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Versions: (draft-rosenberg-sip-guidelines) 00 01 02 03 04 05 06 07 08 09 RFC 4485

SIP                                                         J. Rosenberg
Internet-Draft                                               dynamicsoft
Expires: April 26, 2004                                   H. Schulzrinne
                                                     Columbia University
                                                        October 27, 2003


     Guidelines for Authors of Extensions to the Session Initiation
                             Protocol (SIP)
                      draft-ietf-sip-guidelines-07

Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
   groups may also distribute working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time. It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   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 April 26, 2004.

Copyright Notice

   Copyright (C) The Internet Society (2003). All Rights Reserved.

Abstract

   The Session Initiation Protocol (SIP) is a flexible, yet simple tool
   for establishing interactive connections across the Internet. Part of
   this flexibility is the ease with which it can be extended. In order
   to facilitate effective and interoperable extensions to SIP, some
   guidelines need to be followed when developing SIP extensions. This
   document outlines a set of such guidelines for authors of SIP
   extensions.






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Table of Contents

   1.   Terminology  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.   Introduction . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.   Should I define a SIP Extension? . . . . . . . . . . . . . .   5
   3.1  SIP's Solution Space . . . . . . . . . . . . . . . . . . . .   5
   3.2  SIP Architectural Model  . . . . . . . . . . . . . . . . . .   7
   4.   Issues to be Addressed . . . . . . . . . . . . . . . . . . .  10
   4.1  Backwards Compatibility  . . . . . . . . . . . . . . . . . .  10
   4.2  Security . . . . . . . . . . . . . . . . . . . . . . . . . .  12
   4.3  Terminology  . . . . . . . . . . . . . . . . . . . . . . . .  12
   4.4  Syntactic Issues . . . . . . . . . . . . . . . . . . . . . .  13
   4.5  Semantics, Semantics, Semantics  . . . . . . . . . . . . . .  15
   4.6  Examples Section . . . . . . . . . . . . . . . . . . . . . .  16
   4.7  Overview Section . . . . . . . . . . . . . . . . . . . . . .  16
   4.8  IANA Considerations Section  . . . . . . . . . . . . . . . .  16
   4.9  Document Naming Conventions  . . . . . . . . . . . . . . . .  17
   4.10 Additional Considerations for New Methods  . . . . . . . . .  18
   4.11 Additional Considerations for New Header Fields or Header
        Field Parameters . . . . . . . . . . . . . . . . . . . . . .  19
   4.12 Additional Considerations for New Body Types . . . . . . . .  19
   5.   Interactions with SIP Features . . . . . . . . . . . . . . .  20
   6.   Security Considerations  . . . . . . . . . . . . . . . . . .  21
   7.   IANA Considerations  . . . . . . . . . . . . . . . . . . . .  22
   8.   Acknowledgements . . . . . . . . . . . . . . . . . . . . . .  23
        Normative References . . . . . . . . . . . . . . . . . . . .  24
        Informative References . . . . . . . . . . . . . . . . . . .  25
        Authors' Addresses . . . . . . . . . . . . . . . . . . . . .  26
        Intellectual Property and Copyright Statements . . . . . . .  27






















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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.













































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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 [8]. 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, [8] 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.




























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3. Should I define a SIP Extension?

   The first question to be addressed when defining a SIP extension is:
   is a SIP extension the best solution to my problem? SIP has been
   proposed as a solution for numerous problems, including mobility,
   configuration and management, QoS control, call control, caller
   preferences, device control, third party call control, and MPLS path
   setup, to name a few. Clearly, not every problem can be solved by a
   SIP extension. More importantly, some problems that could be solved
   by a SIP extension, probably shouldn't.

   To assist engineers in determining whether a SIP extension is an
   appropriate solution to their problem, we present two broad criteria.
   First, the problem SHOULD fit into the general purvey of SIP's
   solution space. Secondly, the solution MUST conform to the general
   SIP architectural model.

   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 [9].
   SIP's multicast registration mechanisms might supply an alternate way
   to obtain an IP address. This would eliminate the need for DHCP in
   clients. However, we do not believe such extensions are appropriate.
   We believe that protocols should be defined to provide specific,
   narrow functions, rather than being defined based on all
   communications requirements between a pair of devices. The latter
   approach to protocol design yields modular protocols with broad
   application. It also facilitates extensibility and growth; single
   protocols can be removed and changed without affecting the entire
   system. We observe that this approach to protocol engineering mirrors
   object oriented software engineering.

