<pre>Network Working Group                                       J. Rosenberg
Request for Comments: 4485                                 Cisco Systems
Category: Informational                                   H. Schulzrinne
                                                     Columbia University
                                                                May 2006


                <span class="h1">Guidelines for Authors of Extensions to</span>
                 <span class="h1">the Session Initiation Protocol (SIP)</span>

Status of This Memo

   This memo provides information for the Internet community.  It does
   not specify an Internet standard of any kind.  Distribution of this
   memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

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






















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Table of Contents

   <a href="#section-1">1</a>. Introduction ....................................................<a href="#page-2">2</a>
   <a href="#section-2">2</a>. Terminology .....................................................<a href="#page-3">3</a>
   <a href="#section-3">3</a>. Should I Define a SIP Extension? ................................<a href="#page-3">3</a>
      <a href="#section-3.1">3.1</a>. SIP's Solution Space .......................................<a href="#page-4">4</a>
      <a href="#section-3.2">3.2</a>. SIP Architectural Model ....................................<a href="#page-5">5</a>
   <a href="#section-4">4</a>. Issues to Be Addressed ..........................................<a href="#page-7">7</a>
      <a href="#section-4.1">4.1</a>. Backwards Compatibility ....................................<a href="#page-7">7</a>
      <a href="#section-4.2">4.2</a>. Security ..................................................<a href="#page-10">10</a>
      <a href="#section-4.3">4.3</a>. Terminology ...............................................<a href="#page-10">10</a>
      <a href="#section-4.4">4.4</a>. Syntactic Issues ..........................................<a href="#page-10">10</a>
      <a href="#section-4.5">4.5</a>. Semantics, Semantics, Semantics ...........................<a href="#page-13">13</a>
      <a href="#section-4.6">4.6</a>. Examples Section ..........................................<a href="#page-14">14</a>
      <a href="#section-4.7">4.7</a>. Overview Section ..........................................<a href="#page-14">14</a>
      <a href="#section-4.8">4.8</a>. IANA Considerations Section ...............................<a href="#page-14">14</a>
      <a href="#section-4.9">4.9</a>. Document-Naming Conventions ...............................<a href="#page-16">16</a>
      <a href="#section-4.10">4.10</a>. Additional Considerations for New Methods ................<a href="#page-16">16</a>
      4.11. Additional Considerations for New Header Fields
            or Header Field ..........................................<a href="#page-17">17</a>
      <a href="#section-4.12">4.12</a>. Additional Considerations for New Body Types .............<a href="#page-18">18</a>
   <a href="#section-5">5</a>. Interactions with SIP Features .................................<a href="#page-18">18</a>
   <a href="#section-6">6</a>. Security Considerations ........................................<a href="#page-19">19</a>
   <a href="#section-7">7</a>. Acknowledgements ...............................................<a href="#page-19">19</a>
   <a href="#section-8">8</a>. References .....................................................<a href="#page-19">19</a>
      <a href="#section-8.1">8.1</a>. Normative References ......................................<a href="#page-19">19</a>
      <a href="#section-8.2">8.2</a>. Informative References ....................................<a href="#page-20">20</a>

<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

   The Session Initiation Protocol (SIP) [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] is a flexible yet simple
   tool for establishing interactive communications sessions across the
   Internet.  Part of this flexibility is the ease with which it can be
   extended (with new methods, new header fields, new body types, and
   new parameters), and there have been countless proposals that have
   been made to do just that.  An IETF process has been put into place
   that defines how extensions are to be made to the SIP protocol [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>].
   That process is designed to ensure that extensions are made that are
   appropriate for SIP (as opposed to being done in some other
   protocol), that these extensions fit within the model and framework
   provided by SIP and are consistent with its operation, and that these
   extensions solve problems generically rather than for a specific use
   case.  However, [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>] does not provide the technical guidelines needed
   to assist that process.  This specification helps to meet that need.

   This specification first provides a set of guidelines to help decide
   whether a certain piece of functionality is appropriately done in
   SIP.  Assuming the functionality is appropriate, it then points out



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   issues that extensions should deal with from within their
   specification.  Finally, it discusses common interactions with
   existing SIP features that often cause difficulties in extensions.

<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  Terminology</span>

   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 <a href="./rfc2119">RFC 2119</a> [<a href="#ref-1" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">1</a>] and
   indicate requirement levels for compliant implementations.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Should I Define a SIP Extension?</span>

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

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

   Although the first criteria might seem obvious, we have observed that
   numerous extensions to SIP have been proposed because some function
   is needed in a device that also speaks SIP.  The argument is
   generally given that "I'd rather implement one protocol than many".
   As an example, user agents, like all other IP hosts, need some way to
   obtain their IP address.  This is generally done through DHCP [<a href="#ref-11" title="&quot;Dynamic Host Configuration Protocol&quot;">11</a>].
   SIP's multicast registration mechanisms might supply an alternate way
   to obtain an IP address.  This would eliminate the need for DHCP in
   clients.  However, we do not believe such extensions are appropriate.
   We believe that protocols should be defined to provide specific,
   narrow functions, rather than be defined for all protocols needed
   between a pair of devices.  The former approach to protocol design
   yields modular protocols with broad application.  It also facilitates
   extensibility and growth; single protocols can be removed and changed
   without affecting the entire system.  We observe that this approach
   to protocol engineering mirrors object-oriented software engineering.

