Internet Engineering Task Force SIP WG
Internet Draft J. Rosenberg
dynamicsoft
H. Schulzrinne
Columbia U.
draft-ietf-sip-guidelines-06.txt
November 4, 2002
Expires: May 2003
Guidelines for Authors of Extensions to
the Session Initiation Protocol (SIP)
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
The Session Initiation Protocol (SIP) is a flexible, yet simple tool
for establishing interactive connections across the Internet. Part of
this flexibility is the ease with which it can be extended. In order
to facilitate effective and interoperable extensions to SIP, some
guidelines need to be followed when developing SIP extensions. This
document outlines a set of such guidelines for authors of SIP
extensions.
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Table of Contents
1 Terminology ......................................... 3
2 Introduction ........................................ 3
3 Should I define a SIP Extension? ................... 3
3.1 SIP's Solution Space ................................ 4
3.2 SIP Architectural Model ............................. 6
4 Issues to be Addressed .............................. 8
4.1 Backwards Compatibility ............................. 8
4.2 Security ............................................ 10
4.3 Terminology ......................................... 10
4.4 Syntactic Issues .................................... 11
4.5 Semantics, Semantics, Semantics ..................... 13
4.6 Examples Section .................................... 14
4.7 Overview Section .................................... 14
4.8 IANA Considerations Section ......................... 15
4.9 Document Naming Conventions ......................... 15
4.10 Additional Considerations for New Methods ........... 16
4.11 Additional Considerations for New Header Fields or
Header Field Parameters ........................................ 17
4.12 Additional Considerations for New Body Types ........ 17
5 Interactions with SIP Features ...................... 18
6 Security Considerations ............................. 18
7 IANA Considerations ................................. 19
8 Acknowledgements .................................... 19
9 Authors Addresses ................................... 19
10 Normative References ................................ 19
11 Informative References .............................. 20
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1 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 [1] and
indicate requirement levels for compliant SIP implementations.
2 Introduction
The Session Initiation Protocol (SIP) [2] is a flexible, yet simple
tool for establishing interactive connections across the Internet.
Part of this flexibility is the ease with which it can be extended
(with new methods, new header fields, new body types, and new
parameters), and there have been countless proposals that have been
made to do just that. An IETF process has been put into place which
defines how extensions are to be made to the SIP protocol [9]. That
process is designed to ensure that extensions are made which are
appropriate for SIP (as opposed to being done in some other
protocol), that these extensions fit within the model and framework
provided by SIP and are consistent with its operation, and that these
extensions solve problems generically rather than for a specific use
case. However, [9] does not provide the technical guidelines needed
to assist that process. This specification helps to meet that need.
This specification first provides a set of guidelines to help decide
whether a certain piece of functionality is appropriately done in
SIP. Assuming the functionality is appropriate, it then points out
issues which extensions should deal with from within their
specification. Finally, it discusses common interactions with
existing SIP features which often cause difficulties in extensions.
3 Should I define a SIP Extension?
The first question to be addressed when defining a SIP extension is:
is a SIP extension the best solution to my problem? SIP has been
proposed as a solution for numerous problems, including mobility,
configuration and management, QoS control, call control, caller
preferences, device control, third party call control, and MPLS path
setup, to name a few. Clearly, not every problem can be solved by a
SIP extension. More importantly, some problems that could be solved
by a SIP extension, probably shouldn't.
To assist engineers in determining whether a SIP extension is an
appropriate solution to their problem, we present two broad criteria.
First, the problem SHOULD fit into the general purvey of SIP's
solution space. Secondly, the solution MUST conform to the general
SIP architectural model.
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While the first criteria might seem obvious, we have observed that
numerous extensions to SIP have been proposed because some function
is needed in a device which also speaks SIP. The argument is
generally given that "I'd rather implement one protocol than many".
As an example, user agents, like all other IP hosts, need some way to
obtain their IP address. This is generally done through DHCP [10].
SIP's multicast registration mechanisms might supply an alternate way
to obtain an IP address. This would eliminate the need for DHCP in
clients. However, we do not believe such extensions are appropriate.
We believe that protocols should be defined to provide specific,
narrow functions, rather than being defined based on all
communications requirements between a pair of devices. The latter
approach to protocol design yields modular protocols with broad
application. It also facilitates extensibility and growth; single
protocols can be removed and changed without affecting the entire
system. We observe that this approach to protocol engineering mirrors
object oriented software engineering.