   Our second criteria, that the extension must conform to the general
   SIP architectural model, ensures that the protocol remains manageable
   and broadly applicable.

3.1 SIP's Solution Space

   In order to evaluate the first criteria, it is necessary to define
   exactly what SIP's solution space is, and what it is not.

   SIP is a protocol for initiating, modifying, and terminating
   interactive sessions. This process involves the discovery of users,
   (or more generally, entities that can be communicated with, including
   services, such as voicemail or translation devices) wherever they may



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   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
   have enough information about the session itself in order to make a
   decision on whether to join or not. That information includes data
   about the caller, the purpose for the invitation, and parameters of
   the session itself. For this reason, SIP includes this kind of
   information.

   Part of the process of session initiation is the communication of
   progress and the final results of establishment of the session. SIP
   provides this information as well.

   SIP itself is independent of the session, and the session description
   is delivered as an opaque body within SIP messages. Keeping SIP
   independent of the sessions it initiates and terminates is
   fundamental. As such, there are many functions that SIP explicitly
   does not provide. It is not a session management protocol or a
   conference control protocol. The particulars of the communications
   within the session are outside of SIP. This includes features such as
   media transport, voting and polling, virtual microphone passing,
   chairman election, floor control, and feedback on session quality.

   SIP is not a resource reservation protocol for sessions. This is
   fundamentally because (1) SIP is independent of the underlying
   session it establishes, and (2) the path of SIP messages is
   completely independent from the path that session packets may take.
   The path independence refers to paths within a providers network, and
   the set of providers itself. For example, it is perfectly reasonable
   for a SIP message to traverse a completely different set of
   autonomous systems than the audio in a session SIP establishes.

   SIP is not a general purpose transfer protocol. It is not meant to
   send large amounts of data unrelated to SIP's operation. It is not



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   meant as a replacement for HTTP. This is not to say that carrying
   payloads in SIP messages is never a good thing; in many cases, the
   data is very much related to SIP's operation. In those cases,
   congestion controlled transports end-to-end are critical.

   SIP is not meant to be a general Remote Procedure Call (RPC)
   mechanism. None of its user discovery and registration capabilities
   are needed for RPC, neither are most of its proxy functions.

   SIP is not meant to be used as a strict PSTN signaling replacement.
   It is not a superset of ISUP. While it can support gatewaying of PSTN
   signaling, and can provide many features present in the PSTN, the
   mere existence of a feature or capability in the PSTN is not a
   justification for its inclusion in SIP. Extensions needed to support
   telephony MUST meet the other criteria described here.

   SIP is a poor control protocol. It is not meant to be used for one
   entity to tell another to pick up or answer a phone, send audio using
   a particular codec, or to provide a new value for a configuration
   parameter. Control protocols have different trust relationships than
   is assumed in SIP, and are more centralized in architecture than SIP,
   which is a very distributed protocol.

   There are many network layer services needed to make SIP function.
   These include quality of service, mobility, and security, among
   others. Rather than building these capabilities into SIP itself, they
   SHOULD be developed outside of SIP, and then used by it.
   Specifically, any protocol mechanisms that are needed by SIP, but are
   also needed by many other application layer protocols, SHOULD NOT be
   addressed within SIP.

3.2 SIP Architectural Model

   We describe here some of the primary architectual assumptions which
   underly SIP. Extensions which violate these assumptions should be
   examined more carefully to determine their appropriateness for SIP.

   Session independence: SIP is independent of the session it
      establishes. This includes the type of session, be it audio,
      video, game, chat session, or virtual reality. SIP operation
      SHOULD NOT be dependent on some characteristic of the session. SIP
      is not specific to VoIP only. Any extensions to SIP MUST consider
      the application of SIP to a variety of different session types.

   SIP and Session Path Independence: We have already touched on this
      once, but it is worth noting again. The set of routers and/or
      networks and/or autonomous systems traversed by SIP messages are
      unrelated to the set of routers and/or networks and/or autonomous



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      systems traversed by session packets. They may be the same in some
      cases, but it is fundamental to SIP's architecture that they need
      not be the same. Extensions which only work under some assumption
      of overlap are not generally applicable to SIP's operation and
      should be scrutinized carefully.