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



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<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  SIP's Solution Space</span>

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

   SIP is a protocol for initiating, modifying, and terminating
   interactive sessions.  This process involves the discovery of users,
   (or, more generally, entities that can be communicated with,
   including services, such as voicemail or translation devices)
   wherever they may be located, so that a description of the session
   can be delivered to the user.  It is assumed that these users or
   communications entities are mobile, and that their point of
   attachment to the network changes over time.  The primary purpose of
   SIP is a rendezvous function, to allow a request initiator to deliver
   a message to a recipient wherever they may be.  Such a rendezvous is
   needed to establish a session, but it can be used for other purposes
   related to communications, such as querying for capabilities or
   delivery of an instant message.

   Much of SIP focuses on this discovery and rendezvous component.  Its
   ability to fork, its registration capabilities, and its routing
   capabilities are all present for the singular purpose of finding the
   desired user wherever they may be.  As such, features and
   capabilities such as personal mobility, automatic call distribution,
   and follow-me are well within the SIP solution space.

   Session initiation also depends on the ability of the called party to
   have enough information about the session itself to make a decision
   on whether to join.  That information includes data about the caller,
   the purpose for the invitation, and parameters of the session itself.
   For this reason, SIP includes this kind of information.

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

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

   SIP is not a resource reservation protocol for sessions.  This is
   fundamentally because (1) SIP is independent of the underlying



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   session it establishes, and (2) the path of SIP messages is
   completely independent from the path that session packets may take.
   The path independence refers to paths within a provider's network and
   the set of providers itself.  For example, it is perfectly reasonable
   for a SIP message to traverse a completely different set of
   autonomous systems than the audio in a session SIP establishes.

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

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

   SIP is not meant to be used as a strict Public Switched Telephone
   Network (PSTN) signaling replacement.  It is not a superset of the
   Integrated Services Digital Network (ISDN) User Part (ISUP).
   Although it can support gatewaying of PSTN signaling and can provide
   many features present in the PSTN, the mere existence of a feature or
   capability in the PSTN is not a justification for its inclusion in
   SIP.  Extensions needed to support telephony MUST meet the other
   criteria described here.

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

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

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  SIP Architectural Model</span>

   We describe here some of the primary architectural assumptions that
   underlie SIP.  Extensions that violate these assumptions should be
   examined more carefully to determine their appropriateness for SIP.




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

   SIP and Session path independence:  We have already touched on this
      once, but it is worth noting again.  The set of routers, networks,
      and/or autonomous systems traversed by SIP messages are unrelated
      to the set of routers, networks, and/or autonomous systems
      traversed by session packets.  They may be the same in some cases,
      but it is fundamental to SIP's architecture that they need not be
      the same.  Standards-track extensions MUST NOT be defined that
      work only when the signaling and session paths are coupled.  Non-
      standard P-header extensions [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>] are required for any extension
      that only works in such a case.

   Multi-provider and multi-hop:  SIP assumes that its messages will
      traverse the Internet.  That is, SIP works through multiple
      networks administered by different providers.  It is also assumed
      that SIP messages traverse many hops (where each hop is a proxy).
      Extensions MUST NOT work only under the assumption of a single hop
      or specialized network topology.  They SHOULD avoid the assumption
      of a single SIP provider (but see the use of P-Headers, per <a href="./rfc3427">RFC</a>
      <a href="./rfc3427">3427</a> [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>]).

   Transactional:  SIP is a request/response protocol, possibly enhanced
      with intermediate responses.  Many of the rules of operation in
      SIP are based on general processing of requests and responses.
      This includes the reliability mechanisms, routing mechanisms, and
      state maintenance rules.  Extensions SHOULD NOT add messages that
      are not within the request-response model.

   Proxies can ignore bodies:  In order for proxies to scale well, they
      must be able to operate with minimal message processing.  SIP has
      been engineered so that proxies can always ignore bodies.
      Extensions SHOULD NOT require proxies to examine bodies.

   Proxies don't need to understand the method:  Processing of requests
      in proxies does not depend on the method, except for the well-
      known methods INVITE, ACK, and CANCEL.  This allows for
      extensibility.  Extensions MUST NOT define new methods that must
      be understood by proxies.







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   INVITE messages carry full state:  An initial INVITE message for a
      session is nearly identical (the exception is the tag) to a re-
      INVITE message to modify some characteristic of the session.  This
      full state property is fundamental to SIP and is critical for
      robustness of SIP systems.  Extensions SHOULD NOT modify INVITE
      processing such that data spanning multiple INVITEs must be
      collected in order to perform some feature.

   Generality over efficiency:  Wherever possible, SIP has favored
      general-purpose components rather than narrow ones.  If some
      capability is added to support one service but a slightly broader
      capability can support a larger variety of services (at the cost
      of complexity or message sizes), the broader capability SHOULD be
      preferred.

