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Keywords: interactive communication
Network Working Group J. Rosenberg
Request for Comments: 4485 Cisco Systems
Category: Informational H. Schulzrinne
Columbia University
May 2006
Guidelines for Authors of Extensions to
the Session Initiation Protocol (SIP)
Status of This Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
The Session Initiation Protocol (SIP) is a flexible yet simple tool
for establishing interactive communications sessions 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. Introduction ....................................................2
2. Terminology .....................................................3
3. Should I Define a SIP Extension? ................................3
3.1. SIP's Solution Space .......................................4
3.2. SIP Architectural Model ....................................5
4. Issues to Be Addressed ..........................................7
4.1. Backwards Compatibility ....................................7
4.2. Security ..................................................10
4.3. Terminology ...............................................10
4.4. Syntactic Issues ..........................................10
4.5. Semantics, Semantics, Semantics ...........................13
4.6. Examples Section ..........................................14
4.7. Overview Section ..........................................14
4.8. IANA Considerations Section ...............................14
4.9. Document-Naming Conventions ...............................16
4.10. Additional Considerations for New Methods ................16
4.11. Additional Considerations for New Header Fields
or Header Field ..........................................17
4.12. Additional Considerations for New Body Types .............18
5. Interactions with SIP Features .................................18
6. Security Considerations ........................................19
7. Acknowledgements ...............................................19
8. References .....................................................19
8.1. Normative References ......................................19
8.2. Informative References ....................................20
1. Introduction
The Session Initiation Protocol (SIP) [2] is a flexible yet simple
tool for establishing interactive communications sessions 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
that defines how extensions are to be made to the SIP protocol [10].
That process is designed to ensure that extensions are made that 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, [10] 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
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issues that extensions should deal with from within their
specification. Finally, it discusses common interactions with
existing SIP features that often cause difficulties in extensions.
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 [1] and
indicate requirement levels for compliant implementations.
3. Should I Define a SIP Extension?
The first question to be addressed when defining a SIP extension is
whether a SIP extension is the best solution to the 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 purview of SIP's
solution space. Secondly, the solution MUST conform to the general
SIP architectural model.
Although 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 that 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 [11].
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 be defined for all protocols needed
between a pair of devices. The former 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.
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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 that 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 a rendezvous is
needed to establish a session, but it 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
have enough information about the session itself to make a decision
on whether to join. 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
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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 provider's 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, and neither are most of its proxy functions.
SIP is not meant to be used as a strict Public Switched Telephone
Network (PSTN) signaling replacement. It is not a superset of the
Integrated Services Digital Network (ISDN) User Part (ISUP).
Although 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
entity to tell another to pick up or answer a phone, to send audio
using a particular codec, or to provide a new value for a
configuration parameter. Control protocols have different trust
relationships from that assumed in SIP and are more centralized in
architecture than SIP is, as SIP 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 build 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
that 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 architectural assumptions that
underlie SIP. Extensions that violate these assumptions should be
examined more carefully to determine their appropriateness for SIP.
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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 depend on some characteristic of the session. SIP is
not specific to voice 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, networks,
and/or autonomous systems traversed by SIP messages are unrelated
to the set of routers, 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. Standards-track extensions MUST NOT be defined that
work only when the signaling and session paths are coupled. Non-
standard P-header extensions [10] are required for any extension
that only works in such a case.
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 MUST NOT work only under the assumption of a single hop
or specialized network topology. They SHOULD avoid the assumption
of a single SIP provider (but see the use of P-Headers, per RFC
3427 [10]).
Transactional: SIP is a request/response protocol, possibly enhanced
with intermediate responses. Many of the rules of 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 that must
be understood by proxies.
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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 modify the semantics of the
request URI.
