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Versions: (draft-allen-sipping-poc-p-headers) 00 01 02 03 04 05 RFC 4964

SIPPING Working Group                                      A. Allen, Ed.
Internet-Draft                                  Research in Motion (RIM)
Intended status: Experimental                                    J. Holm
Expires: September 5, 2007                                      Ericsson
                                                               T. Hallin
                                                                Motorola
                                                           March 4, 2007


 The P-Answer-State Header Extension to the Session Initiation Protocol
        for the Open Mobile Alliance Push-to-talk over Cellular
            draft-allen-sipping-poc-p-answer-state-header-05

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
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   This Internet-Draft will expire on September 5, 2007.

Copyright Notice

   Copyright (C) The IETF Trust (2007).

Abstract

   This document describes a private Session Initiation Protocol(SIP)
   header (P-header) used by the Open Mobile Alliance (OMA), for Push-
   to-talk over Cellular (PoC) along with its applicability, which is
   limited to the OMA PoC application.  The P-Answer-State header is



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   used for indicating the answering mode of the handset which is
   particular to the PoC application.


Table of Contents

   1.  Overall Applicability  . . . . . . . . . . . . . . . . . . . .  3

   2.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3

   3.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  3

   4.  Background for the Extension . . . . . . . . . . . . . . . . .  4

   5.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  5

   6.  The P-Answer-State Header  . . . . . . . . . . . . . . . . . .  6
     6.1.  Requirements . . . . . . . . . . . . . . . . . . . . . . .  7
     6.2.  Alternatives Considered  . . . . . . . . . . . . . . . . .  8
     6.3.  Applicability Statement for the P-Answer-State Header  . .  9
     6.4.  Usage of the P-Answer-State Header . . . . . . . . . . . . 10
       6.4.1.  Procedures at the UA (Terminal)  . . . . . . . . . . . 11
       6.4.2.  Procedures at the UA (PTT Server)  . . . . . . . . . . 11
       6.4.3.  Procedures at the Proxy Server . . . . . . . . . . . . 13

   7.  Formal Syntax  . . . . . . . . . . . . . . . . . . . . . . . . 14
     7.1.  P-Answer-State Header Syntax . . . . . . . . . . . . . . . 14
     7.2.  Table of the New Header  . . . . . . . . . . . . . . . . . 14

   8.  Example Usage Session Flows  . . . . . . . . . . . . . . . . . 14
     8.1.  Pre-arranged Group Call Using On-demand Session  . . . . . 15
     8.2.  1-1 Call Using Pre-established Session . . . . . . . . . . 20

   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 26

   10. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 27
     10.1. Registration of Header Fields  . . . . . . . . . . . . . . 27

   11. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 28

   12. References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
     12.1. Normative References . . . . . . . . . . . . . . . . . . . 28
     12.2. Informative References . . . . . . . . . . . . . . . . . . 28

   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 29
   Intellectual Property and Copyright Statements . . . . . . . . . . 31





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1.  Overall Applicability

   The SIP extension specified in this document makes certain
   assumptions regarding network topology and the existence of
   transitive trust.  These assumptions are generally NOT APPLICABLE in
   the Internet as a whole.  The mechanism specified here was designed
   to satisfy the requirements specified by the Open Mobile Alliance for
   Push-to-talk over Cellular for which either no general-purpose
   solution was found, where insufficient operational experience was
   available to understand if a general solution is needed, or where a
   more general solution is not yet mature.  For more details about the
   assumptions made about this extension, consult the Applicability
   subsection 6.3.


2.  Introduction

   The Open Mobile Alliance (OMA) (http://www.openmobilealliance.org) is
   specifying the Push-to-talk Over Cellular (PoC) service where SIP is
   the protocol used to establish half duplex media sessions across
   different participants.  This document describes a private extension
   to address specific requirements of the PoC service and may not be
   applicable to the general Internet.

   The PoC service allows a SIP User Agent (UA) (PoC terminal) to
   establish a session to one or more SIP UAs simultaneously, usually
   initiated by the initiating user pushing a button.

   OMA has defined a collection of very stringent requirements in
   support of the PoC service.  In order to provide the user with a
   satisfactory experience the initial session establishment from the
   time the user presses the button to the time they get an indication
   to speak must be minimized.


3.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in [1].

   The terms "PTT Server" (Push-to-talk Server), "Unconfirmed
   Indication", "Unconfirmed Response", "Confirmed Indication" and
   "Confirmed Response" are introduced in this document.

   A "PTT Server" as referred to here is a SIP network server that
   performs the network based functions for the Push to Talk service.
   The PTT Server can act as a SIP Proxy as defined in [2] or a back-to-



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   back UA (B2BUA) as defined in [2] based on the functions it needs to
   perform.  There can be one or more PTT Servers involved in a SIP Push
   to Talk session.

   An "Unconfirmed Indication" as referred to here is an indication that
   the final target UA for the request has yet to be contacted and an
   intermediate SIP node is indicating that it has information that
   hints that the request is likely to be answered by the target UA.

   An "Unconfirmed Response" is a SIP 18x or 2xx response containing an
   "Unconfirmed Indication".

   A "Confirmed Indication" as referred to here is an indication that
   the target UA has accepted the session invitation and is ready to
   receive media.

   A "Confirmed Response" is a SIP 200 (OK) response containing a
   "Confirmed Indication" and has the usual semantics of a SIP 200 (OK)
   response containing an answer (such as a Session Description Protocol
   (SDP) answer).