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

   Heterogeneity is the norm:  SIP supports heterogeneous devices.  It
      has built-in mechanisms for determining the set of overlapping
      protocol functionalities.  Extensions SHOULD NOT be defined that
      only function if all devices support the extension.

<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  Issues to Be Addressed</span>

   Given an extension has met the litmus tests in the previous section,
   there are several issues that all extensions should take into
   consideration.

<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a>.  Backward Compatibility</span>

   One of the most important issues to consider is whether the new
   extension is backward compatible with baseline SIP.  This is tightly
   coupled with how the Require, Proxy-Require, and Supported header
   fields are used.

   If an extension consists of new header fields or header field
   parameters inserted by a user agent in a request with an existing
   method, and the request cannot be processed reasonably by a proxy
   and/or user agent without understanding the header fields or
   parameters, the extension MUST mandate the usage of the Require
   and/or Proxy-Require header fields in the request.  These extensions



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   are not backwards compatible with SIP.  The result of mandating usage
   of these header fields means that requests cannot be serviced unless
   the entities being communicated with also understand the extension.
   If some entity does not understand the extension, the request will be
   rejected.  The UAC can then handle this in one of two ways.  In the
   first, the request simply fails, and the service cannot be provided.
   This is basically an interoperability failure.  In the second case,
   the UAC retries the request without the extension.  This will
   preserve interoperability, at the cost of a "dual stack"
   implementation in a UAC (processing rules for operation with and
   without the extension).  As the number of extensions increases, this
   leads to an exponential explosion in the sets of processing rules a
   UAC may need to implement.  The result is excessive complexity.

   Because of the possibility of interoperability and complexity
   problems that result from the usage of Require and Proxy-Require, we
   believe the following guidelines are appropriate:

   o  The usage of these header fields in requests for basic SIP
      services (in particular, session initiation and termination) is
      NOT RECOMMENDED.  The less frequently a particular extension is
      needed in a request, the more reasonable it is to use these header
      fields.

   o  The Proxy-Require header field SHOULD be avoided at all costs.
      The failure likelihood in an individual proxy stays constant, but
      the path failure grows exponentially with the number of hops.  On
      the other hand, the Require header field only mandates that a
      single entity, the UAS, support the extension.  Usage of
      Proxy-Require is thus considered exponentially worse than usage of
      the Require header field.

   o  If either Require or Proxy-Require are used by an extension, the
      extension SHOULD discuss how to fall back to baseline SIP
      operation if the request is rejected with a 420 response.

   Extensions that define new methods do not need to use the Require
   header field.  SIP defines mechanisms that allow a UAC to know
   whether a new method is understood by a UAS.  This includes both the
   OPTIONS request and the 405 (Method Not Allowed) response with the
   Allow header field.  It is fundamental to SIP that proxies need not
   understand the semantics of a new method in order to process it.  If
   an extension defines a new method that must be understood by proxies
   in order to be processed, a Proxy-Require header field is needed.  As
   discussed above, these kinds of extensions are frowned upon.

   In order to achieve backwards compatibility for extensions that
   define new methods, the Allow header field is used.  There are two



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   types of new methods - those that are used for established dialogs
   (initiated by INVITE, for example), and those that are sent as the
   initial request to a UA.  Since INVITE and its response both SHOULD
   contain an Allow header field, a UA can readily determine whether the
   new method can be supported within the dialog.  For example, once an
   INVITE dialog is established, a user agent could determine whether
   the REFER method [<a href="#ref-12" title="&quot;The Session Initiation Protocol (SIP) Refer Method&quot;">12</a>] is supported if it is present in an Allow
   header field.  If it wasn't, the "transfer" button on the UI could be
   "greyed out" once the call is established.

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

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

   If an extension requires a proxy to insert a header field or
   parameter into a request and this header field or parameter needs to
   be understood by both UAC and UAS to be executed correctly, a
   combination of the Require and the Supported mechanism will need to
   be used.  The proxy can insert a Require header field into the
   request if the Supported header field is present.  An example of such
   an extension is the SIP Session Timer [<a href="#ref-13" title="&quot;Session Timers in the Session Initiation Protocol (SIP)&quot;">13</a>].

   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 [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].  Extensions that define or require new body types
   SHOULD make them optional for the user agent to process.

   When a body must be understood to properly process a request or
   response, it is preferred that the sending entity know ahead of time
   whether the new body is understood by the recipient.  For requests
   that establish a dialog, inclusion of Accept in the request and its



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   success responses is RECOMMENDED.  This will allow both parties to
   determine what body types are supported by their peers.  Subsequent
   messaging between the peers would then only include body types that
   were indicated as being understood.

<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a>.  Security</span>

   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 <a href="./rfc3261">RFC 3261</a>.
   Frequently, authorization requirements and requirements for end-to-
   end integrity are the most overlooked.

   SIP extensions MUST consider how (or if) they affect usage of the
   general SIP security mechanisms.  Most extensions should not require
   any new security capabilities beyond general-purpose SIP.  If they
   do, it is likely that the security mechanism has more general-purpose
   application and should be considered an extension in its own right.

