location and username that was being tried but specify
additional transport parameters such as a multicast address to
try or a change of SIP transport from UDP to TCP or vice versa.
Note that the Location header 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 information announcement
such as "The number you have dialed has been changed."
A Location response header may contain any suitable URI indicating
where the called party may be reached, not limited to SIP URLs. For
example, it may contain a phone or fax URL [22], a mailto: URL [19]
or irc: URL.
The following parameters are defined. Additional parameters may be
defined in other specifications.
q: The qvalue indicates the relative preference among the locations
given. qvalue values are decimal numbers from 0.0 to 1.0, with
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action: The action is only used when registering with the REGISTER
request. It indicates how the client wishes forwarding to occur,
by proxying or by redirection. The action taken if this
parameter is not specified depends on server configuration. In
its response, the registrar SHOULD indicate the mode used. This
parameter is ignored for other requests.
Location = ( "Location" | "m" ) ":" ("*" | (1# (( SIP-URL | URI )
*( ";" location-params )))
location-params = "q" "=" qvalue
| "action" "=" "proxy" | "redirect"
| extension-attribute
extension-attribute = extension-name [ "=" extension-value ]
Example:
Location: sip:watson@worcester.bell-telephone.com;tag=123
;q=0.7,
mailto:watson@bell-telephone.com ;q=0.1
6.23 Max-Forwards
The Max-Forwards request-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 inbound 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. [H14.31]
Max-Forwards = "Max-Forwards" ":" 1*DIGIT
The Max-Forwards value is a decimal integer indicating the remaining
number of times this request message may be forwarded.
Each proxy or gateway recipient of a request containing a Max-
Forwards header field MUST check and update its value prior to
forwarding the request. If the received value is zero (0), the
recipient MUST NOT forward the request. Instead, for the OPTIONS and
REGISTER methods, it MUST respond as the final recipient. For all
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If the received Max-Forwards value is greater than zero, then the
forwarded message MUST contain an updated Max-Forwards field with a
value decremented by one (1).
Example:
Max-Forwards: 6
6.24 Organization
The Organization request-header field conveys the name of the
organization to which the callee belongs. It may also be inserted by
proxies at the boundary of an organization and may be used by client
software to filter calls.
Organization = "Organization" ":" *text
6.25 Priority
The Priority request header signals the urgency of the call to the
callee.
Priority = "Priority" ":" priority-value
priority-value = "emergency" | "urgent" | "normal"
| "non-urgent"
The value of "emergency" should only be used when life, limb or
property are in imminent danger.
Examples:
Subject: A tornado is heading our way!
Priority: emergency
Subject: Weekend plans
Priority: non-urgent
These are the values of RFC 2076, with the addition of
"emergency".
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The Proxy-Authenticate response-header field MUST be included as
part of a 407 (Proxy Authentication Required) response. The field
value consists of a challenge that indicates the authentication
scheme and parameters applicable to the proxy for this Request-URI.
See [H14.33] for further details.
A client SHOULD cache the credentials used for a particular proxy
server and realm for the next request to that server. Credentials
are, in general, valid for a specific value of the Request-URI at a
particular proxy server. If a client contacts a proxy server that has
required authentication in the past, but the client does not have
credentials for the particular Request-URI, it MAY attempt to use
the most-recently used credential. The server responds with 401
(Unauthorized) if the client guessed wrong.
This suggested caching behavior is motivated by proxies
restricting phone calls to authenticated users. It seems
likely that in most cases, all destinations require the
same password. Note that end-to-end authentication is
likely to be destination-specific.
6.27 Proxy-Authorization
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. See
[H14.34] for further details.
6.28 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive
features that MUST be supported by the proxy. Any Proxy-Require
header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client if not supported.
Servers treat this field identically to the Require field.
See Section 6.29 for more details on the mechanics of this message
and a usage example.
6.29 Require
The Require request header 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. If a server does not
understand the option, it MUST respond by returning status code 420
(Bad Extension) and list those options it does not understand in the
Unsupported header.
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Require = "Require" ":" 1#option-tag
Example:
C->S: INVITE sip:watson@bell-telephone.com SIP/2.0
Require: com.example.billing
Payment: sheep_skins, conch_shells
S->C: 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.
Proxy and redirect servers MUST ignore features that are not
understood. If a particular extension requires that intermediate
devices support it, the extension should be tagged in the Proxy-
Require field instead (see Section 6.28).
6.30 Record-Route
The Record-Route request and response header field is added to an
INVITE request by any proxy that insists on being in the path of
subsequent ACK and BYE requests for the same call. It contains a
globally reachable Request-URI that identifies the proxy server.
Each proxy server adds its Request-URI to the beginning of the list.
The server copies the Record-Route header unchanged into the
response. ( Record-Route is only relevant for 2xx responses.)
The calling user agent client copies the Record-Route header into a
Route header of subsequent requests, reversing the order of requests,
so that the first entry is closest to the caller. If the response
contained a Location header field, the calling user agent adds its
content as the last Route header. Unless this would cause a loop,
any clientMUST send any subsequent requests for this Call-ID to the
first Request-URI in the Route request header and remove that entry.
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call state and thus need to receive any BYE and ACK
packets for the call.
The Record-Route header field has the following syntax:
Record-Route = "Record-Route" ":" 1# request-uri
Example for a request that has traversed the hosts ieee.org and
bell-telephone.com , in that order:
Record-Route: sip:a.g.bell@bell-telephone.com, sip:a.bell@ieee.org
6.31 Response-Key
The Response-Key request header field can be used by a client to
request the key that the called user agent SHOULD use to encrypt the
response with. The syntax is:
Response-Key = "Response-Key" ":" key-scheme 1*SP #key-param
key-scheme = token
key-param = token "=" ( token | quoted-string )
The key-scheme gives the type of encryption to be used for response.
Section 12 describes security schemes.
If the client insists that the server return an encrypted response,
it includes a
Require: org.ietf.sip.encrypt-response
header field in its request. If the client cannot encrypt for
whatever reason, it MUST follow normal Require header field
procedures and return an 420 (Bad Extension) response. If this
Require header is not present, a client SHOULD still encrypt, but MAY
return an unencrypted response if unable to.
6.32 Route
The Route request header determines the route taken by a request.
Each host removes the first entry and then proxies the request to the
host listed in that entry, also using it as the Request-URI. The
operation is further described in Section 6.30.
The Route header field has the following syntax:
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6.33 Retry-After
The Retry-After response 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 may be available again. The value of this field can
be either an HTTP-date or an integer number of seconds (in decimal)
after the time of the response.
A REGISTER request may include this header field when deleting
registrations with Location: *; Expires: 0. The Retry-After value
then indicates when the user might again be reachable. The registrar
MAY then include this information in responses to future calls.
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.
Retry-After = "Retry-After" ":" ( HTTP-date | delta-seconds )
[ comment ] [ ";duration" "=" delta-seconds
Examples of its use are
Retry-After: Mon, 21 Jul 1997 18:48:34 GMT (I'm in a meeting)
Retry-After: Mon, 1 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.
6.34 Server
The Server response-header field contains information about the
software used by the user agent server to handle the request. See
[H14.39].
6.35 Subject
This is intended to provide a summary, or indicate the nature, of the
call, allowing call filtering without having to parse the session
description. (Also, the session description may not necessarily use
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Subject = ( "Subject" | "s" ) ":" *text
Example:
Subject: Tune in - they are talking about your work!
6.36 Timestamp
The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and may use any timescale. The server MUST echo
the exact same value and MAY, if it has accurate information about
this, add a floating point number indicating the number of seconds
that have elapsed since it has received the request. The timestamp is
used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
6.37 To
The To general-header field specifies the invited user, with the
same SIP URL syntax as the From field.
To = ( "To" | "t" ) ":" ( name-addr | addr-spec )
A SIP server returns a 400 (Bad Request) response if it receives a
request with a To header field containing a URI with a scheme it
does not recognize.
Example:
To: The Operator <sip:operator@cs.columbia.edu>
To: sip:+12125551212@server.phone2net.com
Call-ID, To and From are needed to identify a call leg
matters in calls with third-party control.
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The Unsupported response header lists the features not supported by
the server. See Section 6.29 for a usage example and motivation.
6.39 User-Agent
The User-Agent request-header field contains information about the
client user agent originating the request. See [H14.42].
6.40 Via
The Via field indicates the path taken by the request so far. This
prevents request looping and ensures replies take the same path as
the requests, which assists in firewall traversal and other unusual
routing situations.
6.40.1 Requests
The client originating the request MUST insert into the request a Via
field containing its host name or network address and, if not the
default port number, the port number it wishes to receive responses
at. (Note that this port number may differ from the UDP source port
number of the request.) A fully-qualified domain name is RECOMMENDED.
Each subsequent proxy server that sends the request onwards MUST add
its own additional Via field before any existing Via fields.
A proxy that receives a redirection (3xx) response and then searches
recursively, MUST use the same Via headers as on the original
request.
A proxy SHOULD check the top-most Via header to ensure that it
contains the sender's correct network address, as seen from that
proxy. If the sender's address is incorrect, the proxy should add an
additional received attribute, as described below.
A host behind a network address translator (NAT) or
firewall may not be able to insert a network address into
the Via header that can be reached by the next hop beyond
the NAT. Hosts behind NATs or NAPTs should insert the local
port number of the outgoing socket, rather than the port
number for incoming requests, as NAPTs assume that
responses return with reversed source and destination
ports.
Additionally, if the message goes to a multicast address, an extra
Via field is added by the sender before all the other Via fields
giving the multicast address and TTL.
If a proxy server receives a request which contains its own address,
it MUST respond with a 482 (Loop Detected) status code.
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loops. Also, it cannot be guaranteed that a proxy server
can always detect that the address returned by a location
service refers to a host listed in the Via list, as a
single host may have aliases or several network interfaces.
6.40.2 Receiver-tagged Via Fields
Normally every host that sends or forwards a SIP message adds a Via
field indicating the path traversed. However, it is possible that
Network Address Translators (NAT) may change the source address of
the request, in which case the Via field cannot be relied on to
route replies. To prevent this, a proxy SHOULD check the top-most
Via header to ensure that it contains the sender's correct network
address, as seen from that proxy. If the sender's address is
incorrect, the proxy should add a received tag to the Via field
inserted by the previous hop. Such a modified Via field is known as a
receiver-tagged Via field. An example is:
Via: SIP/2.0/UDP erlang.bell-telephone.com:5060
Via: SIP/2.0/UDP 10.0.0.1:5060 ;received=199.172.136.3
In this example, the message went from 10.0.0.1 and through a NAT
using external address border.ieee.org (199.172.136.3) to
erlang.bell-telephone.com tagged the previous hop's Via field with
the address that it actually came from.
6.40.3 Responses
In the return path, Via fields are processed by a proxy or client
according to the following rules:
1. The first Via field should indicate the proxy or client
processing this message. If it does not, discard the
message. Otherwise, remove this Via field.
2. If the second Via field in a response is a multicast
address, remove that Via field, and send the message to
the multicast address indicated.
3. If the second Via field is a receiver-tagged field
(Section 6.40.2), send the message to the address in the
received tag. Otherwise, send send the message to the
address indicated in the sent-by parameter.
4. If there is no second Via field, this response is destined
for this client.
These rules ensure that a client only has to check the first Via
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A user agent server or redirect server returns the response to the
network address where the request came from. (Since these servers do
not forward the request, they do not add a received tag.)