4.  Background for the Extension

   The PoC terminal could support such hardware capabilities as a
   speaker phone and/or headset and software that provide the capability
   for the user to configure the PoC terminal to accept the session
   invitations immediately and play out the media as soon as it is
   received without requiring the intervention of the called user.  This
   mode of operation is known as Automatic Answer mode.  The user can
   alternatively configure the PoC terminal to first alert the user and
   require the user to manually accept the session invitation before
   media is accepted.  This mode of operation is known as Manual Answer
   mode.  The PoC terminal could support both or only one of these modes
   of operation.  The user can change the Answer Mode (AM) configuration
   of the PoC terminal frequently based on their current circumstances
   and preference,(perhaps because the user is busy, or in a public area
   where she cannot use a speaker phone, etc).

   The OMA PoC Architecture [3] utilizes PTT Servers within the network
   that can perform such roles as a conference focus [10], a real-time
   transport protocol (RTP) translator or a network policy enforcement
   server.  A possible optimization to minimize the delay in the
   providing of the caller with an indication to speak is for the PTT
   server to perform buffering of media packets in order to provide an
   early or "Unconfirmed Indication" back to the caller and allow the
   caller to start speaking before the called PoC terminal has answered.
   An event package and mechanisms for a SIP UA to indicate its current



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   answer mode to a PTT Server in order to enable buffering are defined
   in [11].  In addition, particularly when multiple domains are
   involved in the session, more than one PTT server could be involved
   in the signaling path for the session.  Also the PTT Server that
   performs the buffering might not be the PTT Server that has knowledge
   of the current answer mode of the SIP UA that is the final
   destination for the SIP INVITE request.  A mechanism to allow a
   terminal that acts as a SIP UA or a PTT server that acts as a SIP UA
   to indicate a preference to the final destination SIP User Agent
   Server (UAS) to answer in a particular mode is defined in [12].
   However a mechanism is required for a PTT Server to relay the
   "Unconfirmed Indication" in a response back towards the originating
   SIP User Agent Client (UAC).


5.  Overview

   The purpose of this extension is to support an optimization that
   makes it possible for the network to provide a faster push-to-talk
   experience, through an intermediate SIP agent (PTT Server) providing
   a SIP 200 (OK) response before the called UA does, and a PTT Server
   buffering the media generated by the calling UA for replay to the
   called UA when it answers.  Because of the half duplex nature of the
   call where media bursts are short typically in the order of 10-30
   seconds the additional end to end latency can be tolerated and this
   considerably improves the user experience.  However the PTT Server
   only can do this when there is a high probability the called SIP UA
   is in Automatic Answer mode.  It is likely that PTT Servers near the
   called UA have up-to-date knowledge of the answering mode of the
   called UA, and due to the restricted bandwidth nature of the cellular
   network, they can pass upstream an indication of the called SIP UA's
   answering mode faster than the called UA can deliver an automatically
   generated SIP 200 (OK) response.

   This document proposes a new SIP header field the P-Answer-State
   header field to support an "Unconfirmed Indication".  The new SIP
   header field can be optionally included in a response to a SIP INVITE
   request or in a sipfrag of a response included in a SIP NOTIFY
   request sent as a result of a SIP REFER request that requests a SIP
   INVITE request to be sent.  The header field is used to provide an
   indication from a PTT Server acting as a SIP proxy or back-to-back UA
   that it has information that hints that the terminating UA will
   likely answer automatically.  This provides an "Unconfirmed
   Indication" back towards the inviting SIP UA to transmit media prior
   to receiving a final response from the final destination of the SIP
   INVITE request.  No supported or require headers are needed because
   the sender of the P-Answer-State header field does not depend on the
   receiver to understand the extension and if the extension is not



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   understood the header field is simply ignored by the recipient.  The
   extension is described below.

   Thus, when a PTT Server forwards a SIP INVITE request and knows that
   the called UA is likely to be in Automatic Answer mode, it also
   generates a SIP 183 provisional response with a P-Answer-State header
   field with a parameter of "Unconfirmed" to signal to upstream PTT
   Servers that they can buffer the caller's media.

   A PTT Server that wishes to buffer the caller's media, upon seeing
   the provisional response with a P-Answer-State header field with a
   parameter of "Unconfirmed" absorbs it and generates a SIP 200 (OK)
   response for the caller's SIP UA with an appropriate answer.

   When the called UA generates a SIP 200 (OK) response, the PTT Server
   that generated the provisional response with a P-Answer-State header
   field with a parameter "Unconfirmed" adds to the SIP 200 (OK)
   response a P-Answer-State header field with a parameter of
   "Confirmed".  The SIP 200 (OK) response is absorbed by the PTT Server
   that is buffering the caller's media, as it has already generated a
   SIP 200 (OK) response.  The buffering PTT Server then starts playing
   out the buffered media.


6.  The P-Answer-State Header

   The purpose of the P-Answer-State header field is to provide an
   indication from a PTT Server acting as a SIP proxy or back-to-back UA
   that is has information that hints that the terminating UA identified
   in the Request-URI of the request will likely answer automatically.
   Thus enabling the PTT Server to provide an "Unconfirmed Indication"
   back towards the inviting SIP UA permitting it to transmit media
   prior to receiving a final response from the final destination of the
   SIP INVITE request.  If a provisional response contains the P-Answer-
   State header field with the value "Unconfirmed" and does not contain
   an answer then a receiving PTT Server can send a SIP 200 (OK)
   response containing an answer and a P-Answer-State header field with
   the value "Unconfirmed" if the PTT Server is willing to perform media
   buffering.  If the response containing the P-Answer-State header
   field with the value "Unconfirmed" also contains an answer the PTT
   Server that included the P-Answer-State header field and answer in
   the response is also indicating that it is willing to buffer the
   media until a final "Confirmed Indication" is received.