   Overall system security requires that both the SIP signaling and the
   media sessions it established be secured.  The media sessions
   normally use their own security techniques, which are quite distinct
   from those used by SIP itself.  Extensions should take care not to
   conflate the two.  However, specifications that define extensions
   that impact the media sessions in any way SHOULD consider the
   interactions between SIP and session security mechanisms.

<span class="h3"><a class="selflink" id="section-4.3" href="#section-4.3">4.3</a>.  Terminology</span>

   <a href="./rfc3261">RFC 3261</a> has an extensive terminology section that defines terms such
   as caller, callee, user agent, and header field.  All SIP extensions
   MUST conform to this terminology.  They MUST NOT define new terms
   that describe concepts already defined by a term in another SIP
   specification.  If new terminology is needed, it SHOULD appear in a
   separate section towards the beginning of the document.

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

<span class="h3"><a class="selflink" id="section-4.4" href="#section-4.4">4.4</a>.  Syntactic Issues</span>

   Extensions that define new methods SHOULD use all capitals for the
   method name.  Method names SHOULD be shorter than 10 characters and
   SHOULD attempt to convey the general meaning of the request.  Method
   names are case sensitive, and therefore, strictly speaking, they
   don't have to be capitalized.  However, using capitalized method



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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   names keeps with a long-standing convention in SIP and many similar
   protocols, such as HTTP [<a href="#ref-15" title="&quot;Hypertext Transfer Protocol -- HTTP/1.1&quot;">15</a>] and RTSP [<a href="#ref-16" title="&quot;Real Time Streaming Protocol (RTSP)&quot;">16</a>].

   Extensions that define new header fields that are anticipated to be
   heavily used MAY define a compact form if those header fields are
   more than six characters.  "Heavily used" means that the percentage
   of all emitted messages that contain that header field is over thirty
   percent.  Usage of compact forms in these cases is only a MAY because
   there are better approaches for reducing message overhead [<a href="#ref-20" title="&quot;Signaling Compression (SigComp)&quot;">20</a>].
   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
   <a href="./rfc3261#section-25.1">RFC 3261, Section&nbsp;25.1</a>, and thus include the upper and lowercase
   letters, the digits 0 through 9, the HYPHEN-MINUS (-), FULL STOP (.),
   EXCLAMATION MARK (!), PERCENT SIGN (%), ASTERISK (*), LOW LINE (_),
   PLUS SIGN (+), GRAVE ACCENT (`), APOSTROPHE ('), and TILDE (~).  They
   SHOULD be descriptive but reasonably brief.  Although header field
   names are case insensitive, a single common capitalization SHOULD be
   used throughout the document.  It is RECOMMENDED that each English
   word present in the header field name have its first letter
   capitalized.  For example, "ThisIsANewHeader".

   As an example, the following are poor choices for header field names:

   ThisIsMyNewHeaderThatDoesntDoVeryMuchButItHasANiceName
   --.!A
   Function

   Case sensitivity of parameters and values is a constant source of
   confusion, a difficulty that plagued <a href="./rfc2543">RFC 2543</a> [<a href="#ref-17" title="&quot;SIP: Session Initiation Protocol&quot;">17</a>].  This has been
   simplified through the usage of the BNF constructs of <a href="./rfc4234">RFC 4234</a> [<a href="#ref-5" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">5</a>],
   which have clear rules of case sensitivity and insensitivity.
   Therefore, the BNF for an extension completely defines the matching
   rules.

   Extensions MUST be consistent with the SIP conventions for case
   sensitivity.  Methods MUST be case sensitive.  Header field names
   MUST be case insensitive.  Header field parameter names MUST be case
   insensitive.  Header field values and parameter values are sometimes
   case sensitive, and sometimes case insensitive.  However, generally,
   they SHOULD be case insensitive.  Defining a case-sensitive component
   requires explicitly listing each character through its ASCII code.

   Extensions that contain freeform text MUST allow that text to be
   UTF-8, as per the IETF policies on character set usage [<a href="#ref-3" title="&quot;IETF Policy on Character Sets and Languages&quot;">3</a>].  This
   ensures that SIP remains an internationalized standard.  As a general
   guideline, freeform text is never needed by programs to perform
   protocol processing.  It is usually entered by and displayed to the



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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   user.  If an extension uses a parameter that can contain UTF-8-
   encoded characters, and that extension requires a comparison to be
   made of this parameter to other parameters, the comparison MUST be
   case sensitive.  Case-insensitive comparison rules for UTF-8 text
   are, at this time, impossible and MUST be avoided.

   Extensions that make use of dates MUST use the SIP-Date BNF defined
   in <a href="./rfc3261">RFC 3261</a>.  No other date formats are allowed.  However, the usage
   of absolute dates to determine intervals (for example, the time at
   which some timer fires) is NOT RECOMMENDED.  This is because it
   requires synchronized time between peers, and this is frequently not
   the case.  Therefore, relative times, expressed in numbers of
   seconds, SHOULD be used.