Heterogeneity is the norm: SIP supports heterogeneous devices. It
has built-in mechanisms for determining the set of overlapping
protocol functionalities. Extensions SHOULD NOT be defined that
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. Backward Compatibility
One of the most important issues to consider is whether the new
extension is backward 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
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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, 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 that define new methods do not need to use the Require
header field. SIP defines mechanisms that 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 need not
understand the semantics of a new method in order to process it. If
an extension defines a new method that 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
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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 whether
the REFER method [12] is supported if it is present in an Allow
header field. If it wasn't, the "transfer" button on the UI could be
"greyed out" once the call is established.
Another type of extension is that which requires 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
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 if the Supported header field is present. An example of such
an extension is the SIP Session Timer [13].
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 by user agents 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]. Extensions that define or require new body types
SHOULD make them optional for the user agent to process.
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
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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 RFC 3261.
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.
Overall system security requires that both the SIP signaling and the
media sessions it established be secured. The media sessions
normally use their own security techniques, which are quite distinct
from those used by SIP itself. Extensions should take care not to
conflate the two. However, specifications that define extensions
that impact the media sessions in any way SHOULD consider the
interactions between SIP and session security mechanisms.
4.3. Terminology
RFC 3261 has an extensive terminology section that defines terms such
as caller, callee, user agent, and header field. 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
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 shorter than 10 characters and
SHOULD attempt to convey the general meaning of the request. Method
names are case sensitive, and therefore, strictly speaking, they
don't have to be capitalized. However, using capitalized method
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names keeps with a long-standing convention in SIP and many similar
protocols, such as HTTP [15] and RTSP [16].
Extensions that define new header fields that are anticipated to be
heavily used MAY define a compact form if those header fields are
more than six characters. "Heavily used" means that the percentage
of all emitted messages that contain that header field is over thirty
percent. Usage of compact forms in these cases is only a MAY because
there are better approaches for reducing message overhead [20].
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 are defined by the "token" production in
RFC 3261, Section 25.1, and thus include the upper and lowercase
letters, the digits 0 through 9, the HYPHEN-MINUS (-), FULL STOP (.),
EXCLAMATION MARK (!), PERCENT SIGN (%), ASTERISK (*), LOW LINE (_),
PLUS SIGN (+), GRAVE ACCENT (`), APOSTROPHE ('), and TILDE (~). 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 each English
word present in the header field name have its first letter
capitalized. For example, "ThisIsANewHeader".
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 [17]. This has been
simplified through the usage of the BNF constructs of RFC 4234 [5],
which have clear rules of case sensitivity and insensitivity.
Therefore, the BNF for an extension completely defines the matching
rules.
Extensions MUST be consistent with the SIP conventions for case
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. Defining a case-sensitive component
requires explicitly listing each character through its ASCII code.
Extensions that contain freeform text MUST allow that text to be
UTF-8, as per the IETF policies on character set usage [3]. This
ensures that SIP remains an internationalized standard. As a general
guideline, freeform text is never needed by programs to perform
protocol processing. It is usually entered by and displayed to the
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user. If an extension uses a parameter that 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 that make use of dates MUST use the SIP-Date BNF defined
in RFC 3261. No other date formats are allowed. However, the usage
of absolute dates 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 that include network-layer addresses SHOULD permit dotted
quad IPv4 addresses, IPv6 addresses in the format described in [4],
and domain names.
Extensions that have header fields containing URIs SHOULD be explicit
about which URI schemes can be used in that header field. Header
fields SHOULD allow the broadest set of URI schemes possible that are
a match for the semantics of the header field.
Header fields MUST follow the standard formatting for SIP, defined as
follows:
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
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.
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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 wrap these URI in angle brackets, "<" and ">", at all
times. We have found this to be a frequently misimplemented feature.
Beyond the 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
[18] or signalling compression [20].
Syntax for header fields is expressed in Augmented Backus-Naur Form
and MUST follow the format of RFC 4234 [5]. 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
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 be taken (in terms of
messages to be sent, variables to be stored, and rules to be
followed) 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.
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The extension SHOULD specify procedures to be taken in exceptional
conditions that are recoverable, or that require some kind of user
intervention. Handling of unrecoverable errors does not require
specification.