   The P-Answer-State header field can be included in a provisional or
   final response to a SIP INVITE request or in the sipfrag of a SIP
   NOTIFY request sent as a result of a SIP REFER request to send a SIP
   INVITE request.  If the P-Answer-State header field with value



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   "Unconfirmed" is included in a provisional response that contains an
   answer the PTT Server is leaving the decision where to do buffering
   to other PTT Servers upstream and will forward upstream a "Confirmed
   indication" in a SIP 200 (OK) response when the final response is
   received from the destination UA.

   NOTE It is not intended that multiple PTT servers perform buffering
   serially.  If a PTT server includes an answer along with P-Answer-
   State header field with the value "Unconfirmed" in a provisional
   response then a receiving PTT Server can determine whether it buffers
   the media or whether to forward the media and allow the downstrean
   PTT Server that sent the "Unconfirmed Indication" to buffer the
   media.  It is intended that if a PTT Server buffers media it does so
   until a final "Confirmed Indication" is received and therefore serial
   buffering by multiple PTT Servers does not take place

   The P-Answer-State header is only included in a provisional response
   when the node that sends the response has knowledge that there is a
   PTT Server that acts as a B2BUA that understands this extension in
   the signaling path between itself and the originating UAC that will
   only pass the header field on in either a SIP 200 (OK) response or in
   the sipfrag as defined in [4] of a SIP NOTIFY request as defined in
   [5] sent as a result of a SIP REFER request as defined in [6].  Such
   a situation only occurs with specific network topologies which is
   another reason why use of this header field is not relevant to the
   general internet.  The originating UAC will only receive the
   P-Answer-state header field in a SIP 200 (OK) response or in the
   sipfrag of a SIP NOTIFY request.

   Provisional responses containing the P-Answer-State header field can
   be sent reliably using the mechanism defined in [13] but this is not
   required.  This is a performance optimization and the impact of a
   provisional response sent unreliably failing to arrive is simply that
   buffering does not take place.  However, if the provisional responses
   are sent reliably and the provisional response fails to arrive the
   time taken for the provisional response sender to timeout on the
   receipt of a SIP PRACK request is likely to be such that by the time
   the provisional response has been resent the "Confirmed Response"
   could have already been received.  Because when provisional responses
   that contain an answer are sent reliably, the 200 (OK) response for
   the SIP INVITE request cannot be sent before the SIP PRACK request is
   received, sending provisional responses reliably could potentially
   delay the sending of the "Confirmed Response".

6.1.  Requirements

   The OMA PoC service has initial setup performance requirements that
   can be met by a PTT Server acting as a B2BUA spooling media from the



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   inviting PoC subscriber until one or more invited PoC subscribers
   have accepted the session.  The specific requirements are

   REQ-1:  An intermediate server MAY spool media from the inviting SIP
      UA until one or more invited PoC SIP UAs has accepted the
      invitation.

   REQ-2:  An intermediate server that is capable of spooling media MAY
      accept a SIP INVITE request from an inviting SIP UAC even if no
      invited SIP UAS has accepted the SIP INVITE request if it has a
      hint that the invited SIP UAC is likely to accept the request
      without requiring user intervention.

   REQ-3:  An intermediate server or proxy that is incapable of spooling
      media or does not wish to, but has a hint that the invited SIP UAC
      is likely to automatically accept the session invitation MUST be
      able to indicate back to another intermediate server that can
      spool media that it has some hint that the invited UAC is likely
      to automatically accept the session invitation.

   REQ-4:  An intermediate server that is willing to spool media from
      the inviting SIP UA until one or more invited SIP UAs have
      accepted the SIP INVITE request SHOULD indicate that it is
      spooling media to the inviting SIP UAC.


6.2.  Alternatives Considered

   In order to meet REQ-3, a PTT Server needs to receive an indication
   back that the invited SIP UA is likely to accept the SIP INVITE
   request without requiring user intervention.  In this case, the PTT
   Server that has a hint that the invited SIP UAC is likely to accept
   the request can include an answer state indication in the SIP 183
   (Session Progress) response or SIP 200 (OK) response.

   A number of alternatives were considered for the PTT Server to inform
   another PTT Server or the inviting SIP UAC of the invited PoC SIP UAs
   answer mode settings.

   One proposal was to create a unique reason-phrase in the SIP 183
   response and SIP 200 (OK) response.  This was rejected because the
   reason phrases are normally intended for human readers and not meant
   to be parsed by servers for special syntactic and semantic meaning.

   Another proposal was to use a Reason header [14] in the SIP 183
   response and SIP 200 (OK) response.  This was rejected because this
   would be inconsistent with the intended use of the reason header and
   its usage is not defined for these response codes and would have



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   required creating and registering a new protocol identifier.

   Another proposal was to use a feature-tag in the returned Contact
   header as defined in [15].  This was rejected because it was not a
   different feature, but is an attribute of the session and can be
   applied to many different features.

   Another proposal was to use a new SDP attribute.  The choice of an
   SDP parameter was rejected because the answer state applies to the
   session and not to a media stream.