   Extensions that include network-layer addresses SHOULD permit dotted
   quad IPv4 addresses, IPv6 addresses in the format described in [<a href="#ref-4" title="&quot;Uniform Resource Identifier (URI): Generic Syntax&quot;">4</a>],
   and domain names.

   Extensions that have header fields containing URIs SHOULD be explicit
   about which URI schemes can be used in that header field.  Header
   fields SHOULD allow the broadest set of URI schemes possible that are
   a match for the semantics of the header field.

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

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

   In some cases, this form is not sufficient.  That is the case for
   header fields that express descriptive text meant for human
   consumption.  An example is the Subject header field in SIP [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].  In
   this case, an alternate form is:

   header          = header-name HCOLON [TEXT-UTF8-TRIM]

   Developers of extensions SHOULD allow for extension parameters in
   their header fields.








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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   Header fields that contain a list of URIs SHOULD follow the same
   syntax as the Contact header field in SIP.  Implementors are also
   encouraged to wrap these URI in angle brackets, "&lt;" and "&gt;", at all
   times.  We have found this to be a frequently misimplemented feature.

   Beyond the compact form, there is no need to define compressed
   versions of header field values.  Compression of SIP messages SHOULD
   be handled at lower layers, for example, using IP payload compression
   [<a href="#ref-18" title="&quot;IP Payload Compression Protocol (IPComp)&quot;">18</a>] or signalling compression [<a href="#ref-20" title="&quot;Signaling Compression (SigComp)&quot;">20</a>].

   Syntax for header fields is expressed in Augmented Backus-Naur Form
   and MUST follow the format of <a href="./rfc4234">RFC 4234</a> [<a href="#ref-5" title="&quot;Augmented BNF for Syntax Specifications: ABNF&quot;">5</a>].  Extensions MUST make use
   of the primitive components defined in <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].  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.

<span class="h3"><a class="selflink" id="section-4.5" href="#section-4.5">4.5</a>.  Semantics, Semantics, Semantics</span>

   Developers of protocols often get caught up in syntax issues, without
   spending enough time on semantics.  The semantics of a protocol are
   far more important.  SIP extensions MUST clearly define the semantics
   of the extensions.  Specifically, the extension MUST specify the
   behaviors expected of a UAC, UAS, and proxy in processing the
   extension.  This is often best described by having separate sections
   for each of these three elements.  Each section SHOULD step through
   the processing rules in temporal order of the most common messaging
   scenario.

   Processing rules generally specify actions to be taken (in terms of
   messages to be sent, variables to be stored, and rules to be
   followed) on receipt of messages and expiration of timers.  If an
   action requires transmission of a message, the rule SHOULD outline
   requirements for insertion of header fields or other information in
   the message.



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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   The extension SHOULD specify procedures to be taken in exceptional
   conditions that are recoverable, or that require some kind of user
   intervention.  Handling of unrecoverable errors does not require
   specification.

<span class="h3"><a class="selflink" id="section-4.6" href="#section-4.6">4.6</a>.  Examples Section</span>

   The specification SHOULD contain a section that gives examples of
   call flows and message formatting.  Extensions that define
   substantial new syntax SHOULD include examples of messages containing
   that syntax.  Examples of message flows should be given to cover
   common cases and at least one failure or unusual case.

   For an example of how to construct a good examples section, see the
   message flows and message formatting defined in the Basic Call Flows
   specification [<a href="#ref-21" title="&quot;Session Initiation Protocol (SIP) Basic Call Flow Examples&quot;">21</a>].  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.

<span class="h3"><a class="selflink" id="section-4.7" href="#section-4.7">4.7</a>.  Overview Section</span>

   Too often, extension documents dive into detailed syntax and
   semantics without giving a general overview of operation.  This makes
   understanding of the extension harder.  It is RECOMMENDED that
   extensions have a protocol overview section that discusses the basic
   operation of the extension.  Basic operation usually consists of the
   message flow, in temporal order, for the most common case covered by
   the extension.  The most important processing rules for the elements
   in the call flow SHOULD be mentioned.  Usage of the <a href="./rfc2119">RFC 2119</a> [<a href="#ref-1" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">1</a>]
   terminology in the overview section is NOT RECOMMENDED, and the
   specification should explicitly state that the overview is tutorial
   in nature only.  This section SHOULD expand all acronyms, even those
   common in SIP systems, and SHOULD be understandable to readers who
   are not SIP experts. [<a href="#ref-27" title="&quot;Writing Protocol Models&quot;">27</a>] provides additional guidance on writing
   good overview sections.

<span class="h3"><a class="selflink" id="section-4.8" href="#section-4.8">4.8</a>.  IANA Considerations Section</span>

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

   If your extension is defining a new event package, you MUST register
   that package.  <a href="./rfc3265">RFC 3265</a> [<a href="#ref-6" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">6</a>] provides the registration template.  See



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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   [<a id="ref-22">22</a>] for an example of the registration of a new event package.  As
   discussed in <a href="./rfc3427">RFC 3427</a> [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>], 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.  <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] provides a registration template.
   See <a href="./rfc3262#section-8.2">Section&nbsp;8.2 of RFC 3262</a> [<a href="#ref-23" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">23</a>] for an example of how to register
   new SIP header fields.  Both standards-track and informational
   P-header specifications can register new header fields [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>].