Our second criteria, that the extension must conform to the general
SIP architectural model, ensures that the protocol remains manageable
and broadly applicable.
3.1 SIP's Solution Space
In order to evaluate the first criteria, it is necessary to define
exactly what SIP's solution space is, and what it is not.
SIP is a protocol for initiating, modifying, and terminating
interactive sessions. This process involves the discovery of users,
(or more generally, entities that can be communicated with, including
services, such as voicemail or translation devices) wherever they may
be located, so that a description of the session can be delivered to
the user. It is assumed that these users or communications entities
are mobile, and their point of attachment to the network changes over
time. The primary purpose of SIP is a rendezvous function, to allow a
request initiator to deliver a message to a recipient wherever they
may be. Such rendezvous is needed to establish a session, but can be
used for other purposes related to communications, such as querying
for capabilities or delivery of an instant message.
Much of SIP focuses on this discovery and rendezvous component. Its
ability to fork, its registration capabilities, and its routing
capabilities are all present for the singular purpose of finding the
desired user wherever they may be. As such, features and capabilities
such as personal mobility, automatic call distribution, and follow-me
are well within the SIP solution space.
Session initiation also depends on the ability of the called party to
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have enough information about the session itself in order to make a
decision on whether to join or not. That information includes data
about the caller, the purpose for the invitation, and parameters of
the session itself. For this reason, SIP includes this kind of
information.
Part of the process of session initiation is the communication of
progress and the final results of establishment of the session. SIP
provides this information as well.
SIP itself is independent of the session, and the session description
is delivered as an opaque body within SIP messages. Keeping SIP
independent of the sessions it initiates and terminates is
fundamental. As such, there are many functions that SIP explicitly
does not provide. It is not a session management protocol or a
conference control protocol. The particulars of the communications
within the session are outside of SIP. This includes features such as
media transport, voting and polling, virtual microphone passing,
chairman election, floor control, and feedback on session quality.
SIP is not a resource reservation protocol for sessions. This is
fundamentally because (1) SIP is independent of the underlying
session it establishes, and (2) the path of SIP messages is
completely independent from the path that session packets may take.
The path independence refers to paths within a providers network, and
the set of providers itself. For example, it is perfectly reasonable
for a SIP message to traverse a completely different set of
autonomous systems than the audio in a session SIP establishes.
SIP is not a general purpose transfer protocol. It is not meant to
send large amounts of data unrelated to SIP's operation. It is not
meant as a replacement for HTTP. This is not to say that carrying
payloads in SIP messages is never a good thing; in many cases, the
data is very much related to SIP's operation. In those cases,
congestion controlled transports end-to-end are critical.
SIP is not meant to be a general Remote Procedure Call (RPC)
mechanism. None of its user discovery and registration capabilities
are needed for RPC, neither are most of its proxy functions.
SIP is not meant to be used as a strict PSTN signaling replacement.
It is not a superset of ISUP. While it can support gatewaying of PSTN
signaling, and can provide many features present in the PSTN, the
mere existence of a feature or capability in the PSTN is not a
justification for its inclusion in SIP. Extensions needed to support
telephony MUST meet the other criteria described here.
SIP is a poor control protocol. It is not meant to be used for one
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entity to tell another to pick up or answer a phone, send audio using
a particular codec, or to provide a new value for a configuration
parameter. Control protocols have different trust relationships than
is assumed in SIP, and are more centralized in architecture than SIP,
which is a very distributed protocol.
There are many network layer services needed to make SIP function.
These include quality of service, mobility, and security, among
others. Rather than building these capabilities into SIP itself, they
SHOULD be developed outside of SIP, and then used by it.
Specifically, any protocol mechanisms that are needed by SIP, but are
also needed by many other application layer protocols, SHOULD NOT be
addressed within SIP.
3.2 SIP Architectural Model
We describe here some of the primary architectual assumptions which
underly SIP. Extensions which violate these assumptions should be
examined more carefully to determine their appropriateness for SIP.
Session independence: SIP is independent of the session it
establishes. This includes the type of session, be it
audio, video, game, chat session, or virtual reality. SIP
operation SHOULD NOT be dependent on some characteristic of
the session. SIP is not specific to VoIP only. Any
extensions to SIP MUST consider the application of SIP to a
variety of different session types.