6.40.4 Syntax
The format for a Via header is:
Via = ( "Via" $|$ "v") ":" 1#( sent-protocol sent-by
*( ";" via-params ) [ comment ] )
via-params = via-hidden | via-ttl | via-received
| via-branch
via-hidden = "hidden"
via-ttl = "ttl" "=" ttl
via-received = "received" "=" host
via-branch = "branch" "=" token
sent-protocol = [ protocol-name "/" ] protocol-version
[ "/" transport ]
protocol-name = "SIP" $|$ token
protocol-version = token
transport = "UDP" $|$ "TCP" $|$ token
sent-by = ( host [ ":" port ] ) $|$ ( concealed-host )
concealed-host = token
ttl = 1*3DIGIT ; 0 to 255
The " ttl" parameter is included only if the address is a multicast
address. The " received" parameter is added only for receiver-added
Via fields (Section 6.40.2). For reasons of privacy, a client or
proxy may wish to hide its Via information by encrypting it (see
Section 6.21). The " hidden" parameter is included if this header
was hidden by the upstream proxy (see 6.21).
The " branch" parameter is included by every forking proxy. The
token uniquely identifies a branch of a particular search. The
identifier has to be unique only within a set of isomorphic requests.
Note that privacy of the proxy relies on the cooperation of the next
hop, as the next-hop proxy will, by necessity, know the IP address
and port number of the source host.
Via: SIP/2.0/UDP first.example.com:4000
Via: SIP/2.0/UDP adk8
6.41 Warning
The Warning response-header field is used to carry additional
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Warning = "Warning" ":" 1#warning-value
warning-value = warn-code SP warn-agent SP warn-text
warn-code = 3DIGIT "." 2DIGIT
warn-agent = ( host [ ":" port ] ) | pseudonym
; the name or pseudonym of the server adding
; the Warning header, for use in debugging
warn-text = quoted-string
A response may carry more than one Warning header.
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 may be based on any available knowledge, such as the
location of the cache or user, the Accept-Language field in a
request, the Content-Language field in a response, etc. The default
language is English.
Any server may add Warning headers to a response. New Warning
headers MUST be added after any existing Warning headers. A proxy
server MUST NOT delete any Warning header that it received with a
response.
When multiple Warning headers are attached to a response, the user
agent SHOULD display as many of them as possible, in the order that
they appear in the response. If it is not possible to display all of
the warnings, the user agent first displays warnings that appear
early in the response.
Systems that generate multiple Warning headers should order them
with this user agent behavior in mind.
Example:
Warning: 606.4 isi.edu Multicast not available
Warning: 606.2 isi.edu Incompatible protocol (RTP/XXP)
6.42 WWW-Authenticate
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 [H14.46] and [30].
The WWW-Authenticate response-header field MUST be included in 401
(Unauthorized) response messages.
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A user agent SHOULD cache the authorization credentials for a given
value of the destination ( To header) and realm and attempt to re-use
these values on the next request for that destination.
In addition to the "basic" and "digest" authentication schemes
defined in the specifications cited above, SIP defines a new scheme,
PGP (RFC 2015, [32]), Section 13. Other schemes, such as S-MIME, are
for further study.
7 Status Code Definitions
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.
7.1 Informational 1xx
Informational responses indicate that the server or proxy contacted
is performing some further action and does not yet have a definitive
response. The client SHOULD wait for a further response from the
server, and the server SHOULD send such a response without further
prompting. Typically a server should send a 1xx response if it
expects to take more than 200 ms to obtain a final response. A server
can issue zero or more 1xx responses, with no restriction on their
ordering or uniqueness. Note that 1xx responses are not transmitted
reliably, that is, they do not cause the client to send an ACK.
Servers are free to retransmit informational responses and clients
can inquire about the current state of call processing by re-sending
the request.
7.1.1 100 Trying
Some unspecified action is being taken on behalf of this call (e.g.,
a database is being consulted), but the user has not yet been
located.
7.1.2 180 Ringing
The called user agent has located a possible location where the user
has been recently and is trying to alert them.
7.1.3 181 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 new
destinations are listed in Location headers. Proxies SHOULD be
configurable not to reveal this information.
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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.
7.2 Successful 2xx
The request was successful and MUST terminate a search.
7.2.1 200 OK
The request has succeeded. The information returned with the response
depends on the method used in the request, for example:
BYE: The call has been terminated. The message body is empty.
CANCEL: The search has been cancelled. The message body is empty.
INVITE: The callee has agreed to participate; the message body
indicates the callee's capabilities.
OPTIONS: The callee has agreed to share its capabilities, included in
the message body.
REGISTER: The registration has succeeded. The client treats the
message body according to its Content-Type.
7.3 Redirection 3xx
3xx responses give information about the user's new location, or
about alternative services that may be able to satisfy the call. They
SHOULD terminate an existing search, and MAY cause the initiator to
begin a new search if appropriate.
Any redirection (3xx) response MUST NOT suggest any of the addresses
in the Via (Section 6.40) path of the request in the Location header
field. (Addresses match if their host and port number match.)
To avoid forwarding loops, a user agent client or proxy MUST check
whether the address returned by a redirect server equals an address
tried earlier.
7.3.1 300 Multiple Choices
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.
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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 entity format is specified by the media type given in the
Content-Type header field. The choices SHOULD also be listed as
Location fields (Section 6.22). Unlike HTTP, the SIP response may
contain several Location fields or a list of addresses in a
Location field. User agents MAY use the Location 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 header is appropriate if the callee can be reached at
several different locations and the server cannot or
prefers not to proxy the request.
7.3.2 301 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
Location header field (Section 6.22). 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.
7.3.3 302 Moved Temporarily
The requesting client should retry the request at the new address(es)
given by the Location header field (Section 6.22). The duration of
the redirection can be indicated through an Expires (Section 6.19)
header.
7.3.4 380 Alternative Service
The call was not successful, but alternative services are possible.
The alternative services are described in the message body of the
response.
7.3.5 381 Ambiguous
The callee address provided in the request was ambiguous. The
response MAY contain a listing of possible unambiguous addresses in
Location headers.
Revealing alternatives may 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 :
381 Ambiguous SIP/2.0
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Location: lee.foote@example.com (Lee M. Foote)
Some email and voice mail systems provide this
functionality. A status code separate from 300 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 300 response, user intervention is
required for a 381 response.
7.4 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 may be successful.
7.4.1 400 Bad Request
The request could not be understood due to malformed syntax.
7.4.2 401 Unauthorized
The request requires user authentication.
7.4.3 402 Payment Required
Reserved for future use.
7.4.4 403 Forbidden
The server understood the request, but is refusing to fulfill it.
Authorization will not help, and the request should not be repeated.
7.4.5 404 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.
7.4.6 405 Method Not Allowed
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 containing a list of valid methods for the indicated
address.
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This code is similar to 401 (Unauthorized), but indicates that the
client MUST first authenticate itself with the proxy. The proxy MUST
return a Proxy-Authenticate header field (section 6.26) containing a
challenge applicable to the proxy for the requested resource. The
client MAY repeat the request with a suitable Proxy-Authorization
header field (section 6.27). SIP access authentication is explained
in section 12.3 and [H11].
This status code should be used for applications where access to the
communication channel (e.g., a telephony gateway) rather than the
callee herself requires authentication.
7.4.8 408 Request Timeout
The server could not produce a response, e.g., a user location,
within the time indicated in the request via the Expires header. The
client MAY repeat the request without modifications at any later
time.
7.4.9 420 Bad Extension
The server did not understand the protocol extension specified in a
Require (Section 6.29) header field.
7.4.10 480 Temporarily Unavailable
The callee's end system was contacted successfully but the callee is
currently unavailable (e.g., not logged in or logged in in such a
manner as to preclude communication with the callee). The response
may indicate a better time to call in the Retry-After header. The
user may also be available elsewhere (unbeknownst to this host),
thus, this response does not terminate any searches. The reason
phrase SHOULD indicate the more precise cause as to why the callee is
unavailable. This value SHOULD be setable by the user agent.
7.4.11 481 Invalid Call-ID
The server received a BYE or CANCEL request with a Call-ID (Section
6.12) value it does not recognize. (A server simply discards an ACK
with an invalid Call-ID.)
7.4.12 482 Loop Detected
The server received a request with a Via (Section 6.40) path
containing itself.
7.4.13 483 Too Many Hops
The server received a request that contains more Via entries (hops)
(Section 6.40) than allowed by the Max-Forwards (Section 6.23)
header field.
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The server received a request with a To (Section 6.37) address or
Request-URI that was incomplete. Additional information should be
provided.
This status code allows overlapped dialing.
7.5 Server Failure 5xx
5xx responses are failure responses given when a server itself has
erred. They are not definitive failures, and MUST NOT terminate a
search if other possible locations remain untried.
7.5.1 500 Server Internal Error
The server encountered an unexpected condition that prevented it from
fulfilling the request.
7.5.2 501 Not Implemented
The server does not support the functionality required to fulfill the
request. This is the appropriate response when the server does not
recognize the request method and is not capable of supporting it for
any user.
7.5.3 502 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.
7.5.4 503 Service Unavailable
The server is currently unable to handle the request due to a
temporary overloading 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.
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.
7.5.5 504 Gateway Timeout
The server, while acting as a gateway, did not receive a timely
response from the server (e.g., a location server) it accessed in
attempting to complete the request.
7.5.6 505 Version Not Supported
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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 SHOULD contain an entity describing why
that version is not supported and what other protocols are supported
by that server.
7.6 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. All further searches for this user are doomed to failure
and pending searches SHOULD be terminated.
7.6.1 600 Busy
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 should use status code 603 (Decline) instead.
7.6.2 603 Decline
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.
7.6.3 604 Does Not Exist Anywhere
The server has authoritative information that the user indicated in
the To request field does not exist anywhere. Searching for the user
elsewhere will not yield any results.
7.6.4 606 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 describing why the session described cannot be
supported. These reasons can be one or more of:
606.1 Insufficient Bandwidth: The bandwidth specified in the session
description or defined by the media exceeds that known to be
available.
606.2 Incompatible Protocol: One or more protocols described in the
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606.3 Incompatible Format: One or more media formats described in the
request is not available.
606.4 Multicast Not Available: The site where the user is located
does not support multicast.
606.5 Unicast Not Available: The site where the user is located does
not support unicast communication (usually due to the presence
of a firewall).
Other reasons are likely to be added later. 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.
8 SIP Message Body
8.1 Body Inclusion
For a request message, the presence of a body is signaled by the
inclusion of a Content-Length header. Only ACK, INVITE, OPTIONS
and REGISTER requests may contain message bodies. For ACK, INVITE
and OPTIONS, the message body is always a session description. The
use of message bodies for REGISTER requests is for further study.
For response messages, whether or not a body is included is dependent
on both the request method and the response message's response code.
All responses MAY include a body, although it may be of zero length.
Message bodies for 1xx responses contain advisory information about
the progress of the request, 2xx responses contain session
descriptions; for responses with status 300 or greater, the session
body MAY contain additional, human-readable information about the
reasons for failure. It is RECOMMENDED that information in 1xx and
300 and greater responses be of type text/plain or text/html
8.2 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.
8.3 Message Body Length
The body length in bytes MUST be given by the Content-Length header
field. If no body is present in a message, then the Content-Length
header MUST set to zero. If a server receives a message without
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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.)
9 Compact Form
When SIP is carried over UDP with authentication and a complex
session description, it may be possible that the size of a request or
response is larger than the MTU. To reduce this problem, a more
compact form of SIP is also defined by using alternative names for
common header fields. These short forms are NOT abbreviations, they
are field names. No other header field abbreviations are allowed.
short field name long field name note
c Content-Type
e Content-Encoding
f From
i Call-ID
l Content-Length
m Location from "moved"
s Subject
t To
v Via
Thus the header in section 14.2 could also be written:
INVITE schooler@vlsi.caltech.edu SIP/2.0
v:SIP/2.0/UDP 239.128.16.254 16
v:SIP/2.0/UDP 131.215.131.131
v:SIP/2.0/UDP 128.16.64.19
f:mjh@isi.edu
t:schooler@cs.caltech.edu
i:62729-27@128.16.64.19
c:application/sdp
l:187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
Handley/Schulzrinne/Schooler [Page 61]
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requests. Proxies MUST NOT translate a request between short and long
forms if authentication fields are present.
10 SIP Transport
10.1 General Remarks
SIP is defined so it can use either UDP (unicast or multicast) or TCP
as a transport protocol; it provides its own reliability mechanism.