   If your extension is defining a new response code, you MUST register
   that response code.  <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] provides a registration template.
   See <a href="./rfc3329#section-6.4">Section&nbsp;6.4 of RFC 3329</a> [<a href="#ref-19" title="&quot;Security Mechanism Agreement for the Session Initiation Protocol (SIP)&quot;">19</a>] for an example of how to register a
   new response code.  As discussed in <a href="./rfc3427">RFC 3427</a> [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>], only standards-
   track documents can register new response codes.

   If your extension is defining a new SIP method, you MUST register
   that method.  <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] provides a registration template.  See
   <a href="./rfc3311#section-10">Section&nbsp;10 of RFC 3311</a> [<a href="#ref-24" title="&quot;The Session Initiation Protocol (SIP) UPDATE Method&quot;">24</a>] for an example of how to register a new
   SIP method.  As discussed in <a href="./rfc3427">RFC 3427</a> [<a href="#ref-10" title="&quot;Change Process for the Session Initiation Protocol (SIP)&quot;">10</a>], 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 <a href="./rfc3968">RFC</a>
   <a href="./rfc3968">3968</a> [<a href="#ref-7" title="&quot;The Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the Session Initiation Protocol (SIP)&quot;">7</a>].  <a href="#section-4.1">Section 4.1</a> of that specification provides a template.
   Only IETF approved specifications can register new header field
   parameters.  However, there is no requirement that these be standards
   track.

   If your extension is defining a new SIP URI parameter, you MUST
   register that URI parameter per the guidelines in <a href="./rfc3969">RFC 3969</a> [<a href="#ref-8" title="&quot;The Internet Assigned Number Authority (IANA) Uniform Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP)&quot;">8</a>].
   <a href="#section-4.1">Section 4.1</a> 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.  <a href="#section-4.1">Section 4.1</a> 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.  <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] provides a registration template.  See
   <a href="./rfc3262#section-8.1">Section&nbsp;8.1 of RFC 3262</a> [<a href="#ref-23" title="&quot;Reliability of Provisional Responses in Session Initiation Protocol (SIP)&quot;">23</a>] 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.  <a href="./rfc2434">RFC 2434</a> [<a href="#ref-25" title="">25</a>] provides guidance on how and when IANA
   registries are established.  For an example of how to set one up, see
   <a href="./rfc3265#section-6">Section&nbsp;6 of RFC 3265</a> [<a href="#ref-6" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">6</a>] for an example.




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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


<span class="h3"><a class="selflink" id="section-4.9" href="#section-4.9">4.9</a>.  Document-Naming Conventions</span>

   An important decision to be made about the extension is its title.
   The title MUST indicate that the document is an extension to SIP.  It
   is RECOMMENDED that the title follow the basic form of "A [summary of
   function] for the Session Initiation Protocol (SIP)", where the
   summary of function is a one- to three-word description of the
   extension.  For example, if an extension defines a new header field,
   called Make-Coffee, for making coffee, the title would read, "Making
   Coffee with the Session Initiation Protocol (SIP)".  It is
   RECOMMENDED that these additional words be descriptive rather than
   naming the header field.  For example, the extension for making
   coffee should not be named "The Make-Coffee Header for the Session
   Initiation Protocol".

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

   Note that the acronym SIP MUST be expanded in the titles of RFCs, as
   per [<a href="#ref-26" title="&quot;Instructions to Request for Comments (RFC) Authors&quot;">26</a>].

<span class="h3"><a class="selflink" id="section-4.10" href="#section-4.10">4.10</a>.  Additional Considerations for New Methods</span>

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

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

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

   o  The extension MUST define which header fields can be present in
      requests of that method.  It is RECOMMENDED that this information
      be represented as a new column of Table 2/3 of <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].  The
      table MUST contain rows for all header fields defined in
      standards-track RFCs at the time of writing of the extension.

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

   o  Is it a target refresh request?

   o  Extensions to SIP that define new methods MAY specify whether
      offers and answers can appear in requests of that method or its
      responses.  However, those extensions MUST adhere to the protocol




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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


      rules specified in [<a href="#ref-28" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">28</a>] and MUST adhere to the additional
      constraints for offers and answers as specified in SIP [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].

   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 [<a href="#ref-6" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">6</a>].

   o  The SIP specification [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>] 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 [<a href="#ref-14" title="&quot;Problems Identified Associated with the Session Initiation Protocol's (SIP) Non-INVITE Transaction&quot;">14</a>] has
      determined that non-INVITE transactions must be completed as soon
      as possible.  New methods must not plan for the transaction to
      pend long enough for CANCEL to be meaningful.  Thus, new methods
      MUST declare that transactions initiated by requests with that
      method cannot be canceled.  Future work may relax this
      restriction, at which point these guidelines will be revised.

   o  New methods that establish a new dialog must discuss the impacts
      of forking.  The design of such new methods should follow the
      pattern of requiring an immediate request in the reverse direction
      from the request establishing a dialog, similar to the immediate
      NOTIFY sent when a subscription is created per <a href="./rfc3265">RFC 3265</a> [<a href="#ref-6" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">6</a>].