SIP and Session Path Independence: We have already touched on
this once, but it is worth noting again. The set of routers
and/or networks and/or autonomous systems traversed by SIP
messages are unrelated to the set of routers and/or
networks and/or autonomous systems traversed by session
packets. They may be the same in some cases, but it is
fundamental to SIP's architecture that they need not be the
same. Extensions which only work under some assumption of
overlap are not generally applicable to SIP's operation and
should be scrutinized carefully.
Multi-provider and Multi-hop: SIP assumes that its messages will
traverse the Internet. That is, SIP works through multiple
networks administered by different providers. It is also
assumed that SIP messages traverse many hops (where each
hop is a proxy). Extensions SHOULD NOT work only under the
assumption of a single hop or single provider.
Transactional: SIP is a request/response protocol, possibly
enhanced with intermediate responses. Many of the rules of
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operation in SIP are based on general processing of
requests and responses. This includes the reliability
mechanisms, routing mechanisms, and state maintenance
rules. Extensions SHOULD NOT add messages that are not
within the request-response model.
Proxies can ignore bodies: In order for proxies to scale well,
they must be able to operate with minimal message
processing. SIP has been engineered so that proxies can
always ignore bodies. Extensions SHOULD NOT require proxies
to examine bodies.
Proxies don't need to understand the method: Processing of
requests in proxies does not depend on the method, except
for the well known methods INVITE, ACK, and CANCEL. This
allows for extensibility. Extensions MUST NOT define new
methods which must be understood by proxies.
INVITE messages carry full state: An initial INVITE message for
a session is nearly identical (the exception is the tag) to
a re-INVITE message to modify some characteristic of the
session. This full state property is fundamental to SIP,
and is critical for robustness of SIP systems. Extensions
SHOULD NOT modify INVITE processing such that data spanning
multiple INVITEs must be collected in order to perform some
feature.
Generality over efficiency: Wherever possible, SIP has favored
general purpose components rather than narrow ones. If some
capability is added to support one service, but a slightly
broader capability can support a larger variety of services
(at the cost of complexity or message sizes), the broader
capability SHOULD be preferred.
The Request URI is the primary key for forwarding: Forwarding
logic at SIP servers depends primarily on the request URI
(this is different from request routing in SIP, which uses
the Route header fields to pass a request through
intermediate proxies). It is fundamental to the operation
of SIP that the request URI indicate a resource that, under
normal operations, resolves to the desired recipient.
Extensions SHOULD NOT use other components of the SIP
message as the primary forwarding key, and SHOULD NOT
modify the semantics of the request URI.
Heterogeneity is the norm: SIP supports hetereogeneous devices.
It has built in mechanisms for determining the set of
overlapping protocol functionalities. Extensions SHOULD NOT
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be defined which only function if all devices support the
extension.
4 Issues to be Addressed
Given an extension has met the litmus tests in the previous section,
there are several issues that all extensions should take into
consideration.
4.1 Backwards Compatibility
One of the most important issues to consider is whether the new
extension is backwards compatible with baseline SIP. This is tightly
coupled with how the Require, Proxy-Require, and Supported header
fields are used.
If an extension consists of new header fields or header field
parameters inserted by a user agent in a request with an existing
method, and the request cannot be processed reasonably by a proxy
and/or user agent without understanding the header fields or
parameters, the extension MUST mandate the usage of the Require
and/or Proxy-Require header fields in the request. These extensions
are not backwards compatible with SIP. The result of mandating usage
of these header fields means that requests cannot be serviced unless
the entities being communicated with also understand the extension.
If some entity does not understand the extension, the request will be
rejected. The UAC can then handle this in one of two ways. In the
first, the request simply fails, and the service cannot be provided.
This is basically an interoperability failure. In the second case,
the UAC retries the request without the extension. This will preserve
interoperability, at the cost of a "dual stack" implementation in a
UAC (processing rules for operation with and without the extension).
As the number of extensions increases, this leads to an exponential
explosion in the sets of processing rules a UAC may need to
implement. The result is excessive complexity.
Because of the possibility of interoperability and complexity
problems that result from the usage of Require and Proxy-Require, we
believe the following guidelines are appropriate:
o The usage of these header fields in requests for basic SIP
services (in particular, session initiation and termination)
is NOT RECOMMENDED. The less frequently a particular extension
is needed in a request, the more reasonable it is to use these
header fields.
o The Proxy-Require header field SHOULD be avoided at all costs.
The failure likelihood in an individual proxy stays constant,
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but the path failure grows exponentially with the number of
hops. On the other hand, the Require header field only
mandates that a single entity, the UAS, support the extension.