10.1.1 Requests
Stateful proxies mark outgoing requests with the branch parameter in
the Via header.
Servers ignore isomorphic requests, but retransmit the appropriate
response. (SIP requests are said to be idempotent , i.e., receiving
more than one copy of a request does not change the server state.)
If a stateful proxy, user agent or redirect server cannot respond to
a request with a final response within 200 ms, it MUST issue a
provisional (1xx) response as soon as possible. Stateless proxies
MUST NOT issue provisional responses on their own.
After receiving a CANCEL request from an upstream client, a stateful
proxy server SHOULD send a CANCEL on all branches where it has not
yet received a final response.
10.1.2 Responses
A server MAY issue one or more provisional responses at any time
before sending a final response.
Responses are mapped to requests by the matching To, From, Call-ID,
CSeq headers and the branch parameter of the first Via header.
Responses terminate request retransmissions even if they have Via
headers that cause them to be delivered to an upstream client.
A stateful proxy may receive a response that it does not have state
for, that is, where it has no a record of an isomorphic request. If
the Via header field indicates that the upstream server used TCP, the
proxy actively opens a TCP connection to that address. Thus, proxies
have to be prepared to receive responses on the incoming side of
passive TCP connections, even though most responses will arrive on
the incoming side of an active connection. (An active connection is a
TCP connection initiated by the proxy, a passive connection is one
accepted by the proxy, but initiated by another entity.)
100 responses are not forwarded, other 1xx responses MAY be
forwarded, possibly after the server eliminates responses with status
codes that had already been sent earlier. 2xx responses are forwarded
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with status 300 and higher are retransmitted by each stateful proxy
until the next upstream proxy sends an ACK (see below for timing
details) or CANCEL.
A stateful proxy can remove state for a call attempt and close any
connections 20 seconds after receiving the first final response.
The 20 second window is given by the maximum retransmission
duration of 200 responses (10 times T4), in case the ACK
is lost somewhere on the way to the called user agent or
the next stateful proxy.
10.2 Unicast UDP
UDP packets MUST have a source address that is valid as a destination
for requests and responses. Responses are returned to the address
listed in the Via header field (Section 6.40), not the source
address of the request.
10.3 Multicast UDP
Requests may be multicast. Multicast requests SHOULD have a time-to-
live value of no greater than one, i.e., be restricted to the local
network.
If the request was received via multicast, the response is also
returned via multicast. The server delays its response by a random
interval between zero and one second. Servers do not return 404 (Not
Found) responses and SHOULD suppress responses if they hear a lower-
numbered or 6xx response from another group member prior to sending.
Servers do not respond to CANCEL requests received via multicast.
10.4 BYE, CANCEL, OPTIONS
A SIP client SHOULD retransmit a BYE, CANCEL, or OPTIONS request
periodically with timer T1 until it receives a response, or until it
has reached a set limit on the number of retransmissions. If the
response is provisional, the client continues to retransmit the
request, albeit less frequently, using timer T2. The default values
of timer T1 and T2 are 1 and 5 seconds, respectively. The default
retransmit limit is 20 times. After the server sends a final
response, it cannot be sure the client has received the response, and
thus SHOULD cache the results for at least 100 seconds to avoid
having to, for example, contact the user or user location server
again upon receiving a retransmission.
Each server in a proxy chain generates its own final response to a
CANCEL request. BYE and OPTIONS final responses are generated by
redirect and user agent servers.
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The value of the initial retransmission timer is smaller
than that that for TCP since it is expected that network
paths suitable for interactive communications have round-
trip times smaller than 1.5 seconds. To avoid flooding the
network with packets every second even if the destination
network is unreachable, the retransmission count has to be
bounded. Given that most transactions should consist of one
request and a few responses, round-trip time estimation
seems less than helpful. If RTT estimation is desired to
more quickly discover a missing final response, each
request retransmission needs to be labeled with its own
Timestamp (Section 6.36), returned in the response. The
server caches the result until it can be sure that the
client will not retransmit the same request again.
10.5 REGISTER
A client MAY repeat its registration attempts at intervals of 2, 4,
8, ..., 512, 512, ... seconds if it receives no response.
Retransmitting REGISTER indefinitely ensures that a user
will eventually be able to register after a registrar
recovers from a crash. The period is chosen so that even on
a large LAN, there will not be more than about one
REGISTER request per second.
10.6 ACK
The ACK request does not generate responses. It is only
retransmitted when a response to an INVITE request arrives.
10.7 INVITE
Special considerations apply for the INVITE method.
1. After receiving an invitation, considerable time may elapse
before the server can determine the outcome. For example,
the called party may be "rung" or extensive searches may be
performed, so delays between the request and a definitive
response can reach several tens of seconds. If either
caller or callee are automated servers not directly
controlled by a human being, a call attempt may be
unbounded in time.
2. If a telephony user interface is modeled or if we need to
interface to the PSTN, the caller's user interface will
provide "ringback", a signal that the callee is being
alerted. (The status response 180 (Ringing) may be used to
initiate ringback.) Once the callee picks up, the caller
needs to know so that it can enable the voice path and stop
Handley/Schulzrinne/Schooler [Page 64]
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caller will continue to hear ringback while the callee
assumes that the call exists.
3. The client has to be able to terminate an on-going request,
e.g., because it is no longer willing to wait for the
connection or search to succeed. The server will have to
wait several round-trip times to interpret the lack of
request retransmissions as the end of a call. If the call
succeeds shortly after the caller has given up, the callee
will "pick up the phone" and not be "connected".
A SIP client SHOULD retransmits a SIP INVITE request periodically
with timer T1 until it receives a response, or until it has reached a
set limit on the number of retransmissions. If the response is
provisional, the client continues to retransmit the request, albeit
less frequently, using timer T3. The default values of timer T1 and
T3 are 1 and 30 seconds, respectively. The default retransmit limit
is 20.
The value of T3 was chosen so that for most normal phone
calls, only one INVITE request will be issued. Typically,
a phone switches to an answering machine or voice mail
after about 20--22 seconds.
Upon receiving a 2xx final response, the client sends an ACK to the
address listed in the Location header field contained in the
response. If the response did not contain a Location header, the
client uses the same To header field as for the INVITE request and
sends the ACK to the same destination as the original INVITE
request.
ACKs for final responses other than 2xx are sent to the source of the
response.
The server retransmits the final response at intervals of T4 (default
value of T4 = 2 seconds) until it receives an ACK request for the
same Call-ID and CSeq from the same From source or until it has
retransmitted the final response 10 times. The ACK request MUSTNOT be
acknowledged to prevent a response- ACK feedback loop.
Fig. 8 and 9 show the client and server state diagram for
invitations.
Using the mechanism in Sec. 10.4 does not work well for the
long delays between INVITE and a final response. If the
200 response gets lost, the callee would believe the call
to exist, but the voice path would be dead since the caller
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+===========+
| Initial |
+===========+
|
|
| -
| ------
| INVITE
+------v v
T1 +-----------+
------ | Calling |--------+
INVITE +-----------+ |
+------| | | |
+----------------+ | |
| | 1xx | >= 200
| | --- | ------
| | - | ACK
| | |
| +------v v v-----| |
| T3 +-----------+ 1xx |
| ------ | Ringing | --- |
| INVITE +-----------+ - |
| +------| | |-----+ |
| | |
| 2xx | |
| --- | 2xx |
| ACK | --- |
| | ACK |
+----------------+ | |
+------v | v |
xxx +-----------+ |
--- | Completed |<-------+
ACK +-----------+
+------|
event
-------
message
Figure 8: State transition diagram of client for INVITE method
order of a second or two to limit the duration of this
state confusion.
Blindly retransmitting the response increases the probability of
success, but at the cost of significantly higher processing and
network load.
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+===========+
| Initial |<-------------+
+===========+ |
| |
| |
| INVITE |
| ------ |
| 1xx |
+------v v |
INVITE +-----------+ |
------ | Searching | |
1xx +-----------+ |
+------| | | +---------------->+
| | |
failure | | callee picks up |
------- | | --------------- |
>= 300 | | 200 |
| | | BYE
+------v v v v-----| | ---
INVITE +-----------+ T4 | 200
------ | Answered | ------ |
status +-----------+ status |
+------| | | |-----+ |
| +---------------->+
| |
| ACK |
| --- |
| - |
| |
+------v v |
ACK +-----------+ |
--- | Connected | |
- +-----------+ |
+------| | |
+-----------------+
event
-------
message
Figure 9: State transition diagram of server for INVITE method
A single TCP connection can serve one or more SIP transactions. A
transaction contains zero or more provisional responses followed by
one or more final responses. (Typically, transactions contain exactly
one final response, but there are exceptional circumstances, where,
for example, multiple 200 responses may be generated.)
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status code of 300 or larger, the client sends an ACK. If the
response status code is 2xx and the client is a user agent client, it
sends an ACK. If the client is not a user agent, the response is
forwarded upstream.
The client MAY close the connection at any time. The server SHOULD
NOT close the TCP connection until it has sent its final response, at
which point it MAY close the TCP connection if it wishes to. However,
normally it is the client's responsibility to close the connection.
If the server leaves the connection open, and if the client so
desires it may re-use the connection for further SIP requests or for
requests from the same family of protocols (such as HTTP or stream
control commands).
If a server needs to return a response to a client and no longer has
a connection open to that client, it MAY open a connection to the
address listed in the Via header. Thus, a proxy MUST be prepared to
receive both requests and responses on a "passive" connection.
11 Behavior of SIP Servers
This section describes behavior of a SIP server in detail. Servers
can operate in proxy or redirect mode. Proxy servers can "fork"
connections, i.e., a single incoming request spawns several outgoing
(client) requests.
A proxy server always inserts a Via header field containing its own
address into those requests that are caused by an incoming request.
Each proxy MUST insert a " branch" parameter (Section 6.40). To
prevent loops, a server MUST check if its own address is already
contained in the Via header of the incoming request.
A proxy server MAY convert a version-x SIP request or response to a
version-y request or response, where x may be larger, smaller or
equal to y.
This rule allows a proxy to serve as a go-between between
two servers that have no version of the protocol in common.
We define an "A--B proxy" as a proxy that receives SIP requests over
transport protocol A and issues requests, acting as a SIP client,
using transport protocol B. If not stated explicitly, rules apply to
any combination of transport protocols. For conciseness, we only
describe behavior with UDP and TCP, but the same rules apply for any
unreliable datagram or reliable protocol, respectively.
The detailed connection behavior for UDP and TCP is described in
Section 10.
11.1 Redirect Server
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return a response that refuses or redirects the request. After
receiving an INVITE request, once the server has gathered the list
of alternative locations or has decided to refuse the call, it
returns the final response of class 3xx or 6xx. This ends the SIP
transaction. The redirect server maintains transaction state for the
whole SIP transaction. It is up to the client to detect forwarding
loops between redirect servers.
11.2 User Agent Server
User agent servers behave similarly to redirect servers, except that
they may also accept a call with a response of class 2xx.
11.3 Stateless Proxy: Proxy Servers Issuing Single Unicast Requests
Proxies in this category issue at most a single unicast request for
each incoming SIP request, that is, they do not "fork" requests.
However, servers may choose to always operate in a mode that allows
issuing of several requests, as described in Section 11.4.
The server can forward the request and any responses. It does not
have to maintain any state for the SIP transaction. Reliability is
assured by the next redirect or stateful proxy server in the server
chain.
A proxy server SHOULD cache the result of any address translations
and the response to speed forwarding of retransmissions. After the
cache entry has been expired, the server cannot tell whether an
incoming request is actually a retransmission of an older request.
The server will treat it as a new request and commence another
search.
11.4 Proxy Server Issuing Several INVITE Requests
The server MUST respond to the request immediately with a 100
(Trying) response.