   The reliability mechanisms for all new methods must be the same as
   for BYE.  The delayed response feature of INVITE is only available in
   INVITE, never for new methods.  The design of new methods must
   encourage an immediate response.  If the application being enabled
   requires a delay, the design SHOULD follow a pattern using multiple
   transactions, similar to <a href="./rfc3265">RFC 3265</a>'s use of NOTIFYs with different
   Subscription-State header field values (pending and active in
   particular) in response to SUBSCRIBE [<a href="#ref-6" title="&quot;Session Initiation Protocol (SIP)-Specific Event Notification&quot;">6</a>].

<span class="h3"><a class="selflink" id="section-4.11" href="#section-4.11">4.11</a>.  Additional Considerations for New Header Fields or Header Field</span>
<span class="h3">       Parameters</span>

   The most important issue for extensions that define new header fields
   or header field parameters is backwards compatibility.  See
   <a href="#section-4.1">Section 4.1</a> for a discussion of the issues.  The extension MUST
   detail how backwards compatibility is addressed.

   It is often tempting to avoid creation of a new method by overloading
   an existing method through a header field or parameter.  Header
   fields and parameters are not meant to fundamentally alter the
   meaning of the method of the request.  A new header field cannot



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   change the basic semantic and processing rules of a method.  There is
   no shortage of method names, so when an extension changes the basic
   meaning of a request, a new method SHOULD be defined.

   For extensions that define new header fields, the extension MUST
   define the request methods the header field can appear in, and what
   responses it can be used in.  It is RECOMMENDED that this information
   be represented as a new row of Table 2/3 of <a href="./rfc3261">RFC 3261</a> [<a href="#ref-2" title="&quot;SIP: Session Initiation Protocol&quot;">2</a>].  The table
   MUST contain columns for all methods defined in standards-track RFCs
   at the time of writing of the extension.

<span class="h3"><a class="selflink" id="section-4.12" href="#section-4.12">4.12</a>.  Additional Considerations for New Body Types</span>

   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 [<a href="#ref-9" title="&quot;A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages&quot;">9</a>], 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.

<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Interactions with SIP Features</span>

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

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

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



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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


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

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

   Dialog Usages: SIP allows for requests that normally create their own
      dialog (such as SUBSCRIBE) to be used within a dialog created by
      another method (such as INVITE).  In such a case, there are said
      to be multiple usages of that dialog.  Extensions SHOULD consider
      their interaction with dialog usages.  In particular, extensions
      that define new error response codes SHOULD describe whether that
      response code causes the dialog and all usages to terminate, or
      just a specific usage.

<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Security Considerations</span>

   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.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  Acknowledgements</span>

   The authors would like to thank Rohan Mahy and Spencer Dawkins for
   their comments.  Robert Sparks contributed important text on CANCEL
   issues.  Thanks to Allison Mankin for her support.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  References</span>

<span class="h3"><a class="selflink" id="section-8.1" href="#section-8.1">8.1</a>.  Normative References</span>

   [<a id="ref-1">1</a>]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
        Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-2">2</a>]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
        Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
        Session Initiation Protocol", <a href="./rfc3261">RFC 3261</a>, June 2002.

   [<a id="ref-3">3</a>]  Alvestrand, H., "IETF Policy on Character Sets and Languages",
        <a href="https://www.rfc-editor.org/bcp/bcp18">BCP 18</a>, <a href="./rfc2277">RFC 2277</a>, January 1998.



<span class="grey">Rosenberg &amp; Schulzrinne      Informational                     [Page 19]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-20" ></span>
<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   [<a id="ref-4">4</a>]  Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
        Resource Identifier (URI): Generic Syntax", STD 66, <a href="./rfc3986">RFC 3986</a>,
        January 2005.

   [<a id="ref-5">5</a>]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
        Specifications: ABNF", <a href="./rfc4234">RFC 4234</a>, October 2005.

   [<a id="ref-6">6</a>]  Roach, A.B., "Session Initiation Protocol (SIP)-Specific Event
        Notification", <a href="./rfc3265">RFC 3265</a>, June 2002.

   [<a id="ref-7">7</a>]  Camarillo, G., "The Internet Assigned Number Authority (IANA)
        Header Field Parameter Registry for the Session Initiation
        Protocol (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp98">BCP 98</a>, <a href="./rfc3968">RFC 3968</a>, December 2004.

   [<a id="ref-8">8</a>]  Camarillo, G., "The Internet Assigned Number Authority (IANA)
        Uniform Resource Identifier (URI) Parameter Registry for the
        Session Initiation Protocol (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp99">BCP 99</a>, <a href="./rfc3969">RFC 3969</a>, December
        2004.

   [<a id="ref-9">9</a>]  Burger, E., Ed., "A Mechanism for Content Indirection in Session
        Initiation Protocol (SIP)  Messages", <a href="./rfc4483">RFC 4483</a>, May 2006.