   The P-Answer-State header was chosen to give additional information
   about the state of the SIP session progress and acceptance.  Even
   though the UAC sees that its offer has been answered and accepted,
   the header lets the UAC know whether the invited PoC subscriber has
   accepted the SIP INVITE request or just an intermediary has done the
   acceptance.

6.3.  Applicability Statement for the P-Answer-State Header

   The P-Answer-State header is applicable in the following
   circumstances:

      o In networks where there are UAs that engage in half-duplex
      communication where there is not the possibility for the invited
      user to verbally acknowledge the answering of the session as is
      normal in full duplex communication;

      o Where the invited UA can automatically accept the session
      without user intervention;

      o The network also contains intermediate network SIP servers that
      are trusted;

      o The intermediate network SIP servers have knowledge of the
      current answer mode setting of the terminating UAS; and,

      o The intermediate network SIP servers have knowledge of the media
      types and codecs likely to be accepted by the terminating UAS;
      and,

      o The intermediate network SIP servers can provide buffering of
      the media in order to reduce the time for the inviting user to
      send media.







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      o The intermediate network SIP servers assume knowledge of the
      network topology and the existence of similar intermediate network
      SIP servers in the signaling path.

   Such configurations are generally not applicable to the internet as a
   whole where such trust relationships do not exist.

   In addition security issues have only been considered for networks
   which are trusted and use hop by hop security mechanisms with
   transitive trust and security issues with usage of this mechanism in
   the general internet have not been evaluated.

6.4.  Usage of the P-Answer-State Header

   A UAS B2BUA or proxy MAY include a P-Answer-State header field in any
   SIP 18x or 2xx response that does not contain an offer, sent in
   response to an offer contained in a SIP INVITE request as specified
   in [7].  Typically the P-Answer-State header field is included in
   either a SIP 183 Session Progress or a SIP 200 (OK) response.  A UA
   that receives a SIP REFER request to send a SIP INVITE request MAY
   also include a P-Answer-State header field in the sipfrag of a
   response included in a SIP NOTIFY request it sends as a result of the
   implicit subscription created by the SIP REFER request.

   When the P-Answer-State header field contains the parameter
   "Unconfirmed" the UAC or proxy is indicating that it has information
   that hints that the final destination UAS for the SIP INVITE request
   is likely to automatically accept the session but that this is
   unconfirmed and it is possible that the final destination UAS will
   first alert the user and require manual acceptance of the session or
   not accept the session request.  When the P-Answer-State header field
   contains the parameter "Confirmed" the UAC or proxy is indicating
   that the destination UAS has accepted the session and is ready to
   receive media.  The parameter value of "Confirmed" has the usual
   semantics of a SIP 200 (OK) response containing an answer and is
   included for completeness.  A parameter value of "Confirmed" is only
   included in a SIP 200 (OK) response or in the sipfrag of a 200 (OK)
   contained in the body of a SIP NOTIFY request.

   A received SIP 18x response without a P-Answer-State header field
   SHOULD NOT be treated as an "Unconfirmed Response".  A SIP 18x
   response containing a P-Answer-State header field containing the
   parameter "Confirmed" MUST NOT be treated as a "Confirmed Response"
   because this in an invalid condition.

   A SIP 200 (OK) response without a P-Answer-State Header field MUST be
   treated as a "Confirmed Response".




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6.4.1.  Procedures at the UA (Terminal)

   A UAC (terminal) that receives an "Unconfirmed Response" containing
   an answer MAY send media as specified in [7], however there is no
   guarantee that the media will be received by the final recipient.

   How a UAC confirms whether the media was or was not received by the
   final destination when it has received a SIP 2xx response containing
   an "Unconfirmed Indication" is application specific and outside of
   the scope of this document.  If the application is a conference then
   the mechanism specified in [7] could be used to determine that the
   invited user joined.  Alternatively a SIP BYE request could be
   received or the media could be placed on hold if the final
   destination UAS does not accept the session.

   A UAC (terminal) that receives in response to a SIP REFER request, a
   SIP NOTIFY request containing an "Unconfirmed Response" in a sipfrag
   in the body of the SIP NOTIFY request related to a dialog for which
   there has been a successful offer-answer exchange according to [5]
   MAY send media, however there is no guarantee that the media will be
   received by the final recipient that was indicated in the Refer-To
   header in the original SIP REFER request.  The dialog could be
   related either because the SIP REFER request was sent on the same
   dialog or because the SIP REFER request contained a Target-Dialog
   header as defined in [16] that identified the dialog.

   A UAC (terminal) that receives an "Unconfirmed Response" that does
   not contain an answer MAY buffer media until it receives another
   "Unconfirmed Response" containing an answer or a "Confirmed
   Response".

   There are no P-Answer-State procedures for a terminal acting in the
   UAS role.

6.4.2.  Procedures at the UA (PTT Server)

   A PTT Server that receives a SIP INVITE request at the UAS part of
   its back-to-back UA MAY include in any SIP 18x or 2xx response that
   does not contain an offer, a P-Answer-State header field with the
   parameter "Unconfirmed" in the response if it has not yet received a
   "Confirmed Response" from the final destination UA and it has
   information that hints that that the final destination UA for the SIP
   INVITE request is likely to automatically accept the session.