All requests carry the same Call-ID. For unicast, each of the
requests has a different (host-dependent) Request-URI. For
multicast, a single request is issued, likely with a host-independent
Request-URI. A client receiving a multicast query does not have to
check whether the host part of the Request-URI matches its own host
or domain name. To avoid response implosion, servers MUST NOT answer
multicast requests with a "404 Not Found" status code. Servers MAY
decide not to answer multicast requests if their response would be
5xx. Responses to multicast requests are multicast with the same TTL
as the request, where the TTL is derived from the ttl parameter in
the Via header (Section 6.40).
Successful responses to an INVITE request SHOULD contain a Location
header so that the following ACK or BYE bypasses the proxy search
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The following pseudo-code describes the behavior of a proxy server
issuing several requests in response to an incoming INVITE request.
The function request(r, a, b) sends a SIP request of type r to
address a, with branch id b. await_response() waits until a response
is received and returns the response. close(a) closes the TCP
connection to client with address a. response(s, l, L) sends a
response to the client with status s and list of locations L, with l
entries. ismulticast() returns 1 if the location is a multicast
address and zero otherwise. The variable timeleft indicates the
amount of time left until the maximum response time has expired. The
variable recurse indicates whether the server will recursively try
addresses returned through a 3xx response. A server MAY decide to
recursively try only certain addresses, e.g., those which are within
the same domain as the proxy server. Thus, an initial multicast
request may trigger additional unicast requests.
/* request type */
typedef enum {INVITE, ACK, BYE, OPTIONS, CANCEL, REGISTER} Method;
process_request(Method R, int N, address_t address[])
{
struct {
address_t address; /* address */
int branch; /* branch id */
int done; /* has responded */
} outgoing[];
int done[]; /* address has responded */
char *location[]; /* list of locations */
int heard = 0; /* number of sites heard from */
int class; /* class of status code */
int best = 1000; /* best response so far */
int timeleft = 120; /* sample timeout value */
int loc = 0; /* number of locations */
struct { /* response */
int status; /* response status; -2: BYE; -1: CANCEL */
int locations; /* number of redirect locations */
char *location[]; /* redirect locations */
address_t a; /* address of respondent */
int branch; /* branch identifier */
} r;
int i;
for (i = 0; i < N; i++) {
request(R, address[i], i);
outgoing[i].done = 0;
outgoing[i].branch = i;
}
while (timeleft > 0 && heard < N) {
r = await_response();
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Internet Draft SIP June 13, 1998
if (r.status < 0) {
break;
}
/* If final response, mark branch as done. */
if (class >= 2) {
heard++;
for (i = 0; i < N; i++) {
if (r.branch == outgoing[i].branch) {
outgoing[i].done = 1;
break;
}
}
}
if (class == 2) {
best = r.status;
break;
}
else if (class == 3) {
/* A server may optionally recurse. The server MUST check whether
* it has tried this location before and whether the location is
* part of the Via path of the incoming request. This check is
* omitted here for brevity. Multicast locations MUST NOT be
* returned to the client if the server is not recursing.
*/
if (recurse) {
multicast = 0;
N += r.locations;
for (i = 0; i < r.locations; i++) {
request(R, r.location[i]);
}
} else if (!ismulticast(r.location)) {
locations[loc++] = r.location;
best = r.status;
}
request(ACK, r.a, r.branch);
}
else if (class == 4) {
request(ACK, r.a, r.branch);
if (best >= 400) best = r.status;
}
else if (class == 5) {
request(ACK, r.a, r.branch);
if (best >= 500) best = r.status;
}
else if (class == 6) {
request(ACK, r.a, r.branch);
best = r.status;
break;
}
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/* CANCEL */
if (r.status == -1) {
response(200, loc, 0);
}
/* BYE */
else if (r.status == -2) {
for (i = 0; i < N; i++) {
request(BYE, address[i], i);
}
}
else {
/* We haven't heard anything useful from anybody. */
if (best == 1000) {
best = 404;
}
if (best/100 != 3) loc = 0;
response(best, loc, locations);
}
/*
* If complete or CANCELed, close the other pending transactions by
* sending CANCEL.
*/
if (r.status > 0 || r.status == -1) {
for (i = 0; i < N; i++) {
if (!outgoing[i].done) {
request(CANCEL, address[i], outgoing[i].branch);
if (tcp) close(a);
}
}
}
}
Responses are processed as follows. The process completes when all
requests have been answered by final status responses (for unicast)
or 60 seconds have elapsed (for multicast). A proxy MAY send a
CANCEL to all branches and return a 408 (Timeout) to the client after
120 seconds or more.
1xx: The proxy MAY forward the response upstream towards the client.
2xx: The proxy MUST forward the response upstream towards the client,
without sending an ACK downstream.
3xx: The proxy MUST send an ACK and MAY recurse on the listed
Location addresses. Otherwise, the locations in the response are
added to separate lists for 300, 301 and 302 responses
maintained by the proxy. The lowest-numbered 300 response is
returned to the client on completion.
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Internet Draft SIP June 13, 1998
has a lower status code than any previous 4xx and 5xx responses.
On completion, the lowest-numbered response is returned if there
were no 2xx or 3xx responses.
6xx: The proxy MUST forward the response to the client and send an
ACK. Other pending requests SHOULD be terminated with CANCEL.
The proxy server SHOULD maintain state until all responses have been
received or for 60 seconds if the request was multicast.
After receiving a 2xx or 6xx response, the server SHOULD terminate
all other pending requests by sending a CANCEL request and closing
the TCP connection, if applicable. (Terminating pending requests is
advisable as searches consume resources. Also, INVITE requests may
"ring" on a number of workstations if the callee is currently logged
in more than once.)
When operating in this mode, a proxy server MUST ignore any responses
received for Call-IDs for which it does not have a pending
transaction. (If server were to forward responses not belonging to a
current transaction using the Via field, the requesting client would
get confused if it has just issued another request using the same
Call-ID.)
If a proxy server receives a BYE request for a pending search, the
proxy MUST terminate all pending requests by sending a BYE request.
11.5 Proxy Server Issuing Several ACK and BYE Requests
In most cases, ACK and BYE requests will bypass proxies and reach
the desired party directly, keeping proxies from having to make
forwarding decisions.
User agent clients respond to ACK and BYE requests with unknown
Call-ID with status code 481 (Invalid Call-ID).
A proxy MAY maintain call state for a period of its choosing. If a
proxy still has list of destinations that it forwarded the last
INVITE to, it SHOULD direct ACK requests only to those downstream
servers. It SHOULD direct BYE to only those servers that had
previously responded with 2xx or have not yet responded to the last
INVITE.
If the proxy has no call state for a particular Call-ID and To
destination, it forwards the request to all downstream servers.
12 Security Considerations
12.1 Confidentiality and Privacy: Encryption
12.1.1 SIP Requests and Responses
Handley/Schulzrinne/Schooler [Page 73]
Internet Draft SIP June 13, 1998
the communication patterns and communication content of individuals
and thus should be protected against eavesdropping. The SIP message
body may also contain encryption keys for the session itself.
SIP supports three complementary forms of encryption to protect
privacy:
o End-to-end encryption of the SIP message body and certain
sensitive header fields;
o hop-by-hop encryption to prevent eavesdropping that tracks who
is calling whom;
o hop-by-hop encryption of Via fields to hide the route a
request has taken.
Not all of the SIP request can be encrypted end-to-end because header
fields such as To and Via need to be visible to proxies so that the
SIP request can be routed correctly. Hop-by-hop encryption encrypts
the entire SIP request or response on the wire (the request may
already have been end-to-end encrypted) so that packet sniffers or
other eavesdroppers cannot see who is calling whom. Note that proxies
can still see who is calling whom, and this information may also be
deducible by performing a network traffic analysis, so this provides
a very limited but still worthwhile degree of protection.
SIP Via fields are used to route a response back along the path
taken by the request and to prevent infinite request loops. However,
the information given by them may also provide useful information to
an attacker. Section 6.21 describes how a sender can request that Via
fields be encrypted by cooperating proxies without compromising the
purpose of the Via field.
12.2 End-to-End Encryption
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 limit the security mechanisms that may
be used, but all implementations SHOULD support PGP-based encryption.
A SIP request (or response) is end-to-end encrypted by splitting the
message to be sent into a part to be encrypted and a short header
that will remain in the clear. Some parts of the SIP message, namely
the request line, the response line and certain header fields marked
with "n" in the "enc." column in Table 3 need to be read and returned
by proxies and thus MUST NOT be encrypted end-to-end. Possibly
sensitive information that needs to be made available as plaintext
include destination address ( To) and the forwarding path ( Via) of
the call. The Authorization header MUST remain in the clear if it
contains a digital signature as the signature is generated after
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clear, but MAY be encrypted if required, in which case some proxies
MAY return a 401 (Unauthorized) status if they require a From field.
Other header fields MAY be encrypted or MAY travel in the clear as
desired by the sender. The Subject, Allow, Call-ID, and Content-
Type header fields will typically be encrypted. The Accept,
Accept-Language, Date, Expires, Priority, Require, Cseq, and
Timestamp header fields will remain in the clear.
All fields that will remain in the clear MUST precede those that will
be encrypted. The message is encrypted starting with the first
character of the first header field that will be encrypted and
continuing through to the end of the message body. If no header
fields are to be encrypted, encrypting starts with the second CRLF
pair after the last header field, as shown below. Carriage return and
line feed characters have been made visible as "$", and the encrypted
part of the message is outlined.
INVITE watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: watson@bell-telephone.com (T. A. Watson)$
From: a.g.bell@bell-telephone.com (A. Bell)$
Encryption: PGP version=5.0$
Content-Length: 224$
CSeq: 488$
$
*******************************************************
* Call-ID: 187602141351@worcester.bell-telephone.com$ *
* Subject: Mr. Watson, come here.$ *
* Content-Type: application/sdp$ *
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*******************************************************
An Encryption header field MUST be added to indicate the encryption
mechanism used. A Content-Length field is added that indicates the
length of the encrypted body. The encrypted body is preceded by a
blank line as a normal SIP message body would be.
Upon receipt by the called user agent possessing the correct
decryption key, the message body as indicated by the Content-Length
field is decrypted, and the now-decrypted body is appended to the
clear-text header fields. There is no need for an additional
Content-Length header field within the encrypted body because the
length of the actual message body is unambiguous after decryption.
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been as below. Note that the encrypted body must then include a blank
line (start with CRLF) to disambiguate between any possible SIP
header fields that might have been present and the SIP message body.
INVITE watson@boston.bell-telephone.com SIP/2.0$
Via: SIP/2.0/UDP 169.130.12.5$
To: watson@bell-telephone.com (T. A. Watson)$
From: a.g.bell@bell-telephone.com (A. Bell)$
Encryption: PGP version=5.0$
Content-Type: application/sdp$
Content-Length: 107$
$
*************************************************
* $ *
* v=0$ *
* o=bell 53655765 2353687637 IN IP4 128.3.4.5$ *
* c=IN IP4 135.180.144.94$ *
* m=audio 3456 RTP/AVP 0 3 4 5$ *
*************************************************
12.2.1 Privacy of SIP Responses
SIP requests may be sent securely using end-to-end encryption and
authentication to a called user agent that sends an insecure
response. This is allowed by the SIP security model, but is not a
good idea.
However, unless the correct behaviour is explicit, it would not
always be possible for the called user agent to infer what a
reasonable behaviour was. Thus when end-to-end encryption is used by
the request originator, the encryption key to be used for the
response SHOULD be specified in the request. If this were not done,
it might be possible for the called user agent to incorrectly infer
an appropriate key to use in the response. Thus, to prevent key-
guessing becoming an acceptable strategy, we specify that a called
user agent receiving a request that does not specify a key to be used
for the response SHOULD send that response unencrypted.
Any SIP header fields that were encrypted in a request should also be
encrypted in an encrypted response. Location response fields MAY be
encrypted if the information they contain is sensitive, or MAY be
left in the clear to permit proxies more scope for localized
searches.