Usage of Proxy-Require is thus considered exponentially worse
than usage of the Require header field.
o If either Require or Proxy-Require are used by an extension,
the extension SHOULD discuss how to fall back to baseline SIP
operation if the request is rejected with a 420 response.
Extensions which define new methods do not need to use the Require
header field. SIP defines mechanisms which allow a UAC to know
whether a new method is understood by a UAS. This includes both the
OPTIONS request, and the 405 (Method Not Allowed) response with the
Allow header field. It is fundamental to SIP that proxies do not need
to understand the semantics of a new method in order to process it.
If an extension defines a new method which must be understood by
proxies in order to be processed, a Proxy-Require header field is
needed. As discussed above, these kinds of extensions are frowned
upon.
In order to achieve backwards compatibility for extensions that
define new methods, the Allow header field is used. There are two
types of new methods - those that are used for established dialogs
(initiated by INVITE, for example), and those that are sent as the
initial request to a UA. Since INVITE and its response both SHOULD
contain an Allow header field, a UA can readily determine whether the
new method can be supported within the dialog. For example, once an
INVITE dialog is established, a user agent could determine if the
REFER method [11] is supported if it is present in an Allow header.
If it was, the "transfer" button on the UI could be "greyed out" once
the call is established.
Another type of extension are those which require a proxy to insert
header fields or header field parameters into a request as it
traverses the network, or for the UAS to insert header fields or
header field parameters into a response. For some extensions, if the
UAC or UAS does not understand these header fields, the message can
still be processed correctly. These extensions are completely
backwards compatible.
Most other extensions of this type require that the server only
insert the header field or parameter if it is sure the client
understands it. In this case, these extensions will need to make use
of the Supported request header field mechanism. This mechanism
allows a server to determine if the client can understand some
extension, so that it can apply the extension to the response. By
their nature, these extensions may not always be able to be applied
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to every response.
If an extension requires a proxy to insert a header field or
parameter into a request, and this header field or parameter needs to
be understood by both UAC and UAS to be executed correctly, a
combination of the Require and the Supported mechanism will need to
be used. The proxy can insert a Require header field into the
request, given the Supported header field is present. An example of
such an extension is the SIP Session Timer [12].
Yet another type of extension is that which defines new body types to
be carried in SIP messages. According to the SIP specification,
bodies must be understood in order to process a request. As such, the
interoperability issues are similar to new methods. However, the
Content-Disposition header field has been defined to allow a client
or server to indicate that the message body is optional [2]. Usage of
optional bodies, as opposed to mandatory ones, is RECOMMENDED
wherever possible.
When a body must be understood to properly process a request or
response, it is preferred that the sending entity know ahead of time
whether the new body is understood by the recipient. For requests
that establish a dialog, inclusion of Accept in the request and its
success responses is RECOMMENDED. This will allow both parties to
determine what body types are supported by their peers. Subsequent
messaging between the peers would then only include body types that
were indicated as being understood.
4.2 Security
Security is an important component of any protocol. Designers of SIP
extensions need to carefully consider if additional security
requirements are required over those described in RFC3261. Frequently
authorization requirements, and requirements for end-to-end integrity
are the most overlooked.
SIP extensions MUST consider how (or if) they affect usage of the
general SIP security mechanisms. Most extensions should not require
any new security capabilities beyond general purpose SIP. If they do,
it is likely that the security mechanism has more general purpose
application, and should be considered an extension in its own right.
4.3 Terminology
RFC 3261 has an extensive terminology section that defines terms like
caller, callee, user agent, header field, and so on. All SIP
extensions MUST conform to this terminology. They MUST NOT define new
terms that describe concepts already defined by a term in another SIP
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specification. If new terminology is needed, it SHOULD appear in a
separate section towards the beginning of the document.
Careful attention must be paid to the actual usage of terminology.
Many documents misuse the terms header, header field, and header
field values, for example. Document authors SHOULD do a careful
review of their documents for proper usage of these terms.
4.4 Syntactic Issues
Extensions that define new methods SHOULD use all capitals for the
method name. Method names SHOULD be less than 10 characters, and
SHOULD attempt to convey the general meaning of the request.
Method names are case sensitive, and therefore there is no
requirement that they be capitalized. However, using
capitalized method names keeps with a long-standing
convention in SIP and many similar protocols, such as HTTP
[13] and RTSP [14].