   A PTT Server that receives a SIP 18x response to a SIP INVITE request
   containing a P-Answer-State header field with the parameter
   "Unconfirmed" at the UAC part of its back-to-back UA MAY include the
   P-Answer-State header field with the parameter "Unconfirmed" in a SIP



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   2xx response the UAS part of its back-to-back UA sends as a result of
   receiving that response.  Otherwise a PTT Server that receives a SIP
   18x or 2xx response to a SIP INVITE request containing a P-Answer-
   State header field at the UAC part of its back-to-back UA SHOULD
   include the P-Answer-State header field unmodified in the SIP 18x or
   2xx response the UAS part of its back-to-back UA sends as a result of
   receiving that response.  If the response sent by the UAS part of its
   back-to-back UA is a SIP 18x response then the P-Answer-State header
   field included in the response MUST contain a parameter of
   "Unconfirmed".

   The UAS part of the back-to-back UA of a PTT Server MAY include an
   answer in the "Unconfirmed Response" it sends even if the
   "Unconfirmed Response" received by the UAC part of the back-to-back
   UA did not contain an answer.

   If a PTT Server that receives at the UAC part of its back-to-back UA
   a "Confirmed Response" then the UAS part of its back-to-back UA MAY
   include in the forwarded response a P-Answer-State header field with
   the parameter "Confirmed".  If the UAS part of its back-to-back UA
   previously sent an "Unconfirmed Response" as part of this dialog the
   UAS part of its back-to-back UA SHOULD include in the forwarded
   "Confirmed Response" a P-Answer-State header field with the parameter
   "Confirmed".

   If the UAS part of the back-to-back UA of a PTT Server, includes an
   answer in a response along with a P-Answer-State header field with
   the parameter "Unconfirmed" then the UAS part of its back-to-back UA
   needs to be ready to receive media as specified in [7] and MAY buffer
   any media it receives until it receives a "Confirmed Response" from
   the final destination UA or until its buffer is full.

   A UAS part of the back-to-back UA of a PTT Server that receives a SIP
   REFER request to send a SIP INVITE request to another UA as specified
   in [6], MAY generate a sipfrag of a SIP 200 (OK) response containing
   a P-Answer-State header field with the parameter "Unconfirmed" prior
   to the UAC part of its back-to-back UA receiving a response to the
   SIP INVITE request, if it has information that hints that the final
   destination UA for the SIP INVITE request is likely to automatically
   accept the session.

   If the UAC part of a back-to-back UA of a PTT Server sent a SIP
   INVITE request as a result of receiving a SIP REFER Request, receives
   a SIP 18x or 2xx response containing a P-Answer-State header field at
   the UAC part of its back-to-back UA, then the UAS part of its back-
   to-back UA SHOULD include the P-Answer-State header field and its
   parameters from that response unmodified in the sipfrag of the
   response contained in a SIP NOTIFY request that the UAS part of its



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   back-to-back UA sends in response to the SIP REFER request.  If the
   sipfrag of the response sent in the SIP NOTIFY request is a SIP 18x
   response then the P-Answer-State header field included in the sipfrag
   of the response MUST contain a parameter of "Unconfirmed".  If the
   UAC part of its back-to-back UA receives a "Confirmed Response" that
   does not contain a P-Answer-State header field then the UAS part of
   its back-to-back UA MAY include a P-Answer-State header field with
   the parameter "Confirmed" in the sipfrag of the response contained in
   a SIP NOTIFY request sent in response to the SIP REFER request.

   A PTT Server that's UAS part of its back-to-back UA previously sent a
   SIP NOTIFY request containing a P-Answer-State header field with the
   parameter "Unconfirmed" in the sipfrag of a response included in the
   SIP NOTIFY request, that subsequently receives at the UAC part of its
   back-to-back UA a "Confirmed Response" to the SIP INVITE request sent
   as a result of the SIP REFER request SHOULD include a P-Answer-State
   header field with the parameter "Confirmed" in the sipfrag of the
   response included in the subsequent SIP NOTIFY request that the UAS
   part of its back-to-back UA sends as a result of receiving the
   "Confirmed Response".

   If the SIP REFER request related to an existing dialog established by
   a SIP INVITE request for which there has been a successful offer-
   answer exchange the UAS part of its back-to-back UA MUST be ready to
   receive media as specified in [7] and MAY buffer any media it
   receives until the UAC part of its back-to-back UA receives a
   "Confirmed Response" from the final destination UA or until its
   buffer is full.  The dialog could be related either because the SIP
   REFER request was sent on the same dialog or because the SIP REFER
   request contained a Target-Dialog header as defined in [16] that
   identified the dialog.

   A PTT Server that buffers media SHOULD be prepared for the
   possibility of not receiving a "Confirmed Response" and SHOULD
   release the session if a "Confirmed Response" is not received before
   the buffer overflows.

6.4.3.  Procedures at the Proxy Server

   SIP proxy servers do not need to understand the semantics of the
   P-Answer-State header field.  As part of the regular SIP rules for
   unknown headers, a proxy will forward unknown headers.

   A PTT Server that acts as a proxy MAY include a P-Answer-State header
   field with the parameter "Unconfirmed" in a SIP 18x response that it
   originates compliant with [2] if it has information that hints that
   that the final destination UA for the SIP INVITE request is likely to
   automatically accept the session.



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   A PTT Server that acts as a proxy MAY add a P-Answer-State header
   field with the parameter "Confirmed" to a "Confirmed Response".