   Multi-provider and Multi-hop: SIP assumes that its messages will
      traverse the Internet. That is, SIP works through multiple
      networks administered by different providers. It is also assumed
      that SIP messages traverse many hops (where each hop is a proxy).
      Extensions SHOULD NOT work only under the assumption of a single
      hop or single provider.

   Transactional: SIP is a request/response protocol, possibly enhanced
      with intermediate responses. Many of the rules of 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.






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   The Request URI is the primary key for forwarding: Forwarding logic
      at SIP servers depends primarily on the request URI (this is
      different from request routing in SIP, which uses the Route header
      fields to pass a request through intermediate proxies). It is
      fundamental to the operation of SIP that the request URI indicate
      a resource that, under normal operations, resolves to the desired
      recipient. Extensions SHOULD NOT use other components of the SIP
      message as the primary forwarding key, and SHOULD NOT modify the
      semantics of the request URI.

   Heterogeneity is the norm: SIP supports hetereogeneous devices. It
      has built in mechanisms for determining the set of overlapping
      protocol functionalities. Extensions SHOULD NOT 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 Proxy-Require is



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      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 [10] is supported if it is present in an Allow header.
   If it was, the "transfer" button on the UI could be ``greyed out''
   once the call is established.

   Another type of extension are those which require a proxy to insert
   header fields or header field parameters into a request as it
   traverses the network, or for the UAS to insert header fields or
   header field parameters into a response. For some extensions, if the
   UAC or UAS does not understand these header fields, the message can
   still be processed correctly. These extensions are completely
   backwards compatible.

   Most other extensions of this type require that the server only
   insert the header field or parameter if it is sure the client
   understands it. In this case, these extensions will need to make use
   of the Supported request header field mechanism. This mechanism
   allows a server to determine if the client can understand some
   extension, so that it can apply the extension to the response. By
   their nature, these extensions may not always be able to be applied
   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 [11].

   Yet another type of extension is that which defines new body types to
   be carried in SIP messages. According to the SIP specification,
   bodies must be understood in order to process a request. As such, the
   interoperability issues are similar to new methods. However, the
   Content-Disposition header field has been defined to allow a client
   or server to indicate that the message body is optional [2]. Usage of
   optional bodies, as opposed to mandatory ones, is RECOMMENDED
   wherever possible.

   When a body must be understood to properly process a request or
   response, it is preferred that the sending entity know ahead of time
   whether the new body is understood by the recipient. For requests
   that establish a dialog, inclusion of Accept in the request and its
   success responses is RECOMMENDED. This will allow both parties to
   determine what body types are supported by their peers. Subsequent
   messaging between the peers would then only include body types that
   were indicated as being understood.

4.2 Security

   Security is an important component of any protocol. Designers of SIP
   extensions need to carefully consider if additional security
   requirements are required over those described in 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.

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.




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   Careful attention must be paid to the actual usage of terminology.
   Many documents misuse the terms header, header field, and header
   field values, for example. Document authors SHOULD do a careful
   review of their documents for proper usage of these terms.

4.4 Syntactic Issues

   Extensions that define new methods SHOULD use all capitals for the
   method name. Method names SHOULD be less than 10 characters, and
   SHOULD attempt to convey the general meaning of the request.

      Method names are case sensitive, and therefore there is no
      requirement that they be capitalized. However, using capitalized
      method names keeps with a long-standing convention in SIP and many
      similar protocols, such as HTTP [12] and RTSP [13]

   Extensions that define new header fields that are anticipated to be
   heavily used SHOULD define a compact form if those header fields are
   more than four characters. Compact header fields MUST be a single
   character. When all 26 characters are exhausted, new compact forms
   will no longer be defined. Header field names SHOULD be composed
   primarily of ASCII characters and marks. They SHOULD be descriptive
   but reasonably brief. Although header field names are case
   insensitive, a single common capitalization SHOULD be used throughout
   the document. It is RECOMMENDED that 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 [14]. 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
   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



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   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 \cite{RFC3261}. 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 allow any
   URI, not just SIP URIs.

   Header fields MUST follow the standard formatting for SIP, defined
   as:



   header          = header-name HCOLON header-value
                      *(COMMA header-value)
   header-name     = token
   header-value    = value *(SEMI value-parameter)
   value-parameter = token [EQUAL gen-value]
   gen-value       = token / host / quoted-string
   value           = token / host / quoted-string

   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:





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   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 [15] or
   signalling compression [16].

   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
   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



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   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 [17]. 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 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.