4.6. Examples Section
The specification SHOULD contain a section that gives examples of
call flows and message formatting. Extensions that 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 [21]. 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 meaning 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 that 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]
terminology in the overview section is NOT RECOMMENDED, and the
specification should explicitly state that the overview is tutorial
in nature only. This section SHOULD expand all acronyms, even those
common in SIP systems, and SHOULD be understandable to readers who
are not SIP experts. [27] provides additional guidance on writing
good overview sections.
4.8. IANA Considerations Section
Documents that 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
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[22] for an example of the registration of a new event package. As
discussed in RFC 3427 [10], only standards-track documents can
register new event-template packages. Both standards-track and
informational specifications can register event packages.
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 [23] for an example of how to register
new SIP header fields. Both standards-track and informational
P-header specifications can register new header fields [10].
If your extension is defining a new response code, you MUST register
that response code. RFC 3261 [2] provides a registration template.
See Section 6.4 of RFC 3329 [19] for an example of how to register a
new response code. As discussed in RFC 3427 [10], only standards-
track documents can register new response codes.
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 [24] for an example of how to register a new
SIP method. As discussed in RFC 3427 [10], only standards-track
documents can register new methods.
If your extension is defining a new SIP header field parameter, you
MUST register that header field parameter per the guidelines in RFC
3968 [7]. Section 4.1 of that specification provides a template.
Only IETF approved specifications can register new header field
parameters. However, there is no requirement that these be standards
track.
If your extension is defining a new SIP URI parameter, you MUST
register that URI parameter per the guidelines in RFC 3969 [8].
Section 4.1 of that specification provides a template. Only
standards-track documents can register new URI parameters.
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 [23] for an example of how to register an
option tag. Only standards-track RFCs can register new option tags.
Some SIP extensions will require establishment of their own IANA
registries. RFC 2434 [25] 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.
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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
Coffee with the Session Initiation Protocol (SIP)". It is
RECOMMENDED 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 acronym SIP MUST be expanded in the titles of RFCs, as
per [26].
4.10. Additional Considerations for New Methods
Extensions that 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
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rules specified in [28] 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 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 SIP specification [2] allows new methods to specify whether
transactions using that new method can be canceled using a CANCEL
request. Further study of the non-INVITE transaction [14] has
determined that non-INVITE transactions must be completed as soon
as possible. New methods must not plan for the transaction to
pend long enough for CANCEL to be meaningful. Thus, new methods
MUST declare that transactions initiated by requests with that
method cannot be canceled. Future work may relax this
restriction, at which point these guidelines will be revised.
o New methods that establish a new dialog must discuss the impacts
of forking. The design of such new methods should follow the
pattern of requiring an immediate request in the reverse direction
from the request establishing a dialog, similar to the immediate
NOTIFY sent when a subscription is created per RFC 3265 [6].
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. The design of new methods must
encourage an immediate response. If the application being enabled
requires a delay, the design SHOULD follow a pattern using multiple
transactions, similar to RFC 3265's use of NOTIFYs with different
Subscription-State header field values (pending and active in
particular) in response to SUBSCRIBE [6].
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
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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 the 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 (where large is several times the ethernet MTU) are
frowned upon unless end-to-end congestion controlled transport can be
guaranteed. If at all possible, the content SHOULD be included
indirectly [9], 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 and document any unusual interactions, if they exist.
The following are the most common causes of problems:
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
UASes, 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 that define 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.
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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 that depend on some kind of proxy
processing SHOULD consider how stateless proxies affect that
processing.
Dialog Usages: SIP allows for requests that normally create their own
dialog (such as SUBSCRIBE) to be used within a dialog created by
another method (such as INVITE). In such a case, there are said
to be multiple usages of that dialog. Extensions SHOULD consider
their interaction with dialog usages. In particular, extensions
that define new error response codes SHOULD describe whether that
response code causes the dialog and all usages to terminate, or
just a specific usage.
6. Security Considerations
The nature of this document is such that it does not introduce any
new security considerations. However, many of the principles
described in the document affect whether a potential SIP extension
design is likely to support the SIP security architecture.