7.  Formal Syntax

   The mechanisms specified in this document is described in both prose
   and an augmented Backus-Naur Form (BNF) defined in [8].  Further,
   several BNF definitions are inherited from SIP and are not repeated
   here.  Implementers need to be familiar with the notation and
   contents of SIP [2] and [8] to understand this document.

7.1.  P-Answer-State Header Syntax

   The syntax of the P-Answer-State header is described as follows:

      P-Answer-State = "P-Answer-State" HCOLON answer-type
                       *(SEMI generic-param)
      answer-type = "Confirmed" / "Unconfirmed" / token

7.2.  Table of the New Header

   Table 1 provides the additional table entries for the P-Answer-State
   header needed to extend Table 2 in SIP [2], section 7.1 of the SIP-
   specific event notification [5] tables 1 and 2 in the SIP INFO method
   [17], tables 1 and 2 in Reliability of provisional responses in SIP
   [13], tables 1 and 2 in the SIP UPDATE method [18], tables 1 and 2 in
   the SIP extension for Instant Messaging [19], table 1 in the SIP
   REFER method [6], and table 2 in the SIP PUBLISH method [20]:

         Header field          where  proxy  ACK BYE CAN INV OPT REG SUB
         _______________________________________________________________
         P-Answer-State      18x,2xx    ar    -   -   -   o   -   -   -

         Header field                        NOT PRA INF UPD MSG REF PUB
         _______________________________________________________________
         P-Answer-State          R            -   -   -   -   -   -   -

                                 Figure 1


8.  Example Usage Session Flows

   For simplicity some details such as intermediate proxies and SIP 100
   Trying responses are not shown in the following example flows.






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8.1.  Pre-arranged Group Call Using On-demand Session

   The following flow shows Alice making a Pre-arranged Group Call using
   a Conference URI which has Bob on the member list.  The session
   initiation uses the On-demand Session establishment mechanism where a
   SIP INVITE request containing an SDP offer is sent by Alices's
   terminal when Alice pushes her push to talk button.

   In this example Alice's PTT Server acts a Call Stateful SIP Proxy and
   Bob's PTT Server which is aware that the current Answer Mode setting
   of Bob's terminal is set to Auto Answer acts as a B2BUA.

   For simplicity the invitations by the Conference Focus to the other
   members of the group are not shown in this example.

      Alice's        Alices's     Conference     Bob's          Bob's
      Terminal      PTT Server       focus      PTT Server    Terminal
         |              |              |             |              |
         |--(1)INVITE-->|              |             |              |
         |              |--(2)INVITE-->|             |              |
         |              |              |--(3)INVITE->|              |
         |              |              |             |--(4)INVITE-->|
         |              |              |<--(5)183----|              |
         |              |<---(6)200----|             |              |
         |<---(7)200----|              |             |              |
         |----(8)ACK--->|              |             |              |
         |              |---(9)ACK---->|             |              |
         |              |              |             |              |
         |=====Early Media Session====>|             |              |
         |              |            MEDIA           |              |
         |              |           BUFFERING        |              |
         |              |              |             |<---(10)200---|
         |              |              |             |---(11)ACK--->|
         |              |              |<--(12)200---|              |
         |              |              |--(13)ACK--->|              |
         |              |              |             |              |
         |              |              |========Media Session======>|
         |              |              |             |              |
         |              |              |             |              |

                                 Figure 2

   1 INVITE Alice -> Alices's PTT Server

   INVITE sip:FriendsOfAlice@example.org SIP/2.0
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   Max-Forwards: 70
   To: "Alice's Friends" <sip:FriendsOfAlice@example.org>



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   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:alice@pc33.example.org>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   2 INVITE Alice's PTT Server -> Conference Focus

   INVITE sip:FriendsOfAlice@example.org SIP/2.0
   Via: SIP/2.0/UDP
        AlicesPTTServer.example.org;branch=z9hG4bK77ef4c2312983.1
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   Record-Route: <sip:AlicesPTTServer.example.org>
   Max-Forwards: 69
   To: "Alice's Friends" <sip:FriendsOfAlice@example.org>
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:alice@pc33.example.org>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   The Conference Focus explodes the Conference URI and Invites Bob

   3 INVITE Conference Focus -> Bob's PTT Server

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>
   From: "Alice's Friends"
   <sip:FriendsOfAlice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   CSeq: 301166605 INVITE
   Contact: <sip:AlicesConferenceFocus.example.org>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   4 INVITE Bob's PTT Server -> Bob




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   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/UDP
        BobsPTTServer.example.com;branch=z9hG4bKa27bc93
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>
   From: "Alice's Friends"
   <sip:FriendsOfAlice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 INVITE
   Contact: <sip:BobsPTTServer.example.com>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   5 183 (Session Progress) Bob's PTT Server -> Conference Focus

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   To: "Bob" <sip:bob@example.com>;tag=a6c85cf
   From: "Alice's Friends"
   <sip:FriendsOfAlice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   Contact: <sip:BobsPTTServer.example.com>
   CSeq: 301166605 INVITE
   P-Answer-State: Unconfirmed
   Content-Length: 0

   6 200 (OK) Conference Focus -> Alice's PTT Server

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        AlicesPTTServer.example.org;branch=z9hG4bK77ef4c2312983.1
   Via: SIP/2.0/UDP
        pc33.example.org;branch=z9hG4bKnashds8
   Record-Route: <sip:AlicesPTTServer.example.org>
   To: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=c70ef99
   From: "Alice"
        <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:AlicesConferenceFocus.example.org>
   P-Answer-State: Unconfirmed
   Content-Type: application/sdp
   Content-Length: 131