Extensions that define new header fields that are anticipated to be
heavily used SHOULD define a compact form if those header fields are
more than four characters. Compact header fields MUST be a single
character. When all 26 characters are exhausted, new compact forms
will no longer be defined. Header field names SHOULD be composed
primarily of ASCII characters and marks. They SHOULD be descriptive
but reasonably brief. Although header field names are case
insensitive, a single common capitalization SHOULD be used throughout
the document. It is RECOMMENDED that this be camel case.
As an example, the following are poor choices for header field names:
ThisIsMyNewHeaderThatDoesntDoVeryMuchButItHasANiceName
--.!A
Function
Case sensitivity of parameters and values is a constant source of
confusion, a difficulty that plagued RFC 2543 [15]. This has been
made simple through the usage of the BNF constructs of RFC 2234 [3],
which have clear rules of case sensivitity and insensitivity.
Therefore, the BNF for an extension completely defines the matching
rules.
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Extensions MUST be consistent with the SIP conventions for
sensitivity. Methods MUST be case sensitive. Header field names MUST
be case insensitive. Header field parameter names MUST be case
insensitive. Header field values and parameter values are sometimes
case sensitive, and sometimes case insensitive. However, generally
they SHOULD be case insensitive. Definiting a case sensitive
component requires explicitly listing each character through its
ASCII code.
Extensions which contain freeform text MUST allow that text to be
UTF-8, as per the IETF policies on character set usage [4]. This
ensures that SIP remains an internationalized standard. As a general
guideline, freeform text is never needed by programs in order to
perform protocol processing. It is usually entered by and displayed
to the user. If an extension uses a parameter which can contain UTF-8
encoded characters, and that extension requires a comparison to be
made of this parameter to other parameters, the comparison MUST be
case sensitive. Case insensitive comparison rules for UTF-8 text are,
at this time, impossible and MUST be avoided.
Extensions which make use of dates MUST use the SIP-Date BNF defined
in RFC 3261 [2]. No other date formats are allowed. However, the
usage of absolute dates in order to determine intervals (for example,
the time at which some timer fires) is NOT RECOMMENDED. This is
because it requires synchronized time between peers, and this is
frequently not the case. Therefore, relative times, expressed in
numbers of seconds, SHOULD be used.
Extensions which include network layer addresses SHOULD permit dotted
quad IPv4 addresses, IPv6 addresses in the format described in [5],
and domain names.
Extensions which have header fields containing URIs SHOULD allow any
URI, not just SIP URIs.
Header fields MUST follow the standard formatting for SIP, defined
as:
header = header-name HCOLON header-value
*(COMMA header-value)
header-name = token
header-value = value *(SEMI value-parameter)
value-parameter = token [EQUAL gen-value]
gen-value = token / host / quoted-string
value = token / host / quoted-string
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In some cases, this form is not sufficient. That is the case for
header fields that express descriptive text meant for human
consumption. An example is the Subject header field in SIP [2]. In
this case, an alternate form is:
header = header-name HCOLON [TEXT-UTF8-TRIM]
Developers of extensions SHOULD allow for extension parameters in
their header fields.
Header Fields that contain a list of URIs SHOULD follow the same
syntax as the Contact header field in SIP. Implementors are also
encouraged to always wrap these URI in angle brackets "<" and ">". We
have found this to be a frequently misimplemented feature.
Beyond compact form, there is no need to define compressed versions
of header field values. Compression of SIP messages SHOULD be handled
at lower layers, for example, using IP payload compression [16] or
signalling compression [17].
Syntax for header fields is expressed in Augmented Backus-Naur Form
and MUST follow the format of RFC 2234 [3]. Extensions MUST make use
of the primitive components defined in RFC 3261 [2]. If the
construction for a BNF element is defined in another specification,
it is RECOMMENDED that the construction be referenced rather than
copied. The reference SHOULD include both the document and section
number. All BNF elements must be either defined or referenced.
It is RECOMMENDED that BNF be collected into a single section near
the end of the document.
All tokens and quoted strings are separated by explicit linear white
space. Linear white space, for better or worse, allows for line
folding. Extensions MUST NOT define new header fields that use
alternate linear white space rules.
All SIP extensions MUST verify that any BNF productions that they
define in their grammar do not conflict with any existing grammar
defined in other SIP standards track specifications.