4.8 IANA Considerations Section

   Documents which define new SIP extensions will invariably have IANA
   Considerations sections.




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   If your extension is defining a new event package, you MUST register
   that package. RFC 3265 [6] provides the registration template. See
   [18] for an example of the registration of a new event package.

   If your extension is defining a new header field, you MUST register
   that header field. RFC 3261 [2] provides a registration template. See
   Section 8.2 of RFC 3262 [19] for an example of how to register new
   SIP header fields.

   If your extension is defining a new response code, you MUST register
   that response code. RFC 3261 [2] provides a registration template.
   See [11] for an example of how to register a new response code.

   If your extension is defining a new SIP method, you MUST register
   that method. RFC 3261 [2] provides a registration template. See
   Section 10 of RFC 3311 [20] for an example of how to register a new
   SIP method.

   Many SIP extensions make use of option tags, carried in the Require,
   Proxy-Require and Supported header fields. Section 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 [19] for an example of how to
   register an option tag.

   Some SIP extensions will require establishment of their own IANA
   registries. RFC 2434 [21] 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 RECOMMENED
   that these additional words be descriptive rather than naming the
   header field. For example, the extension for making coffee should not
   be named "The Make-Coffee Header for the Session Initiation
   Protocol".

   For extensions that define new methods, an acceptable template for
   titles is "The Session Initiation Protocol (SIP) X Method" where X is



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   the name of the method.

   Note that the acronymn SIP MUST be expanded in the titles of RFCs, as
   per [22].

4.10 Additional Considerations for New Methods

   Extensions which define new methods SHOULD take into consideration,
   and discuss, the following issues:

   o  Can it contain bodies? If so, what is the meaning of the presence
      of those bodies? What body types are allowed?

   o  Can a transaction with this request method occur while another
      transaction, in the same and/or reverse direction, is in progress?

   o  The extension MUST define which header fields can be present in
      requests of that method. It is RECOMMENDED that this information
      be represented as a new column of Table 2/3 of RFC 3261 [2]. The
      table MUST contain rows for all header fields defined in standards
      track RFCs at the time of writing of the extension.

   o  Can the request be sent within a dialog, or does it establish a
      dialog?

   o  Is it a target refresh request?

   o  Extensions to SIP that define new methods MAY specify whether
      offers and answers can appear in requests of that method or its
      responses. However, those extensions MUST adhere to the protocol
      rules specified in [23], 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 CANCEL request can be used for a particular extension method
      on a method-by-method basis. SIP [2] only allows cancellation of
      INVITE. Extensions that define new methods MUST state whether or
      not transactions initiated by requests with that method can be
      cancelled. Furthermore, the rules a UAS should follow upon
      cancellation of an unanswered request MUST be described. Note
      that, since non-INVITE requests are generally answered



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      immediately, cancellation ususally serves no purpose.

   Note that the reliability mechanisms for all new methods must be the
   same as for BYE. The delayed response feature of INVITE is only
   available in INVITE, never for new methods. This means requests with
   new SIP methods need to be responded to within short time periods (on
   the order of seconds).

4.11 Additional Considerations for New Header Fields or Header Field
     Parameters

   The most important issue for extensions that define new header fields
   or header field parameters is backwards compatibility. See Section
   4.1 for a discussion of the issues. The extension MUST detail how
   backwards compatibility is addressed.

   It is often tempting to avoid creation of a new method by overloading
   an existing method through a header field or parameter. Header fields
   and parameters are not meant to fundamentally alter the meaning of
   the method of the request. A new header field cannot change the basic
   semantic and processing rules of a method. There is no shortage of
   method names, so when an extension changes the basic meaning of a
   request, a new method SHOULD be defined.

   For extensions that define new header fields, the extension MUST
   define the request methods the header field can appear in, and what
   responses it can be used in. It is RECOMMENDED that this information
   be represented as a new row of Table 2/3 of RFC 3261 [2]. The table
   MUST contain columns for all methods defined in standards track RFCs
   at time of writing of the extension.

4.12 Additional Considerations for New Body Types

   Because SIP can run over UDP, extensions that specify the inclusion
   of large bodies are frowned upon unless end-to-end congestion
   controlled transport can be guaranteed. If at all possible, the
   content SHOULD be included indirectly [7] 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.