7. Acknowledgements
The authors would like to thank Rohan Mahy and Spencer Dawkins for
their comments. Robert Sparks contributed important text on CANCEL
issues. Thanks to Allison Mankin for her support.
8. References
8.1. Normative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, March 1997.
[2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[3] Alvestrand, H., "IETF Policy on Character Sets and Languages",
BCP 18, RFC 2277, January 1998.
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[4] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66, RFC 3986,
January 2005.
[5] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", RFC 4234, October 2005.
[6] Roach, A.B., "Session Initiation Protocol (SIP)-Specific Event
Notification", RFC 3265, June 2002.
[7] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Header Field Parameter Registry for the Session Initiation
Protocol (SIP)", BCP 98, RFC 3968, December 2004.
[8] Camarillo, G., "The Internet Assigned Number Authority (IANA)
Uniform Resource Identifier (URI) Parameter Registry for the
Session Initiation Protocol (SIP)", BCP 99, RFC 3969, December
2004.
[9] Burger, E., Ed., "A Mechanism for Content Indirection in Session
Initiation Protocol (SIP) Messages", RFC 4483, May 2006.
8.2. Informative References
[10] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., and B.
Rosen, "Change Process for the Session Initiation Protocol
(SIP)", BCP 67, RFC 3427, December 2002.
[11] Droms, R., "Dynamic Host Configuration Protocol", RFC 2131,
March 1997.
[12] Sparks, R., "The Session Initiation Protocol (SIP) Refer
Method", RFC 3515, April 2003.
[13] Donovan, S. and J. Rosenberg, "Session Timers in the Session
Initiation Protocol (SIP)", RFC 4028, April 2005.
[14] Sparks, R., "Problems Identified Associated with the Session
Initiation Protocol's (SIP) Non-INVITE Transaction", RFC 4321,
January 2006.
[15] Fielding, R., Gettys, J., Mogul, J., Frystyk, H., Masinter,
L., Leach, P., and T. Berners-Lee, "Hypertext Transfer Protocol
-- HTTP/1.1", RFC 2616, June 1999.
[16] Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
Protocol (RTSP)", RFC 2326, April 1998.
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[17] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,
"SIP: Session Initiation Protocol", RFC 2543, March 1999.
[18] Shacham, A., Monsour, B., Pereira, R., and M. Thomas, "IP
Payload Compression Protocol (IPComp)", RFC 3173, September
2001.
[19] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T.
Haukka, "Security Mechanism Agreement for the Session
Initiation Protocol (SIP)", RFC 3329, January 2003.
[20] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu,
Z., and J. Rosenberg, "Signaling Compression (SigComp)", RFC
3320, January 2003.
[21] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K.
Summers, "Session Initiation Protocol (SIP) Basic Call Flow
Examples", BCP 75, RFC 3665, December 2003.
[22] Rosenberg, J., "A Session Initiation Protocol (SIP) Event
Package for Registrations", RFC 3680, March 2004.
[23] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
Responses in Session Initiation Protocol (SIP)", RFC 3262, June
2002.
[24] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
Method", RFC 3311, October 2002.
[25] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
Considerations Section in RFCs", BCP 26, RFC 2434, October
1998.
[26] Reynolds, J. and R. Braden, "Instructions to Request for
Comments (RFC) Authors", Work in Progress, July 2004.
[27] Rescorla, E. and IAB, "Writing Protocol Models", RFC 4101, June
2005.
[28] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
Session Description Protocol (SDP)", RFC 3264, June 2002.
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Authors' Addresses
Jonathan Rosenberg
Cisco Systems
600 Lanidex Plaza
Parsippany, NJ 07054
US
Phone: +1 973 952-5000
EMail: jdrosen@cisco.com
URI: http://www.jdrosen.net
Henning Schulzrinne
Columbia University
M/S 0401
1214 Amsterdam Ave.
New York, NY 10027
US
EMail: schulzrinne@cs.columbia.edu
URI: http://www.cs.columbia.edu/~hgs
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Full Copyright Statement
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