4.5 Semantics, Semantics, Semantics
Developers of protocols often get caught up in syntax issues, without
spending enough time on semantics. The semantics of a protocol are
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far more important. SIP extensions MUST clearly define the semantics
of the extensions. Specifically, the extension MUST specify the
behaviors expected of a UAC, UAS and proxy in processing the
extension. This is often best described by having separate sections
for each of these three elements. Each section SHOULD step through
the processing rules in temporal order of the most common messaging
scenario.
Processing rules generally specify actions to take (in terms of
messages to send, variables to store, rules to follow) on receipt of
messages and expiration of timers. If an action requires transmission
of a message, the rule SHOULD outline requirements for insertion of
header fields or other information in the message.
The extension SHOULD specify procedures to take in exceptional
conditions which are recoverable, or which require some kind of user
intervention. Recovering from unrecoverable problems generally does
not require specification.
4.6 Examples Section
The specification SHOULD contain a section that gives examples of
call flows and message formatting. Extensions which define
substantial new syntax SHOULD include examples of messages containing
that syntax. Examples of message flows should be given to cover
common cases and at least one failure or unusual case.
For an example of how to construct a good examples section, see the
message flows and message formatting defined in the Basic Call Flows
specification [18]. Note that complete messages SHOULD be used. Be
careful to include tags, Via header fields (with the branch ID
cookie), Max-Forwards, Content-Lengths, Record-Route and Route header
fields. Example INVITE messages MAY omit session descriptions, and
Content-Length values MAY be set to "..." to indicate that the value
is not provided. However, the specification MUST explicitly call out
the meanining of the "..." and explicitly indicate that session
descriptions were not included.
4.7 Overview Section
Too often, extension documents dive into detailed syntax and
semantics without giving a general overview of operation. This makes
understanding of the extension harder. It is RECOMMENDED that
extensions have a protocol overview section which discusses the basic
operation of the extension. Basic operation usually consists of the
message flow, in temporal order, for the most common case covered by
the extension. The most important processing rules for the elements
in the call flow SHOULD be mentioned. Usage of the RFC 2119 [1]
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terminology in the overview section is NOT RECOMMENDED, and the
specification should explicitly state that the overview is tutorial
in nature only.
4.8 IANA Considerations Section
Documents which define new SIP extensions will invariably have IANA
Considerations sections.
If your extension is defining a new event package, you MUST register
that package. RFC 3265 [6] provides the registration template. See
[19] for an example of the registration of a new event package.
If your extension is defining a new header field, you MUST register
that header field. RFC 3261 [2] provides a registration template. See
Section 8.2 of RFC 3262 [20] for an example of how to register new
SIP header fields.
If your extension is defining a new response code, you MUST register
that response code. RFC 3261 [2] provides a registration template.
See [12] for an example of how to register a new response code.
If your extension is defining a new SIP method, you MUST register
that method. RFC 3261 [2] provides a registration template. See
Section 10 of RFC 3311 [21] for an example of how to register a new
SIP method.
Many SIP extensions make use of option tags, carried in the Require,
Proxy-Require and Supported header fields. Section 4.1 discusses some
of the issues involved in the usage of these header fields. If your
extension does require them, you MUST register an option tag for your
extension. RFC 3261 [2] provides a registration template. See Section
8.1 of RFC 3262 [20] for an example of how to register an option tag.
Some SIP extensions will require establishment of their own IANA
registries. RFC 2434 [7] provides guidance on how and when IANA
registries are established. For an example of how to set one up, see
Section 6 of RFC 3265 [6] for an example.
4.9 Document Naming Conventions
An important decision to be made about the extension is its title.
The title MUST indicate that the document is an extension to SIP. It
is RECOMMENDED that the title follow the basic form of "A [summary of
function] for the Session Initiation Protocol (SIP)", where the
summary of function is a one to three word description of the
extension. For example, if an extension defines a new header field,
called Make-Coffee, for making coffee, the title would read, "Making
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Coffee with the Session Initiation Protocol (SIP)". It is RECOMMENED
that these additional words be descriptive rather than naming the
header field. For example, the extension for making coffee should not
be named "The Make-Coffee Header for the Session Initiation
Protocol".
For extensions that define new methods, an acceptable template for
titles is "The Session Initiation Protocol (SIP) X Method" where X is
the name of the method.
Note that the acronymn SIP MUST be expanded in the titles of RFCs, as
per [22].