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5. Interactions with SIP Features

   We have observed that certain capabilities of SIP continually
   interact with extensions in unusual ways. Writers of extensions
   SHOULD consider the interactions of their extensions with these SIP
   capabilities, document any unusual interactions if they exist. The
   most common causes of problems are:

   Forking: Forking by far presents the most troublesome interactions
      with extensions. This is generally because it can cause (1) a
      single transmitted request to be received by an unknown number of
      UASs, and (2) a single INVITE request to have multiple responses.

   CANCEL and ACK: CANCEL and ACK are "special" SIP requests, in that
      they are exceptions to many of the general request processing
      rules. The main reason for this special status is that CANCEL and
      ACK are always associated with another request. New methods SHOULD
      consider the meaning of cancellation, as described above.
      Extensions which defined new header fields in INVITE requests
      SHOULD consider whether they also need to be included in ACK and
      CANCEL. Frequently they do, in order to allow a stateless proxy to
      route the CANCEL or ACK identically to the INVITE.

   Routing: The presence of Route header fields in a request can cause
      it to be sent through intermediate proxies. Requests that
      establish dialogs can be record-routed, so that the initial
      request goes through one set of proxies, and subsequent requests
      through a different set. These SIP features can interact in
      unusual ways with extensions.

   Stateless Proxies: SIP allows a proxy to be stateless. Stateless
      proxies are unable to retransmit messages and cannot execute
      certain services. Extensions which depend on some kind of proxy
      processing SHOULD consider how stateless proxies affect that
      processing.
















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6. Security Considerations

   The nature of this document is such that it does not introduce any
   new security considerations.















































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7. IANA Considerations

   There are no IANA considerations associated with this specification.
















































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8. Acknowledgements

   The authors would like to thank Rohan Mahy for his comments.
















































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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]  Olson, S., "A Mechanism for Content Indirection in Session
        Initiation Protocol (SIP)  Messages",
        draft-ietf-sip-content-indirect-mech-03 (work in progress), June
        2003.


























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Informative References

   [8]   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.

   [9]   Droms, R., "Dynamic Host Configuration Protocol", RFC 2131,
         March 1997.

   [10]  Sparks, R., "The Session Initiation Protocol (SIP) Refer
         Method", RFC 3515, April 2003.

   [11]  Donovan, S. and J. Rosenberg, "Session Timers in the Session
         Initiation Protocol (SIP)", draft-ietf-sip-session-timer-12
         (work in progress), October 2003.

   [12]  Fielding, R., Gettys, J., Mogul, J., Nielsen, H., Masinter, L.,
         Leach, P. and T. Berners-Lee, "Hypertext Transfer Protocol --
         HTTP/1.1", RFC 2616, June 1999.

   [13]  Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming
         Protocol (RTSP)", RFC 2326, April 1998.

   [14]  Handley, M., Schulzrinne, H., Schooler, E. and J. Rosenberg,
         "SIP: Session Initiation Protocol", RFC 2543, March 1999.

   [15]  Shacham, A., Monsour, R., Pereira, R. and M. Thomas, "IP
         Payload Compression Protocol (IPComp)", RFC 2393, December
         1998.

   [16]  Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu, Z.
         and J. Rosenberg, "Signaling Compression (SigComp)", RFC 3320,
         January 2003.

   [17]  Johnston, A., "Session Initiation Protocol Basic Call Flow
         Examples", draft-ietf-sipping-basic-call-flows-02 (work in
         progress), April 2003.

   [18]  Rosenberg, J., "A Session Initiation Protocol (SIP) Event
         Package for Registrations", draft-ietf-sipping-reg-event-00
         (work in progress), October 2002.

   [19]  jdrosen@dynamicsoft.com and schulzrinne@cs.columbia.edu,
         "Reliability of Provisional Responses in Session Initiation
         Protocol (SIP)", RFC 3262, June 2002.

   [20]  Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
         Method", RFC 3311, October 2002.



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   [21]  Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs", BCP 26, RFC 2434, October
         1998.

   [22]  Reynolds, J. and R. Braden, "Instructions to Request for
         Comments (RFC) Authors", draft-rfc-editor-rfc2223bis-07 (work
         in progress), August 2003.

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


Authors' Addresses

   Jonathan Rosenberg
   dynamicsoft
   600 Lanidex Plaza
   Parsippany, NJ  07054
   US

   Phone: +1 973 952-5000
   EMail: jdrosen@dynamicsoft.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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Intellectual Property Statement

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   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.


Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.











































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