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   (SDP not shown)

   7 200 (OK) Alice's PTT Server -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   Record-Route: <sip:AlicesPTTServer.example.org>
   To: "Alice's Friends" <sip:FriendsOfAlice@example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:AlicesConferenceFocus.example.org>
   P-Answer-State: Unconfirmed
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   8 ACK Alice -> Alice's PTT Server

   ACK sip:AlicesConferenceFocus.example.org SIP/2.0
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds9
   Route: <sip:AlicesPTTServer.example.org>
   Max-Forwards: 70
   To: "Alice's Friends" <sip:FriendsOfAlice@example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 ACK
   Content-Length: 0

   9 ACK Alice's PTT Server -> Conference Focus

   ACK sip:AlicesConferenceFocus.example.org SIP/2.0
   Via: SIP/2.0/UDP
        AlicesPTTServer.example.org;branch=z9hG4bK77ef4c2312983.1
   Via: SIP/2.0/UDP
        pc33.example.org;branch=z9hG4bKnashds9
   Max-Forwards: 69
   To: "Alice's Friends" <sip:FriendsOfAlice@example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 ACK
   Content-Length: 0

   The early half duplex media session between Alice and the Conference
   Focus is now established and the Conference Focus buffers the media
   it receives from Alice.




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   10 200 (OK) Bob -> Bob's PTT Server

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        BobsPTTServer.example.com;branch=z9hG4bKa27bc93
   To: "Bob" <sip:bob@example.com>;tag=d28119a
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 INVITE
   Contact: <sip:bob@192.0.2.4>
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   11 ACK Bob's PTT Server -> Bob

   ACK sip:bob@192.0.2.4 SIP/2.0
   Via: SIP/2.0/UDP BobsPTTServer.example.com;branch=z9hG4bKa27bc93
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>;tag=d28119a
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 ACK
   Content-Length: 0

   12 200 (OK) Bob's PTT Server -> Conference Focus

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   To: "Bob" <sip:bob@example.com>;tag=a6670811
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   Contact: <sip:BobsPTTServer.example.com>
   CSeq: 301166605 INVITE
   P-Answer-State: Confirmed
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   13 ACK Conference Focus -> Bob's PTT Server

   ACK sip:BobsPTTServer.example.com SIP/2.0



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   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   Max-Forwards: 70
   To: "Bob"
        <sip:bob@example.com>;tag=a6670811
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   CSeq: 301166605 ACK
   Content-Length: 0

   The media session between Alice and Bob is now established and the
   Conference Focus forwards the buffered media to Bob.

8.2.  1-1 Call Using Pre-established Session

   The following flow shows Alice making a 1-1 Call to Bob using a pre-
   established session.  A pre-established session is where a dialog is
   established with Alices's PTT Server using a SIP INVITE SDP offer
   answer exchange to pre-negotiate the codecs and other media
   Parameters to be used for media sessions ahead of Alice initiating a
   Communication.  When Alice initiates a communication to Bob a SIP
   REFER request is used to request Alice's PTT Server to send a SIP
   INVITE request to Bob. In this example Bob's Terminal does not use
   the Pre-established Session mechanism.

   In this example Alice's PTT Server acts a B2BUA and also performs the
   Conference Focus function.  Bob's PTT Server which is aware that the
   current Answer Mode setting of Bob's terminal is set to Auto Answer
   acts as a B2BUA.





















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      Alice's                Alice's               Bob's          Bob's
      Terminal             PTT Server /          PTT Server     Terminal
                        Conference Focus
         |                       |                  |                |
         |-----(1)INVITE-- ----->|                  |                |
         |<-----(2)200-----------|                  |                |
         |-------(3)ACK--------->|                  |                |
         |                       |                  |                |
         |                       |                  |                |
         |                       |                  |                |
         |----(4)REFER---------->|                  |                |
         |<-----(5)202-----------|                  |                |
         |                       |----(6)INVITE---->|                |
         |                       |                  |--(7)INVITE---->|
         |                       |                  |                |
         |                       |<----(8)183-------|                |
         |<---(9)NOTIFY----------|                  |                |
         |-----(10)200---------->|                  |                |
         |                       |                  |                |
         |=Early Media Session==>|                  |                |
         |                     MEDIA                |                |
         |                   BUFFERING              |                |
         |                       |                  |<---(11)200-----|
         |                       |                  |---(12)ACK----->|
         |                       |<----(13)200------|                |
         |                       |-----(14)ACK----->|                |
         |                       |===========Media Session==========>|
         |                       |                  |                |
         |<---(15)NOTIFY---------|                  |                |
         |-----(16)200---------->|                  |                |
         |                       |                  |                |

                                 Figure 3

   1 INVITE Alice -> Alices's PTT Server

   INVITE sip:AlicesConferenceFactoryURI.example.org SIP/2.0
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   Max-Forwards: 70
   To: <sip:AlicesConferenceFactoryURI.example.org>
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:alice@pc33.example.org>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)



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   2 200 (OK) Alice's PTT Server -> Alice

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 INVITE
   Contact: <sip:AlicesPre-establishedSession@
   AlicesPTTServer.example.org>
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   3 ACK Alice -> Alice's PTT Server

   ACK sip:AlicesPre-establishedSession@AlicesPTTServer.example.org
        SIP/2.0
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds9
   Max-Forwards: 70
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314159 ACK
   Content-Length: 0

   Alices's terminal has established a Pre-established Session with
   Alice's PTT Server.  All the media parameters are pre-negotiated for
   use at communication time.