12.2.2 Encryption by Proxies
Normally, proxies are not allowed to alter end-to-end header fields
and message bodies. Proxies MAY, however, encrypt an unsigned request
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Proxies may need to encrypt a SIP request if the end system
cannot perform encryption or to enforce organizational
security policies.
12.2.3 Hop-by-Hop Encryption
It is RECOMMENDED that SIP requests and responses are also protected
by security mechanisms at the transport and network layer.
12.2.4 Via field encryption
When Via fields are to be hidden, a proxy that receives a request
containing an appropriate " Hide: hop" header field (as specified in
section 6.21) SHOULD encrypt the header field. As only the proxy that
encrypts the field will decrypt it, the algorithm chosen is entirely
up to the proxy implementor. Two methods satisfy these requirements:
o The server keeps a cache of Via fields and the associated To
field, and replaces the Via field with an index into the
cache. On the reverse path, take the Via field from the cache
rather than the message.
This is insufficient to prevent message looping, and so an
additional ID must be added so that the proxy can detect loops.
This should not normally be the address of the proxy as the goal
is to hide the route, so instead a sufficiently large random
number should be used by the proxy and maintained in the cache.
Obtaining sufficiently much randomness to give sufficient
protection against clashes may be hard.
It may also be possible for replies to get directed to the wrong
originator if the cache entry gets reused, so great care must be
taken to ensure this does not happen.
o The server may use a secret key to encrypt the Via field, a
timestamp and an appropriate checksum in any such message with
the same secret key. The checksum is needed to detect whether
successful decoding has occurred, and the timestamp is
required to prevent possible response attacks and to ensure
that no two requests from the same previous hop have the same
encrypted Via field.
The latter is the preferred solution, although proxy developers may
devise other methods that might also satisfy the requirements.
12.3 Message Integrity and Access Control: Authentication
An active attacker may be able to modify and replay SIP requests and
responses unless protective measures are taken. In practice, the same
cryptographic measures that are used to ensure the authenticity of
the SIP message also serve to authenticate the originator of the
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Transport-layer or network-layer authentication may be used for hop-
by-hop authentication. SIP also extends the HTTP WWW-Authenticate
(Section 6.42 and Authorization (Section 6.11) header and their
Proxy- counterparts to include cryptographically strong signatures.
SIP also supports the HTTP "basic" authentication scheme [33] that
offers a very rudimentary mechanism of ascertaining the identity of
the caller.
Since SIP requests are often sent to parties with which no
prior communication relationship has existed, we do not
specify authentication based on shared secrets.
SIP requests may be authenticated using the Authorization header
field to include a digital signature of certain header fields, the
request method and version number and the payload, none of which are
modified between client and called user agent. The Authorization
header field may be used in requests to end-to-end authenticate the
request originator to proxies and the called user agent, and in
responses to authenticate the called user agent or proxies returning
their own failure codes. It does not provide hop-by-hop
authentication, which may be provided if required using the IPSEC
Authentication Header.
SIP does not dictate which digital signature scheme is used for
authentication, but does define how to provide authentication using
PGP in Section 13.
To sign a SIP request, the order of the SIP header fields is
important. Via header fields MUST precede all other SIP header
fields as these are modified in transit. When an Authorization
header field is present, it indicates that all the header fields
following the Authorization header field have been included in the
signature. To sign a request, a client removes all of the SIP header
from before where the Authorization field will be added. It then
prepends the request method (in upper case) followed by the SIP
version number field (in upper case) directly to the start of the
message with no whitespace, CR or LF characters inserted. This
extended message is what is signed.
For example, if the SIP request is to be:
INVITE watson@boston.bell-telephone.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
Authorization: PGP version=5.0, signature=...
From: a.g.bell@bell-telephone.com (A. Bell)
To: watson@bell-telephone.com (T. A. Watson)
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
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o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Then the data block that is signed is:
INVITESIP/2.0From: a.g.bell@bell-telephone.com (A. Bell)
To: watson@bell-telephone.com (T. A. Watson)
Call-ID: 187602141351@worcester.bell-telephone.com
Subject: Mr. Watson, come here.
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
Note that if a message is encrypted and authenticated using a digital
signature, when the message is generated encryption is performed
before the digital signature is generated. On receipt, the digital
signature is checked before decryption.
A client MAY require that a server sign its response by including a
Require: org.ietf.sip.signed-response request header field. The
client indicates the desired authentication method via the WWW-
Authenticate header.
The correct behaviour in handling unauthenticated responses to a
request that requires authenticated responses is described in section
12.3.1.
12.3.1 Trusting responses
It may be possible for an eavesdropper to listen to requests and to
inject unauthenticated responses that would terminate, redirect or
otherwise interfere with a call. (Even encrypted requests contain
enough information to fake a response.)
Client should be particularly careful with 3xx redirection responses.
Thus a client receiving, for example, a 301 (Moved Permanently) which
was not authenticated when the public key of the called user agent is
known to the client, and authentication was requested in the request
SHOULD be treated as suspicious. The correct behaviour in such a case
would be for the called-user to form a dated response containing the
Location field to be used, to sign it, and give this signed stub
response to the proxy that will provide the redirection. Thus the
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SHOULD NOT automatically redirect such a request to the new location
without alerting the user to the authentication failure before doing
so.
Another problem might be responses such as 6xx failure responses
which would simply terminate a search, or "4xx" and "5xx" response
failures.
If TCP is being used, a proxy SHOULD treat 4xx and 5xx responses as
valid, as they will not terminate a search. However, 6xx responses
from a rogue proxy may terminate a search incorrectly. 6xx responses
SHOULD be authenticated if requested by the client, and failure to do
so SHOULD cause such a client to ignore the 6xx response and continue
a search.
With UDP, the same problem with 6xx responses exists, but also an
active eavesdropper can generate 4xx and 5xx responses that might
cause a proxy or client to believe a failure occurred when in fact it
did not. Typically 4xx and 5xx responses will not be signed by the
called user agent, and so there is no simple way to detect these
rogue responses. This problem is best prevented by using hop-by-hop
encryption of the SIP request, which removes any additional problems
that UDP might have over TCP.
These attacks are prevented by having the client require response
authentication and dropping unauthenticated responses. A server user
agent that cannot perform response authentication responds using the
normal Require response of 420 (Bad Extension).
12.4 Callee Privacy
User location and SIP-initiated calls may violate a callee's privacy.
An implementation SHOULD be able to restrict, on a per-user basis,
what kind of location and availability information is given out to
certain classes of callers.
12.5 Known Security Problems
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.
13 SIP Security Using PGP
13.1 PGP Authentication Scheme
The "pgp" authentication scheme is based on the model that the client
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if it has access to the public key, preferably signed by a trusted
third party.
13.1.1 The WWW-Authenticate Response Header
WWW-Authenticate = "WWW-Authenticate" ":" "pgp" pgp-challenge
pgp-challenge = 1# ( realm | pgp-version | pgp-algorithm )
realm = "realm" "=" realm-value
realm-value = quoted-string
pgp-version = "version" "=" digit *( "." digit ) *letter
pgp-algorithm = "algorithm" "=" ( "md5" | "sha1" | token )
The meanings of the values of the parameters used above are as
follows:
realm: A string to be displayed to users so they know which identity
to use. This string should contain at least the name of the host
performing the authentication and might additionally indicate
the collection of users who might have access. An example might
be " Users with call-out privileges ".
pgp-algorithm: A string indicating the PGP message integrity check
(MIC) to be used to produce the signature. If this not present
it is assumed to be "md5". The currently defined values are
"md5" for the MD5 checksum, and "sha1" for the SHA.1 algorithm.
pgp-version: The version of PGP that the client MUST use. Common
values are "2.6.2" and "5.0". The default is 5.0.
Example:
WWW-Authenticate: pgp version="5.0",
realm="Your Startrek identity, please", algorithm="md5"
13.1.2 The Authorization Request Header
The client is expected to retry the request, passing an Authorization
header line, which is defined as follows.
Authorization ___ "Authorization" ":" "pgp" pgp-response
pgp-response ___ 1# (realm | pgp-version | pgp-signature | signed-by)
pgp-signature ___ "signature" "=" quoted-string
signed-by ___ "signed-by" "=" URI
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unless the signed-by parameter is provided.
pgp-signature: The PGP ASCII-armored signature, as it appears between
the "BEGIN PGP MESSAGE" and "END PGP MESSAGE" delimiters,
without the version indication. The signature is included
without any linebreaks.
The signature is computed across the request method, request version
and header fields following the Authorization header and the message
body, in the same order as they appear in the message. The request
method and version are prepended to the header fields without any
white space. The signature is computed across the headers as sent,
including any folding and the terminating CRLF. The CRLF following
the Authorization header is NOT included in the signature.
Using the ASCII-armored version is about 25% less space-
efficient than including the binary signature, but it is
significantly easier for the receiver to piece together.
Versions of the PGP program always include the full
(compressed) signed text in their output unless ASCII-
armored mode ( -sta ) is specified. Typical signatures are
about 200 bytes long. -- The PGP signature mechanism allows
the client to simply pass the request to an external PGP
program. This relies on the requirement that proxy servers
are not allowed to reorder or change header fields.
realm: The realm is copied from the corresponding WWW-Authenticate
header field parameter.
signed-by: If and only if the request was not signed by the entity
listed in the From header, the signed-by header indicates the
name of the signing entity, expressed as a URI.
Receivers of signed SIP messages SHOULD discard any end-to-end header
fields above the Authorization header, as they may have been
maliciously added en route by a proxy.
Example:
Authorization: pgp version="5.0",
realm="Your Startrek identity, please",
signature="iQB1AwUBNNJiUaYBnHmiiQh1AQFYsgL/Wt3dk6TWK81/b0gcNDf
VAUGU4rhEBW972IPxFSOZ94L1qhCLInTPaqhHFw1cb3lB01rA0RhpV4t5yCdUt
SRYBSkOK29o5e1KlFeW23EzYPVUm2TlDAhbcjbMdfC+KLFX
=aIrx"
13.2 PGP Encryption Scheme
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Encryption ___ "Encryption" ":" "pgp" pgp-eparams
pgp-eparams ___ 1# ( pgp-version | pgp-encoding )
pgp-encoding ___ "encoding" "=" "ascii" | token
encoding: Describes the encoding or "armor" used by PGP. The value
"ascii" refers to the standard PGP ASCII armor, without the
lines containing "BEGIN PGP MESSAGE" and "END PGP MESSAGE" and
without the version identifier. By default, the encrypted part
is included as binary.
Example:
Encryption: pgp version="2.6.2", encoding="ascii"
13.3 Response-Key Header Field for PGP
Response-Key ___ "Response-Key" ":" "pgp" pgp-eparams
pgp-eparams ___ 1# ( pgp-version | pgp-encoding | pgp-key)
pgp-key ___ "key" "=" quoted-string
If ASCII encoding has been requested via the encoding parameter, the
key parameter contains the user's public key as extracted with the
"pgp -kxa user ".
Example:
Response-Key: pgp version="2.6.2", encoding="ascii",
key="mQBtAzNWHNYAAAEDAL7QvAdK2utY05wuUG+ItYK5tCF8HNJM60sU4rLaV+eUnkMk
mOmJWtc2wXcZx1XaXb2lkydTQOesrUR75IwNXBuZXPEIMThEa5WLsT7VLme7njnx
sE86SgWmAZx5ookIdQAFEbQxSGVubmluZyBTY2h1bHpyaW5uZSA8c2NodWx6cmlu
bmVAY3MuY29sdW1iaWEuZWR1Pg==
=+y19"
14 Examples
14.1 Registration
A user at host saturn.bell-tel.com registers on start-up, via
multicast, with the local SIP server named sip.bell-tel.com the
example, the user agent on saturn expects to receive SIP requests on
UDP port 3890.
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To: sip:watson@bell-tel.com
Location: sip:saturn.bell-tel.com:3890;transport=udp
Expires: 7200
CSeq: 1 REGISTER
The registration expires after two hours. Any future invitations for
watson@bell-tel.com arriving at sip.bell-tel.com will now be
redirected to watson@saturn.bell-tel.com , UDP port 3890.