<span class="h3"><a class="selflink" id="section-8.2" href="#section-8.2">8.2</a>.  Informative References</span>

   [<a id="ref-10">10</a>]  Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., and B.
         Rosen, "Change Process for the Session Initiation Protocol
         (SIP)", <a href="https://www.rfc-editor.org/bcp/bcp67">BCP 67</a>, <a href="./rfc3427">RFC 3427</a>, December 2002.

   [<a id="ref-11">11</a>]  Droms, R., "Dynamic Host Configuration Protocol", <a href="./rfc2131">RFC 2131</a>,
         March 1997.

   [<a id="ref-12">12</a>]  Sparks, R., "The Session Initiation Protocol (SIP) Refer
         Method", <a href="./rfc3515">RFC 3515</a>, April 2003.

   [<a id="ref-13">13</a>]  Donovan, S. and J. Rosenberg, "Session Timers in the Session
         Initiation Protocol (SIP)", <a href="./rfc4028">RFC 4028</a>, April 2005.

   [<a id="ref-14">14</a>]  Sparks, R., "Problems Identified Associated with the Session
         Initiation Protocol's (SIP) Non-INVITE Transaction", <a href="./rfc4321">RFC 4321</a>,
         January 2006.

   [<a id="ref-15">15</a>]  Fielding,  R., Gettys, J., Mogul, J., Frystyk, H., Masinter,
         L., Leach, P., and T. Berners-Lee, "Hypertext Transfer Protocol
         -- HTTP/1.1", <a href="./rfc2616">RFC 2616</a>, June 1999.

   [<a id="ref-16">16</a>]  Schulzrinne, H., Rao, A., and R. Lanphier, "Real Time Streaming
         Protocol (RTSP)", <a href="./rfc2326">RFC 2326</a>, April 1998.




<span class="grey">Rosenberg &amp; Schulzrinne      Informational                     [Page 20]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-21" ></span>
<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


   [<a id="ref-17">17</a>]  Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,
         "SIP: Session Initiation Protocol", <a href="./rfc2543">RFC 2543</a>, March 1999.

   [<a id="ref-18">18</a>]  Shacham, A., Monsour, B., Pereira, R., and M. Thomas, "IP
         Payload Compression Protocol (IPComp)", <a href="./rfc3173">RFC 3173</a>, September
         2001.

   [<a id="ref-19">19</a>]  Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T.
         Haukka, "Security Mechanism Agreement for the Session
         Initiation Protocol (SIP)", <a href="./rfc3329">RFC 3329</a>, January 2003.

   [<a id="ref-20">20</a>]  Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu,
         Z., and J. Rosenberg, "Signaling Compression (SigComp)", <a href="./rfc3320">RFC</a>
         <a href="./rfc3320">3320</a>, January 2003.

   [<a id="ref-21">21</a>]  Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K.
         Summers, "Session Initiation Protocol (SIP) Basic Call Flow
         Examples", <a href="https://www.rfc-editor.org/bcp/bcp75">BCP 75</a>, <a href="./rfc3665">RFC 3665</a>, December 2003.

   [<a id="ref-22">22</a>]  Rosenberg, J., "A Session Initiation Protocol (SIP) Event
         Package for Registrations", <a href="./rfc3680">RFC 3680</a>, March 2004.

   [<a id="ref-23">23</a>]  Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
         Responses in Session Initiation Protocol (SIP)", <a href="./rfc3262">RFC 3262</a>, June
         2002.

   [<a id="ref-24">24</a>]  Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE
         Method", <a href="./rfc3311">RFC 3311</a>, October 2002.

   [<a id="ref-25">25</a>]  Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs", <a href="https://www.rfc-editor.org/bcp/bcp26">BCP 26</a>, <a href="./rfc2434">RFC 2434</a>, October
         1998.

   [<a id="ref-26">26</a>]  Reynolds, J. and R. Braden, "Instructions to Request for
         Comments (RFC) Authors", Work in Progress, July 2004.

   [<a id="ref-27">27</a>]  Rescorla, E. and IAB, "Writing Protocol Models", <a href="./rfc4101">RFC 4101</a>, June
         2005.

   [<a id="ref-28">28</a>]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", <a href="./rfc3264">RFC 3264</a>, June 2002.










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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


Authors' Addresses

   Jonathan Rosenberg
   Cisco Systems
   600 Lanidex Plaza
   Parsippany, NJ  07054
   US

   Phone: +1 973 952-5000
   EMail: jdrosen@cisco.com
   URI:   <a href="http://www.jdrosen.net">http://www.jdrosen.net</a>


   Henning Schulzrinne
   Columbia University
   M/S 0401
   1214 Amsterdam Ave.
   New York, NY  10027
   US

   EMail: schulzrinne@cs.columbia.edu
   URI:   <a href="http://www.cs.columbia.edu/~hgs">http://www.cs.columbia.edu/~hgs</a>





























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<span class="grey"><a href="./rfc4485">RFC 4485</a>                     SIP Guidelines                     May 2006</span>


Full Copyright Statement

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</pre>