4.10 Additional Considerations for New Methods
Extensions which define new methods SHOULD take into consideration,
and discuss, the following issues:
o Can it contain bodies? If so, what is the meaning of the
presence of those bodies? What body types are allowed?
o Can a transaction with this request method occur while another
transaction, in the same and/or reverse direction, is in
progress?
o The extension MUST define which header fields can be present
in requests of that method. It is RECOMMENDED that this
information be represented as a new column of Table 2/3 of RFC
3261 [2]. The table MUST contain rows for all header fields
defined in standards track RFCs at the time of writing of the
extension.
o Can the request be sent within a dialog, or does it establish
a dialog?
o Is it a target refresh request?
o Extensions to SIP that define new methods MAY specify whether
offers and answers can appear in requests of that method or
its responses. However, those extensions MUST adhere to the
protocol rules specified in [8], and MUST adhere to the
additional constraints for offers and answers as specified in
SIP [2].
o Because of the nature of reliability treatment of requests
with new methods, those requests need to be answered
immediately by the UAS. Protocol extensions that require
longer durations for the generation of a response (such as a
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new method that requires human interaction) SHOULD instead use
two transactions - one to send the request, and another in the
reverse direction to convey the result of the request. An
example of that is SUBSCRIBE and NOTIFY [6].
o The CANCEL request can be used for a particular extension
method on a method-by-method basis. SIP [2] only allows
cancellation of INVITE. Extensions that define new methods
MUST state whether or not transactions initiated by requests
with that method can be cancelled. Furthermore, the rules a
UAS should follow upon cancellation of an unanswered request
MUST be described. Note that, since non-INVITE requests are
generally answered immediately, cancellation ususally serves
no purpose.
Note that the reliability mechanisms for all new methods must be the
same as for BYE. The delayed response feature of INVITE is only
available in INVITE, never for new methods. This means requests with
new SIP methods need to be responded to within short time periods (on
the order of seconds).
4.11 Additional Considerations for New Header Fields or Header Field
Parameters
The most important issue for extensions that define new header fields
or header field parameters is backwards compatibility. See Section
4.1 for a discussion of the issues. The extension MUST detail how
backwards compatibility is addressed.
It is often tempting to avoid creation of a new method by overloading
an existing method through a header field or parameter. Header fields
and parameters are not meant to fundamentally alter the meaning of
the method of the request. A new header field cannot change the basic
semantic and processing rules of a method. There is no shortage of
method names, so when an extension changes the basic meaning of a
request, a new method SHOULD be defined.
For extensions that define new header fields, the extension MUST
define the request methods the header field can appear in, and what
responses it can be used in. It is RECOMMENDED that this information
be represented as a new row of Table 2/3 of RFC 3261 [2]. The table
MUST contain columns for all methods defined in standards track RFCs
at time of writing of the extension.
4.12 Additional Considerations for New Body Types
Because SIP can run over UDP, extensions that specify the inclusion
of large bodies are frowned upon unless end-to-end congestion
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controlled transport can be guaranteed. If at all possible, the
content SHOULD be included indirectly [23] even if congestion
controlled transports are available.
Note that the presence of a body MUST NOT change the nature of the
message. That is, bodies cannot alter the state machinery associated
with processing a request of a particular method or a response.
Bodies enhance this processing by providing additional data.
5 Interactions with SIP Features
We have observed that certain capabilities of SIP continually
interact with extensions in unusual ways. Writers of extensions
SHOULD consider the interactions of their extensions with these SIP
capabilities, document any unusual interactions if they exist. The
most common causes of problems are:
Forking: Forking by far presents the most troublesome
interactions with extensions. This is generally because it
can cause (1) a single transmitted request to be received
by an unknown number of UASs, and (2) a single INVITE
request to have multiple responses.
CANCEL and ACK: CANCEL and ACK are "special" SIP requests, in
that they are exceptions to many of the general request
processing rules. The main reason for this special status
is that CANCEL and ACK are always associated with another
request. New methods SHOULD consider the meaning of
cancellation, as described above. Extensions which defined
new header fields in INVITE requests SHOULD consider
whether they also need to be included in ACK and CANCEL.
Frequently they do, in order to allow a stateless proxy to
route the CANCEL or ACK identically to the INVITE.
Routing: The presence of Route header fields in a request can
cause it to be sent through intermediate proxies. Requests
that establish dialogs can be record-routed, so that the
initial request goes through one set of proxies, and
subsequent requests through a different set. These SIP
features can interact in unusual ways with extensions.