If Watson wants to be reached elsewhere, say, an on-line service he
uses while traveling, he updates his reservation after first
cancelling any existing locations:
C->S: REGISTER sip:@bell-tel.com SIP/2.0
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Expire: 0
Location: *
C->S: REGISTER sip:@bell-tel.com SIP/2.0
From: sip:watson@bell-tel.com
To: sip:watson@bell-tel.com
Location: sip:tawatson@example.com
Now, the server will forward any request for Watson to the server at
example.com , using the Request-URI tawatson@example.com
It is possible to use third-party registration. Here, the secretary
jon.diligent registers his boss:
C->S: REGISTER sip:@bell-tel.com SIP/2.0
From: sip:jon.diligent@bell-tel.com
To: sip:watson@bell-tel.com
Location: sip:tawatson@example.com
The request could be send to either the registrar at bell-tel.com or
the server at example.com example.com would proxy the request to the
address indicated in the Request-URI. Then, Max-Forwards header
could be used to restrict the registration to that server.
14.2 Invitation to Multicast Conference
The first example invites schooler@vlsi.cs.caltech.edu to a multicast
session. All examples use the Session Description Protocol (SDP) (RFC
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14.2.1 Request
C->S: INVITE sip:schooler@vlsi.cs.caltech.edu SIP/2.0
Via: SIP/2.0/UDP 239.128.16.254 ;ttl=16
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: Mark Handley <sip:mjh@isi.edu>
To: Eve Schooler <sip:schooler@caltech.edu>
Subject: SIP will be discussed, too
Call-ID: 19971205T234505.56.78@oregon.isi.edu
Content-Type: application/sdp
CSeq: 4711 INVITE
Content-Length: 187
v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
e=mbone@somewhere.com
c=IN IP4 224.2.0.1/127
t=0 0
m=audio 3456 RTP/AVP 0
The Via fields list the hosts along the path from invitation
initiator (the last element of the list) towards the invitee. In the
example above, the message was last multicast to the administratively
scoped group 239.128.16.254 with a ttl of 16 from the host
131.215.131.131
The request header above states that the request was initiated by
mjh@isi.edu from the host 128.16.64.19 schooler@caltech.edu is being
invited; the message is currently being routed to
schooler@vlsi.cs.caltech.edu
In this case, the session description is using the Session
Description Protocol (SDP), as stated in the Content-Type header.
The header is terminated by an empty line and is followed by a
message body containing the session description.
14.2.2 Response
The called user agent, directly or indirectly through proxy servers,
indicates that it is alerting ("ringing") the called party:
S->C: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 239.128.16.254 ;ttl=16 ;branch=17
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Internet Draft SIP June 13, 1998
To: Eve Schooler <sip:schooler@caltech.edu>
From: Mark Handley <sip:mjh@isi.edu>
Call-ID: 19971205T234505.56.78@north.east.isi.edu
Location: sip:es@jove.cs.caltech.edu
CSeq: 4711 INVITE
A sample response to the invitation is given below. The first line of
the response states the SIP version number, that it is a 200 (OK)
response, which means the request was successful. The Via headers
are taken from the request, and entries are removed hop by hop as the
response retraces the path of the request. A new authentication field
MAY be added by the invited user's agent if required. The Call-ID is
taken directly from the original request, along with the remaining
fields of the request message. The original sense of From field is
preserved (i.e., it is the session initiator).
In addition, the Location header gives details of the host where the
user was located, or alternatively the relevant proxy contact point
which should be reachable from the caller's host.
S->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP 239.128.16.254 16 ;branch=17
Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
Via: SIP/2.0/UDP north.east.isi.edu
From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@north.east.isi.edu
Location: sip:es@jove.cs.caltech.edu
CSeq: 4711 INVITE
The caller confirms the invitation by sending a request to the
location named in the Location header:
C->S: ACK sip:es@jove.cs.caltech.edu SIP/2.0
From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@oregon.isi.edu
CSeq: 4711 ACK
14.3 Two-party Call
For two-party Internet phone calls, the response must contain a
description of where to send the data. In the example below, Bell
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C->S: INVITE sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
Subject: Mr. Watson, come here.
CSeq: 17 INVITE
Content-Type: application/sdp
Content-Length: ...
v=0
o=bell 53655765 2353687637 IN IP4 128.3.4.5
c=IN IP4 135.180.144.94
m=audio 3456 RTP/AVP 0 3 4 5
S->C: SIP/2.0 100 Trying
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
Content-Length: 0
S->C: SIP/2.0 180 Ringing
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
Content-Length: 0
S->C: SIP/2.0 182 Queued, 2 callers ahead
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
Content-Length: 0
S->C: SIP/2.0 182 Queued, 1 caller ahead
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
Content-Length: 0
S->C: SIP/2.0 200 OK
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: sip:watson@bell-tel.com
Call-ID: 1985853074@kensington.bell-tel.com
CSeq: 17 INVITE
Location: sip:watson@boston.bell-tel.com
Content-Length: ...
v=0
o=watson 4858949 4858949 IN IP4 192.1.2.3
c=IN IP4 135.180.161.25
m=audio 5004 RTP/AVP 0 3
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The example illustrates the use of informational status responses.
Here, the reception of the call is confirmed immediately (100), then,
possibly after some database mapping delay, the call rings (180) and
is then queued, with periodic status updates.
Watson can only receive PCMU and GSM. Note that Watson's list of
codecs may or may not be a subset of the one offered by Bell, as each
party indicates the data types it is willing to receive. Watson will
send audio data to port 3456 at 135.180.144.94, Bell will send to
port 5004 at 135.180.161.25.
By default, the media session is one RTP session. Watson will receive
RTCP packets on port 5005, while Bell will receive them on port 3457.
Since the two sides have agree on the set of media, Watson confirms
the call without enclosing another session description:
C->S: ACK sip:watson@boston.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP 169.130.12.5
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
CSeq: 17 ACK
Content-Length: 0
14.4 Terminating a Call
To terminate a call, caller or callee can send a BYE request:
C->S: BYE sip:watson@boston.bell-tel.com SIP/2.0
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. A. Watson <sip:watson@bell-tel.com>
Call-ID: 1985853074@kensington.bell-tel.com
CSeq: 18 BYE
If the callee wants to abort the call, it simply reverses the To and
From fields. Note that it is unlikely that an BYE from the callee
will traverse the same proxies as the original INVITE.
14.5 Forking Proxy
In this example, Bell ( a.g.bell@bell-tel.com ) (C), currently seated
at host c.bell-tel.com wants to call Watson ( t.watson@ieee.org ). At
the time of the call, Watson is logged in at two workstations,
watson@x.bell-tel.com (X) and watson@y.bell-tel.com (Y), and has
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Internet Draft SIP June 13, 1998
(H), as well as a permanent registration at watson@acm.org (A). For
brevity, the examples omit the session description.
Watson's user agent sends the invitation to the SIP server for the
ieee.org domain:
C->P: INVITE sip:watson@ieee.org SIP/2.0
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@kensington.bell-tel.com
CSeq: 19 INVITE
Via: SIP/2.0/UDP c.bell-tel.com
The SIP server tries the four addresses in parallel. It sends the
following message to the home machine:
P->H: INVITE sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
This request immediately yields a 404 (Not Found) response, since
Watson is not currently logged in at home:
H->P: SIP/2.0 404 Not Found
Via: SIP/2.0/UDP proxy.ieee.org ;branch=1
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
The proxy ACKs the response so that host H can stop retransmitting
it:
P->H: ACK sip:watson@h.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=1
If-Match: "4711"
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Handley/Schulzrinne/Schooler [Page 89]
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Also, P attempts to reach Watson through the ACM server:
P->A: INVITE sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=2
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
In parallel, the next attempt proceeds, with an INVITE to X and Y:
P->X: INVITE sip:watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=3
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
P->Y: INVITE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=4
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
As it happens, both Watson at X and a colleague in the other lab at
host Y hear the phones ringing and pick up. Both X and Y return 200s
via the proxy to Bell. The tag URI parameter is not strictly
necessary here, since the Location header is unambiguous.
X->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.ieee.org ;branch=3
Via: SIP/2.0/UDP c.bell-tel.com
Location: sip:t.watson@x.bell-tel.com;tag=1620
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
Y->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.ieee.org ;branch=4
Via: SIP/2.0/UDP c.bell-tel.com
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To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 INVITE
Both responses are forwarded to Bell, using the Via information. At
this point, the ACM server is still searching its database. P can now
cancel this attempt:
P->A: CANCEL sip:watson@acm.org SIP/2.0
Via: SIP/2.0/UDP proxy.ieee.org ;branch=2
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 CANCEL
The ACM server gladly stops its neural-network database search and
responds with a 200. The 200 will not travel any further, since P is
the last Via stop.
A->P: SIP/2.0 200 OK
Via: SIP/2.0/UDP proxy.ieee.org ;branch=3
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 CANCEL
Bell gets the two 200 responses from X and Y in short order. Bell's
reaction now depends on his software. He can either send an ACK to
both if human intelligence is needed to determine who he wants to
talk to or he can automatically reject one of the two calls. Here, he
acknowledges both, separately and directly to the final destination:
C->X: ACK sip:watson@x.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
If-Match: "1620"
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 ACK
C->Y: ACK sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
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To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 19 ACK
After a brief discussion between the three, it becomes clear that
Watson is at X, thus Bell sends a BYE to Y, which is replied to:
C->Y: BYE sip:watson@y.bell-tel.com SIP/2.0
Via: SIP/2.0/UDP c.bell-tel.com
If-Match: "2016"
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 20 BYE
Y->C: SIP/2.0 200 OK
Via: SIP/2.0/UDP c.bell-tel.com
From: A. Bell <sip:a.g.bell@bell-tel.com>
To: T. Watson <sip:t.watson@ieee.org>
Call-ID: 1985853074@c.bell-tel.com
CSeq: 20 BYE
14.6 Redirects
Replies with status codes 301 (Moved Permanently) or 302 (Moved
Temporarily) specify another location using the Location field:
S->C: SIP/2.0 302 Moved temporarily
Via: SIP/2.0/UDP csvax.cs.caltech.edu ;branch=8348
Via: SIP/2.0/UDP 128.16.64.19
From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu
Call-ID: 3779067998@oregon.isi.edu
Location: sip:@239.128.16.254;ttl=16;transport=udp
CSeq: 19 INVITE
Content-Length: 0
In this example, the proxy located at csvax.cs.caltech.edu is being
advised to contact the multicast group 239.128.16.254 with a ttl of
16 and UDP transport. In normal situations, a server would not
suggest a redirect to a local multicast group unless, as in the above
situation, it knows that the previous proxy or client is within the
scope of the local group. If the request is redirected to a multicast
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may be scoped; this allows a proxy within the appropriate scope
regions to make the query.
14.7 Alternative Services
An example of a 350 (Alternative Service) response is:
S->C: SIP/2.0 350 Alternative Service
Via: SIP/2.0/UDP 131.215.131.131
Via: SIP/2.0/UDP 128.16.64.19
From: sip:mjh@isi.edu
To: schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@oregon.isi.edu
Location: sip:ecorder@131.215.131.131
CSeq: 19 INVITE
Content-Type: application/sdp
Content-Length: 146
v=0
o=mm-server 2523535 0 IN IP4 131.215.131.131
s=Answering Machine
i=Leave an audio message
c=IN IP4 131.215.131.131
t=0 0
m=audio 12345 RTP/AVP 0
In this case, the answering server provides a session description
that describes an "answering machine". If the invitation initiator
decides to take advantage of this service, it should send an
invitation request to the answering machine at 131.215.131.131 with
the session description provided (modified as appropriate for a
unicast session to contain the appropriate local address and port for
the invitation initiator). This request SHOULD contain a different
Call-ID from the one in the original request. An example would be:
C->S: INVITE sip:mm-server@131.215.131.131 SIP/2.0
Via: SIP/2.0/UDP 128.16.64.19
From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@128.16.64.19
CSeq: 20 INVITE
Content-Type: application/sdp
Content-Length: 146
v=0
o=mm-server 2523535 0 IN IP4 131.215.131.131
s=Answering Machine
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t=0 0
m=audio 26472 RTP/AVP 0
Invitation initiators MAY choose to treat a 350 (Alternative Service)
response as a failure if they wish to do so.
