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Guidelines for Authors of Extensions to the Session Initiation Protocol (SIP)
draft-ietf-sip-guidelines-09

The information below is for an old version of the document that is already published as an RFC.
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This is an older version of an Internet-Draft that was ultimately published as RFC 4485.
Authors Henning Schulzrinne , Jonathan Rosenberg
Last updated 2015-10-14 (Latest revision 2005-02-16)
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Informational
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IESG IESG state Became RFC 4485 (Informational)
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Responsible AD Allison J. Mankin
Send notices to <rohan@ekabal.com>
draft-ietf-sip-guidelines-09
SIP                                                         J. Rosenberg
Internet-Draft                                             Cisco Systems
Expires: August 17, 2005                                  H. Schulzrinne
                                                     Columbia University
                                                       February 16, 2005

     Guidelines for Authors of Extensions to the Session Initiation
                             Protocol (SIP)
                      draft-ietf-sip-guidelines-09

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of section 3 of RFC 3667.  By submitting this Internet-Draft, each
   author represents that any applicable patent or other IPR claims of
   which he or she is aware have been or will be disclosed, and any of
   which he or she become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
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   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.

   This Internet-Draft will expire on August 17, 2005.

Copyright Notice

   Copyright (C) The Internet Society (2005).

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.

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   This document outlines a set of such guidelines for authors of SIP
   extensions.

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  . . . . . . . . . . . . . . . . . . . . . . .  11
     4.4  Syntactic Issues . . . . . . . . . . . . . . . . . . . . .  11
     4.5  Semantics, Semantics, Semantics  . . . . . . . . . . . . .  14
     4.6  Examples Section . . . . . . . . . . . . . . . . . . . . .  14
     4.7  Overview Section . . . . . . . . . . . . . . . . . . . . .  14
     4.8  IANA Considerations Section  . . . . . . . . . . . . . . .  15
     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 Parameters  . . . . . . . . . . . . . . . .  18
     4.12   Additional Considerations for New Body Types . . . . . .  18
   5.   Interactions with SIP Features . . . . . . . . . . . . . . .  18
   6.   Security Considerations  . . . . . . . . . . . . . . . . . .  19
   7.   IANA Considerations  . . . . . . . . . . . . . . . . . . . .  19
   8.   Acknowledgements . . . . . . . . . . . . . . . . . . . . . .  19
   9.   References . . . . . . . . . . . . . . . . . . . . . . . . .  20
   9.1  Normative References . . . . . . . . . . . . . . . . . . . .  20
   9.2  Informative References . . . . . . . . . . . . . . . . . . .  20
        Authors' Addresses . . . . . . . . . . . . . . . . . . . . .  22
        Intellectual Property and Copyright Statements . . . . . . .  23

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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 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 [10].  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, [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
   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 purview 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 [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 being defined based on 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.

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 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, 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).  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.

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   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, 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 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 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.  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 which 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, RFC 3427
      [10]).

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   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 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 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 be defined which
      only function if all devices support the extension.

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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, 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

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      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 [12] is supported if it is present in an Allow header.
   If it wasn't, 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
   to every response.

   If an extension requires a proxy to insert a header field or

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   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 [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
   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 of their own security techniques that are quite distinct
   by those used by SIP itself.  Extensions should take care not to
   conflate the two.  However, specifications that define extensions
   which impact the media sessions in any way SHOULD consider the
   interactions between SIP and session security mechanisms.

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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 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, strictly speaking, they
   don't have to be capitalized.  However, using capitalized method
   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 which 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

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   Function

   Case sensitivity of parameters and values is a constant source of
   confusion, a difficulty that plagued RFC 2543 [17].  This has been
   made simple through the usage of the BNF constructs of RFC 2234 [5],
   which have clear rules of case sensivitity 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.  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 [3].  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.  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 [4],
   and domain names.

   Extensions which 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:

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   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.

   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
   [18] or signalling compression [20].

   Syntax for header fields is expressed in Augmented Backus-Naur Form
   and MUST follow the format of RFC 2234 [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

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   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 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 [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

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   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]
   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 that
   are not SIP experts.  [27] provides additional guidance on writing
   good overview sections.

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
   [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

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   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.

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 acronymn SIP MUST be expanded in the titles of RFCs, as
   per [26].

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

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      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 [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 complete 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

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   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
   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 (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

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   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

   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.  IANA Considerations

   There are no IANA considerations associated with this specification.

8.  Acknowledgements

   The authors would like to thank Rohan Mahy and Spencer Dawkins for
   their comments.  Robert Sparks contributed important text on CANCEL

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   issues.  Thanks to Allison Mankin for her support.

9.  References

9.1  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.

   [4]  Hinden, R., Carpenter, B. and L. Masinter, "Format for Literal
        IPv6 Addresses in URL's", RFC 2732, December 1999.

   [5]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", RFC 2234, November 1997.

   [6]  Roach, A., "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., "A Mechanism for Content Indirection in Session
        Initiation Protocol (SIP)  Messages",
        draft-ietf-sip-content-indirect-mech-05 (work in progress),
        October 2004.

9.2  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.

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   [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)", draft-ietf-sip-session-timer-15
         (work in progress), August 2004.

   [14]  Sparks, R., "Problems identified associated with the Session
         Initiation Protocol's  non-INVITE Transaction",
         draft-sparks-sip-nit-problems-02 (work in progress), January
         2005.

   [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.

   [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.

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   [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", draft-rfc-editor-rfc2223bis-08 (work
         in progress), July 2004.

   [27]  Rescorla, E., "Writing Protocol Models", draft-iab-model-02
         (work in progress), September 2004.

   [28]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", RFC 3264, June 2002.

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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