Stateless Proxies: SIP allows a proxy to be stateless. Stateless
proxies are unable to retransmit messages and cannot
execute certain services. Extensions which depend on some
kind of proxy processing SHOULD consider how stateless
proxies affect that processing.
6 Security Considerations
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Internet Draft SIP Guidelines November 4, 2002
The nature of this document is such that it does not introduce any
new security considerations.
7 IANA Considerations
There are no IANA considerations associated with this specification.
8 Acknowledgements
The authors would like to thank Rohan Mahy for his comments.
9 Authors Addresses
Jonathan Rosenberg
dynamicsoft
72 Eagle Rock Avenue
East Hanover, NJ 07936
email: jdrosen@dynamicsoft.com
Henning Schulzrinne
Columbia University
M/S 0401
1214 Amsterdam Ave.
New York, NY 10027-7003
email: schulzrinne@cs.columbia.edu
10 Normative References
[1] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[2] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J.
Peterson, R. Sparks, M. Handley, and E. Schooler, "SIP: session
initiation protocol," RFC 3261, Internet Engineering Task Force, June
2002.
[3] "Augmented BNF for syntax specifications: ABNF," RFC 2234,
Internet Engineering Task Force, Nov. 1997.
[4] H. Alvestrand, "IETF policy on character sets and languages," RFC
2277, Internet Engineering Task Force, Jan. 1998.
[5] R. Hinden, B. Carpenter, and L. Masinter, "Format for literal
IPv6 addresses in URL's," RFC 2732, Internet Engineering Task Force,
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Internet Draft SIP Guidelines November 4, 2002
Dec. 1999.
[6] A. B. Roach, "Session initiation protocol (sip)-specific event
notification," RFC 3265, Internet Engineering Task Force, June 2002.
[7] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
considerations section in RFCs," RFC 2434, Internet Engineering Task
Force, Oct. 1998.
[8] J. Rosenberg and H. Schulzrinne, "An offer/answer model with
session description protocol (SDP)," RFC 3264, Internet Engineering
Task Force, June 2002.
11 Informative References
[9] A. Mankin, S. Bradner, and R. Mahy, "Change process for the
session initiation protocol (SIP)," Internet Draft, Internet
Engineering Task Force, Aug. 2002. Work in progress.
[10] R. Droms, "Dynamic host configuration protocol," RFC 2131,
Internet Engineering Task Force, Mar. 1997.
[11] R. Sparks, "The SIP refer method," Internet Draft, Internet
Engineering Task Force, July 2002. Work in progress.
[12] S. Donovan and J. Rosenberg, "Session initiation protocol
extension for session timer," Internet Draft, Internet Engineering
Task Force, July 2002. Work in progress.
[13] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, L. Masinter, P.
Leach, and T. Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1,"
RFC 2616, Internet Engineering Task Force, June 1999.
[14] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
1998.
[15] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
session initiation protocol," RFC 2543, Internet Engineering Task
Force, Mar. 1999.
[16] A. Shacham, R. Monsour, R. Pereira, and M. Thomas, "IP payload
compression protocol (ipcomp)," RFC 2393, Internet Engineering Task
Force, Dec. 1998.
[17] R. Price et al. , "Signaling compression," Internet Draft,
Internet Engineering Task Force, June 2002. Work in progress.
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Internet Draft SIP Guidelines November 4, 2002
[18] A. Johnston et al. , "Session initiation protocol basic call
flow examples," Internet Draft, Internet Engineering Task Force, Aug.
2002. Work in progress.
[19] J. Rosenberg, "A sip event package for registration state,"
Internet Draft, Internet Engineering Task Force, Oct. 2002. Work in
progress.
[20] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
responses in session initiation protocol (SIP)," RFC 3262, Internet
Engineering Task Force, June 2002.
[21] J. Rosenberg, "The session initiation protocol (SIP) UPDATE
method," RFC 3311, Internet Engineering Task Force, Oct. 2002.
[22] J. Reynolds and B. Braden, "Instructions to request for comments
(RFC) authors," Internet Draft, Internet Engineering Task Force, Apr.
2002. Work in progress.
[23] S. Olson, "Requirements for content indirection in SIP
messages," Internet Draft, Internet Engineering Task Force, July
2002. Work in progress.
Full Copyright Statement
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or assist in its implementation may be prepared, copied, published
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TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
J. Rosenberg et. al. [Page 21]
Internet Draft SIP Guidelines November 4, 2002
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
J. Rosenberg et. al. [Page 22]