14.8 Negotiation
An example of a 606 (Not Acceptable) response is:
S->C: SIP/2.0 606 Not Acceptable
From: sip:mjh@isi.edu
To: sip:schooler@cs.caltech.edu
Call-ID: 19971205T234505.56.78@128.16.64.19
Location: sip:mjh@131.215.131.131
Warning: 606.1 Insufficient bandwidth (only have ISDN),
606.3 Incompatible format,
606.4 Multicast not available
Content-Type: application/sdp
Content-Length: 50
v=0
s=Lets talk
b=CT:128
c=IN IP4 131.215.131.131
m=audio 3456 RTP/AVP 7 0 13
m=video 2232 RTP/AVP 31
In this example, the original request specified 256 kb/s total
bandwidth, and the response states that only 128 kb/s is available.
The original request specified GSM audio, H.261 video, and WB
whiteboard. The audio coding and whiteboard are not available, but
the response states that DVI, PCM or LPC audio could be supported in
order of preference. The response also states that multicast is not
available. In such a case, it might be appropriate to set up a
transcoding gateway and re-invite the user.
14.9 OPTIONS Request
A caller Alice can use an OPTIONS request to find out the
capabilities of a potential callee Bob, without "ringing" the
designated address. Bob returns a description indicating that he is
capable of receiving audio and video, with a list of supported
encodings.
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To: Bob <sip:bob@example.com>
Accept: application/sdp
S->C: SIP/2.0 200 OK
Content-Length: 81
Content-Type: application/sdp
v=0
m=audio 0 RTP/AVP 0 1 3 99
m=video 0 RTP/AVP 29 30
a=rtpmap:99 SX7300/8000
A Minimal Implementation
A.1 Client
All clients MUST be able to generate the INVITE and ACK requests
and MUST be able to include the Call-ID, Content-Length, Content-
Type, CSeq, From and To headers. A minimal implementation MUST
understand SDP (RFC 2327, [7]). In responses, it must be able to
parse the Call-ID, Content-Length, Content-Type, Require headers.
It MUST be able to recognize the status code classes 1 through 6 and
act accordingly.
The following capability sets build on top of a minimal
implementation:
Basic: A basic implementation SHOULD add support for the BYE method
to allow the interruption of a pending call attempt. It SHOULD
include a User-Agent header in its requests and indicate its
preferred language in the Accept-Language header.
Redirection: To support call forwarding, a client needs to be able to
understand the Location header, but only the SIP-URL part, not
the parameters.
Negotiation: A client MUST be able to request the OPTIONS method and
understand the 380 (Alternative Service) status and the Location
parameters to participate in terminal and media negotiation. It
SHOULD be able to parse the Warning response header to provide
useful feedback to the caller.
Authentication: If a client wishes to invite callees that require
caller authentication, it must be able to recognize the 401
(Unauthorized) status code, must be able to generate the
Authorization request header and MUST understand the WWW-
Authenticate response header.
If a client wishes to use proxies that require caller authentication,
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Internet Draft SIP June 13, 1998
header and understand the Proxy-Authenticate response header.
A.2 Server
A minimally compliant server implementation MUST understand the
INVITE, ACK, OPTIONS and BYE requests. It MUST parse and generate,
as appropriate, the Call-ID, Content-Length, Content-Type,
Expires, From, Max-Forwards, Require, To and Via headers. It MUST
echo the CSeq header in the response. It SHOULD include the Server
header in its responses.
A.3 Header Processing
Table 4 lists the headers that different implementations support. UAC
refers to a user-agent client (calling user agent), UAS to a user-
agent server (called user-agent).
B Usage of SDP
By default, the nth media session in a unicast INVITE request will
become a single RTP session with the nth media session in the
response. Thus, the callee should be careful to order media
descriptions appropriately.
It is assumed that if caller or callee include a particular media
type, they want to both send and receive media data. If the callee
does not want to send a particular media type, it should mark the
media entry as recvonly receive a particular media type, it may mark
it as sendonly wants to neither receive nor send a particular media
type, it should set the port to zero. (RTCP ports are not needed in
this case.)
The caller should include all media types that it is willing to send
so that the receiver can provide matching media descriptions.
The callee should set the port to zero if callee and caller only want
to receive a media type.
C Summary of Augmented BNF
In this specification we use the Augmented Backus-Naur Form notation
described in RFC 2234 [24]. For quick reference, the following is a
brief summary of the main features of this ABNF.
"abc"
The case-insensitive string of characters "abc" (or "Abc",
"aBC", etc.);
%d32
The character with ASCII code decimal 32 (space);
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zero of more instances of term;
3*term
three or more instances of term;
2*4term
two, three or four instances of term;
[ term ]
term is optional;
term1 term2 term3
set notation: term1, term2 and term3 must all appear but
their order is unimportant;
term1 | term2
either term1 or term2 may appear but not both;
#term
a comma separated list of term;
2#term
a comma separated list of term containing at least 2 items;
2#4term
a comma separated list of term containing 2 to 4 items.
Common Tokens
Certain tokens are used frequently in the BNF this document, and not
defined elsewhere. Their meaning is well understood but we include it
here for completeness.
CR = %d13 ; carriage return character
LF = %d10 ; line feed character
CRLF = CR LF ; typically the end of a line
SP = %d32 ; space character
TAB = %d09 ; tab character
LWS = *( SP | TAB) ; linear whitespace
DIGIT = "0" .. "9" ; a single decimal digit
D IANA Considerations
Section 4.4 describes a name space and mechanism for registering SIP
options.
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type UAC proxy UAS
____________________________________________________
Accept R - o o
Accept-Language R - b b
Allow 405 o - -
Authorization R a o a
Call-ID g m m m
Content-Length g m m m
Content-Type g m - m
CSeq g o m m
Date g o o o
Encryption g e - e
Expires g - o o
From R m o m
Hide R o o -
Location R - - -
Location r r r -
Max-Forwards R - b -
Organization R - o o
Proxy-Authenticate 407 a - -
Proxy-Authorization R - a -
Proxy-Require R - m -
Priority R - o o
Require R m - m
Retry-After 600,603 o o -
Response-Key R - - e
Server r o o -
Subject R o o o
Timestamp g o o o
To g m m m
Unsupported r b b -
User-Agent R - o o
Via g - m m
Warning r o o -
WWW-Authenticate 401 a - -
Table 4: This table indicates which systems should be able to parse
which response header fields. Type is as table 3. "-" indicates the
field is not meaningful to this system (although it might be
generated by it). "m" indicates the field MUST be understood. "b"
indicates the field SHOULD be understood by a Basic implementation.
"r" indicates the field SHOULD be understood if the system claims to
understand redirection. "a" indicates the field SHOULD be understood
if the system claims to support authentication. "e" indicates the
field SHOULD be understood if the system claims to support
encryption. "o" indicates support of the field is purely optional.
Changes in Version -06
Since version -05, the following changes have been made.
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o Clarified definition of transaction and relation to ACK (since
ACK is not really part of the transaction, even though it has
the same CSeq and Call-ID).
o Postscript version had old proxy and redirect drawing.
o Vestiges of URL short form removed.
o Clarified that clients of all sorts (user agent, proxy) MAY
issue CANCEL.
o CSeq reduced to 32 bits for implementation convenience.
o Bug fixes in process_request code.
o Moved "Queued" response to main spec, since the decision to
queue can (will typically) also be taken by the callee, rather
than just the caller. This does not complicate the client or
server behavior.
o ETag removed in favor of the tag parameter in the URI.
E Open Issues
E.1 H.323
Problem: Detailed interaction with H.323 and H.245.
Solution: Leave to separate document.
Status: Closed.
E.2 REGISTER
Problem: How does the UA get a "personalized" multicast address for
multicast searches? It would be helpful if the server could
return an indication of any local search multicast address the
user agent is supposed to listen on. The server might also want
to indicate whether all outgoing calls should be proxied through
the server. Use of message bodies for REGISTER requests.
Solution: Leave for follow-on work. Possible to use Location with 200
response, but already otherwise used to indicate registrations.
Status: Closed.
E.3 Max-Forwards
Problem: Extend Max-Forwards with a max fan out field. [Not really
related.] How do you limit fanout for multicast? What's the
advantage of doing this?
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actually set this value to anything meaningful.
Status: Closed.
E.4 Cancellation and BYE
Problem: Determine how to cancel
- searches
- ringing
- non-selected end systems
- calls
Solution: In spec.
Status: Closed.
E.5 IPv6 URLs
Problem: Numeric IPv6 addresses in URLs.
Solution: None; wait for others to take the lead. Note that this
affects only numeric addresses, which should be rarely used.
Status: Defered.
F Acknowledgments
We wish to thank the members of the IETF MMUSIC WG for their comments
and suggestions. Detailed comments were provided by Dave Devanathan,
Yaron Goland, Christian Huitema, Jonathan Lennox, Jonathan Rosenberg,
Moshe J. Sambol, and Eric Tremblay.
This work is based, inter alia, on [34,35].
G Authors' Addresses
Mark Handley
USC Information Sciences Institute
c/o MIT Laboratory for Computer Science
545 Technology Square
Cambridge, MA 02139
USA
electronic mail: mjh@isi.edu
Henning Schulzrinne
Dept. of Computer Science
Columbia University
Handley/Schulzrinne/Schooler [Page 100]
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USA
electronic mail: schulzrinne@cs.columbia.edu
Eve Schooler
Computer Science Department 256-80
California Institute of Technology
Pasadena, CA 91125
USA
electronic mail: schooler@cs.caltech.edu
H Bibliography
[1] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Internet Draft, Internet Engineering Task Force,
Feb. 1998. Work in progress.
[2] M. Handley, "SAP: Session announcement protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress.
[3] R. Pandya, "Emerging mobile and personal communication systems,"
IEEE Communications Magazine , vol. 33, pp. 44--52, June 1995.
[4] M. Handley, J. Crowcroft, C. Bormann, and J. Ott, "The internet
multimedia conferencing architecture," Internet Draft, Internet
Engineering Task Force, July 1997. Work in progress.
[5] B. Braden, L. Zhang, S. Berson, S. Herzog, and S. Jamin,
"Resource ReSerVation protocol (RSVP) -- version 1 functional
specification," RFC 2205, Internet Engineering Task Force, Oct. 1997.
[6] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
transport protocol for real-time applications," RFC 1889, Internet
Engineering Task Force, Jan. 1996.
[7] M. Handley and V. Jacobson, "SDP: session description protocol,"
RFC 2327, Internet Engineering Task Force, Apr. 1998.
[8] P. Lantz, "Usage of H.323 on the Internet," Internet Draft,
Internet Engineering Task Force, Feb. 1997. Work in progress.
[9] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[10] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-
Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[11] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
1994.
Handley/Schulzrinne/Schooler [Page 101]
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Force, Oct. 1996.
[13] C. Partridge, "Mail routing and the domain system," RFC STD 14,
974, Internet Engineering Task Force, Jan. 1986.
[14] P. Mockapetris, "Domain names - implementation and
specification," RFC STD 13, 1035, Internet Engineering Task Force,
Nov. 1987.
[15] B. Braden, "Requirements for internet hosts - application and
support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
1989.
[16] D. Zimmerman, "The finger user information protocol," RFC 1288,
Internet Engineering Task Force, Dec. 1991.
[17] S. Williamson, M. Kosters, D. Blacka, J. Singh, and K. Zeilstra,
"Referral whois (rwhois) protocol V1.5," RFC 2167, Internet
Engineering Task Force, June 1997.
[18] W. Yeong, T. Howes, and S. Kille, "Lightweight directory access
protocol," RFC 1777, Internet Engineering Task Force, Mar. 1995.
[19] L. Masinter, P. Hoffman, and J. Zawinski, "The mailto URL
scheme," Internet Draft, Internet Engineering Task Force, Jan. 1998.
Work in progress.
[20] T. Berners-Lee, "Universal resource identifiers in WWW: a
unifying syntax for the expression of names and addresses of objects
on the network as used in the world-wide web," RFC 1630, Internet
Engineering Task Force, June 1994.
[21] T. Berners-Lee, L. Masinter, and R. Fielding, "Uniform resource
identifiers (URI): generic syntax," Internet Draft, Internet
Engineering Task Force, Mar. 1998. Work in progress.
[22] A. Vaha-Sipila, "URLs for telephony," Internet Draft, Internet
Engineering Task Force, Feb. 1998. Work in progress.
[23] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
2279, Internet Engineering Task Force, Jan. 1998.
[24] D. Crocker and P. Overell, "Augmented BNF for syntax
specifications: ABNF," RFC 2234, Internet Engineering Task Force,
Nov. 1997.
[25] W. R. Stevens, TCP/IP illustrated: the protocols , vol. 1.
Reading, Massachusetts: Addison-Wesley, 1994.
[26] J. Mogul and S. Deering, "Path MTU discovery," RFC 1191,
Internet Engineering Task Force, Nov. 1990.
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messages," RFC STD 11, 822, Internet Engineering Task Force, Aug.
1982.
[28] E. M. Schooler, "A multicast user directory service for
synchronous rendezvous," Master's Thesis CS-TR-96-18, Department of
Computer Science, California Institute of Technology, Pasadena,
California, Aug. 1996.
[29] P. Resnick, "Internet message format standard," Internet Draft,
Internet Engineering Task Force, Mar. 1998. Work in progress.
[30] J. Mogul, T. Berners-Lee, L. Masinter, P. Leach, R. Fielding, H.
Nielsen, and J. Gettys, "Hypertext transfer protocol -- HTTP/1.1,"
Internet Draft, Internet Engineering Task Force, Mar. 1998. Work in
progress.
[31] P. Leach and R. Salz, "UUIDs and GUIDs," Internet Draft,
Internet Engineering Task Force, Feb. 1998. Work in progress.
[32] M. Elkins, "MIME security with pretty good privacy (PGP)," RFC
2015, Internet Engineering Task Force, Oct. 1996.
[33] J. Franks, E. Sink, P. Leach, J. Hostetler, P. Hallam-Baker, L.
Stewart, S. Lawrence, and A. Luotonen, "HTTP authentication: Basic
and digest access authentication," Internet Draft, Internet
Engineering Task Force, Mar. 1998. Work in progress.
[34] E. M. Schooler, "Case study: multimedia conference control in a
packet-switched teleconferencing system," Journal of Internetworking:
Research and Experience , vol. 4, pp. 99--120, June 1993. ISI
reprint series ISI/RS-93-359.
[35] H. Schulzrinne, "Personal mobility for multimedia services in
the Internet," in European Workshop on Interactive Distributed
Multimedia Systems and Services , (Berlin, Germany), Mar. 1996.
Full Copyright Statement
Copyright (c) The Internet Society (1998). 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
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
Handley/Schulzrinne/Schooler [Page 103]
Internet Draft SIP June 13, 1998
English.
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Table of Contents
1 Introduction ........................................ 2
1.1 Overview of SIP Functionality ....................... 2
1.2 Terminology ......................................... 3
1.3 Definitions ......................................... 4
1.4 Summary of SIP Operation ............................ 7
1.4.1 SIP Addressing ...................................... 7
1.4.2 Locating a SIP Server ............................... 8
1.4.3 SIP Transaction ..................................... 9
1.4.4 SIP Invitation ...................................... 10
1.4.5 Locating a User ..................................... 13
1.4.6 Changing an Existing Session ........................ 13
1.4.7 Registration Services ............................... 14
1.5 Protocol Properties ................................. 14
1.5.1 Minimal State ....................................... 14
1.5.2 Lower-Layer-Protocol Neutral ........................ 14
1.5.3 Text-Based .......................................... 15
2 SIP Uniform Resource Locators ....................... 15
3 SIP Message Overview ................................ 17
4 Request ............................................. 19
4.1 Request-Line ........................................ 20
4.2 Methods ............................................. 20
4.2.1 INVITE ............................................. 20
4.2.2 ACK ................................................ 21
4.2.3 OPTIONS ............................................ 21
4.2.4 BYE ................................................ 21
4.2.5 CANCEL ............................................. 22
4.2.6 REGISTER ........................................... 22
4.3 Request-URI ......................................... 24
4.3.1 SIP Version ......................................... 24
4.4 Option Tags ......................................... 24
4.4.1 Registering New Option Tags with IANA ............... 25
5 Response ............................................ 25
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6 Header Field Definitions ............................ 28
6.1 General Header Fields ............................... 29
6.2 Entity Header Fields ................................ 30
6.3 Request Header Fields ............................... 31
6.4 Response Header Fields .............................. 31
6.5 End-to-end and Hop-by-hop Headers ................... 31
6.6 Header Field Format ................................. 31
6.7 Accept ............................................. 32
6.8 Accept-Encoding .................................... 32
6.9 Accept-Language .................................... 32
6.10 Allow .............................................. 33
6.11 Authorization ...................................... 33
6.12 Call-ID ............................................ 33
6.13 Content-Encoding ................................... 34
6.14 Content-Length ..................................... 34
6.15 Content-Type ....................................... 35
6.16 CSeq ............................................... 35
6.17 Date ............................................... 36
6.18 Encryption ......................................... 36
6.19 Expires ............................................ 38
6.20 From ............................................... 39
6.21 Hide ............................................... 39
6.22 Location ........................................... 40
6.23 Max-Forwards ....................................... 42
6.24 Organization ....................................... 43
6.25 Priority ........................................... 43
6.26 Proxy-Authenticate ................................. 43
6.27 Proxy-Authorization ................................ 44
6.28 Proxy-Require ...................................... 44
6.29 Require ............................................ 44
6.30 Record-Route ....................................... 45
6.31 Response-Key ....................................... 46
6.32 Route .............................................. 46
6.33 Retry-After ........................................ 47
6.34 Server ............................................. 47
6.35 Subject ............................................ 47
6.36 Timestamp .......................................... 48
6.37 To ................................................. 48
6.38 Unsupported ........................................ 48
6.39 User-Agent ......................................... 49
6.40 Via ................................................ 49
6.40.1 Requests ............................................ 49
6.40.2 Receiver-tagged Via Fields ......................... 50
6.40.3 Responses ........................................... 50
6.40.4 Syntax .............................................. 51
6.41 Warning ............................................ 51
6.42 WWW-Authenticate ................................... 52
7 Status Code Definitions ............................. 53
7.1 Informational 1xx ................................... 53
7.1.1 100 Trying .......................................... 53
7.1.2 180 Ringing ......................................... 53
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7.2 Successful 2xx ...................................... 54
7.2.1 200 OK .............................................. 54
7.3 Redirection 3xx ..................................... 54
7.3.1 300 Multiple Choices ................................ 54
7.3.2 301 Moved Permanently ............................... 55
7.3.3 302 Moved Temporarily ............................... 55
7.3.4 380 Alternative Service ............................. 55
7.3.5 381 Ambiguous ....................................... 55
7.4 Request Failure 4xx ................................. 56
7.4.1 400 Bad Request ..................................... 56
7.4.2 401 Unauthorized .................................... 56
7.4.3 402 Payment Required ................................ 56
7.4.4 403 Forbidden ....................................... 56
7.4.5 404 Not Found ....................................... 56
7.4.6 405 Method Not Allowed .............................. 56
7.4.7 407 Proxy Authentication Required ................... 56
7.4.8 408 Request Timeout ................................. 57
7.4.9 420 Bad Extension ................................... 57
7.4.10 480 Temporarily Unavailable ......................... 57
7.4.11 481 Invalid Call-ID ................................. 57
7.4.12 482 Loop Detected ................................... 57
7.4.13 483 Too Many Hops ................................... 57
7.4.14 484 Address Incomplete .............................. 57
7.5 Server Failure 5xx .................................. 58
7.5.1 500 Server Internal Error ........................... 58
7.5.2 501 Not Implemented ................................. 58
7.5.3 502 Bad Gateway ..................................... 58
7.5.4 503 Service Unavailable ............................. 58
7.5.5 504 Gateway Timeout ................................. 58
7.5.6 505 Version Not Supported ........................... 58
7.6 Global Failures 6xx ................................. 59
7.6.1 600 Busy ............................................ 59
7.6.2 603 Decline ......................................... 59
7.6.3 604 Does Not Exist Anywhere ......................... 59
7.6.4 606 Not Acceptable .................................. 59
8 SIP Message Body .................................... 60
8.1 Body Inclusion ...................................... 60
8.2 Message Body Type ................................... 60
8.3 Message Body Length ................................. 60
9 Compact Form ........................................ 61
10 SIP Transport ....................................... 62
10.1 General Remarks ..................................... 62
10.1.1 Requests ............................................ 62
10.1.2 Responses ........................................... 62
10.2 Unicast UDP ......................................... 63
10.3 Multicast UDP ....................................... 63
10.4 BYE, CANCEL, OPTIONS ............................. 63
10.5 REGISTER ........................................... 64
10.6 ACK ................................................ 64
10.7 INVITE ............................................. 64
10.8 TCP Connections ..................................... 66
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11.2 User Agent Server ................................... 69
11.3 Stateless Proxy: Proxy Servers Issuing Single
Unicast Requests ............................................... 69
11.4 Proxy Server Issuing Several INVITE Requests ....... 69
11.5 Proxy Server Issuing Several ACK and BYE
Requests ....................................................... 73
12 Security Considerations ............................. 73
12.1 Confidentiality and Privacy: Encryption ............. 73
12.1.1 SIP Requests and Responses .......................... 73
12.2 End-to-End Encryption ............................... 74
12.2.1 Privacy of SIP Responses ............................ 76
12.2.2 Encryption by Proxies ............................... 76
12.2.3 Hop-by-Hop Encryption ............................... 77
12.2.4 Via field encryption ................................ 77
12.3 Message Integrity and Access Control:
Authentication ................................................. 77
12.3.1 Trusting responses .................................. 79
12.4 Callee Privacy ...................................... 80
12.5 Known Security Problems ............................. 80
13 SIP Security Using PGP .............................. 80
13.1 PGP Authentication Scheme ........................... 80
13.1.1 The WWW-Authenticate Response Header ............... 81
13.1.2 The Authorization Request Header ................... 81
13.2 PGP Encryption Scheme ............................... 82
13.3 Response-Key Header Field for PGP .................. 83
14 Examples ............................................ 83
14.1 Registration ........................................ 83
14.2 Invitation to Multicast Conference .................. 84
14.2.1 Request ............................................. 85
14.2.2 Response ............................................ 85
14.3 Two-party Call ...................................... 86
14.4 Terminating a Call .................................. 88
14.5 Forking Proxy ....................................... 88
14.6 Redirects ........................................... 92
14.7 Alternative Services ................................ 93
14.8 Negotiation ......................................... 94
14.9 OPTIONS Request .................................... 94
A Minimal Implementation .............................. 95
A.1 Client .............................................. 95
A.2 Server .............................................. 96
A.3 Header Processing ................................... 96
B Usage of SDP ........................................ 96
C Summary of Augmented BNF ............................ 96
D IANA Considerations ................................. 97
E Open Issues ......................................... 99
E.1 H.323 ............................................... 99
E.2 REGISTER ............................................ 99
E.3 Max-Forwards ........................................ 99
E.4 Cancellation and BYE ................................ 100
E.5 IPv6 URLs ........................................... 100
F Acknowledgments ..................................... 100
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