   Alice initiates a Communication to Bob

   4 REFER Alice -> Alices's PTT Server

   REFER sip:AlicesPre-establishedSession@AlicesPTTServer.example.org
        SIP/2.0
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   Max-Forwards: 70
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314160 REFER
   Refer-To: "Bob" <sip:bob@example.com>
   Contact: <sip:alice@pc33.example.org>

   5 202 (ACCEPTED) Alice's PTT Server -> Alice




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   SIP/2.0 202 ACCEPTED
   Via: SIP/2.0/UDP pc33.example.org;branch=z9hG4bKnashds8
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314160 REFER
   Contact: <sip:AlicesPre-establishedSession@
   AlicesPTTServer.example.org>

   6 INVITE Conference Focus -> Bob's PTT Server

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bk4721d8
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>
   From: "Alice" <sip:Alice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   CSeq: 301166605 INVITE
   Contact: <sip:AlicesConferenceFocus.example.org>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   7 INVITE Bob's PTT Server -> Bob

   INVITE sip:bob@example.com SIP/2.0
   Via: SIP/2.0/UDP
        BobsPTTServer.example.com;branch=z9hG4bKa27bc93
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>
   From: "Alice" <sip:Alice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 INVITE
   Contact: <sip:BobsPTTServer.example.com>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)

   8 183 (Session Progress) Bob's PTT Server -> Conference Focus

   SIP/2.0 183 Session Progress
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   To: "Bob" <sip:bob@example.com>;tag=a6c85cf
   From: "Alice" <sip:Alice@example.org>;tag=2178309898



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   Call-ID: e60a4c784b6716
   Contact: <sip:BobsPTTServer.example.com>
   CSeq: 301166605 INVITE
   P-Answer-State: Unconfirmed
   Content-Length: 0

   9 NOTIFY Alices's PTT Server -> Alice

   NOTIFY sip:alice@pc33.example.org SIP/2.0
   Via: SIP/2.0/UDP
        AlicesPre-establishedSession@AlicesPTTServer.example.org;
        branch=z9hG4bKnashds8
   Max-Forwards: 70
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314161 NOTIFY
   Contact:
        <sip:AlicesPre-establishedSession@AlicesPTTServer.example.org>
   Event: refer
   Subscription-State: Active;Expires=60
   Content-Type: message/sipfrag;version=2.0
   Content-Length: 99

   SIP/2.0 183 Session Progress
   To: "Bob" <sip:bob@example.com>;tag=d28119a
   P-Answer-State: Unconfirmed

   10 200 (OK) Alice -> Alice's PTT Server

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        AlicesPre-establishedSession@AlicesPTTServer.example.org;
        branch=z9hG4bKnashds8
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314161 NOTIFY

   The early half duplex media session between Alice and the Conference
   Focus is now established and the Conference Focus buffers the media
   it receives from Alice.

   11 200 (OK) Bob -> Bob's PTT Server

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        BobsPTTServer.example.com;branch=z9hG4bK927bc93



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   To: "Bob" <sip:bob@example.com>;tag=d28119a
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 INVITE
   Contact: <sip:bob@192.0.2.4>
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   12 ACK Bob's PTT Server -> Bob

   ACK sip:bob@192.0.2.4 SIP/2.0
   Via: SIP/2.0/UDP BobsPTTServer.example.com;branch=z9hG4bK927bc93
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>;tag=d28119a
   From: "Alice" <sip:Alice@example.org>;tag=781299330
   Call-ID: 6eb4c66a847710
   CSeq: 478209 ACK
   Content-Length: 0

   F13 200 (OK) Bob's PTT Server -> Conference Focus

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   To: "Bob" <sip:bob@example.com>;tag=a6670811
   From: "Alice's Friends"
        <sip:FriendsOfAlice@example.org>;tag=2178309898
   Call-ID: e60a4c784b6716
   Contact: <sip:BobsPTTServer.example.com>
   CSeq: 301166605 INVITE
   P-Answer-State: Confirmed
   Content-Type: application/sdp
   Content-Length: 131

   (SDP not shown)

   14 ACK Conference Focus -> Bob's PTT Server

   ACK sip:BobsPTTServer.example.com SIP/2.0
   Via: SIP/2.0/UDP
        AlicesConferenceFocus.example.org;branch=z9hG4bK4721d8
   Max-Forwards: 70
   To: "Bob" <sip:bob@example.com>;tag=a6670811
   From: "Alice" <sip:Alice@example.org>;tag=1928301774
   Call-ID: e60a4c784b6716



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   CSeq: 301166605 ACK
   Content-Length: 0

   The media session between Alice and Bob is now established and the
   Conference Focus forwards the buffered media to Bob.

   15 NOTIFY Alices's PTT Server -> Alice

   NOTIFY sip:alice@pc33.example.org SIP/2.0
   Via: SIP/2.0/UDP
        AlicesPre-establishedSession@AlicesPTTServer.example.org;
        branch=z9hG4bKnashds8
   Max-Forwards: 70
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314162 NOTIFY
   Contact: <sip:AlicesPre-establishedSession@
   AlicesPTTServer.example.org>
   Event: refer
   Subscription-State: Active;Expires=60
   Content-Type: message/sipfrag;version=2.0
   Content-Length: 83

   SIP/2.0 200 OK
   To: "Bob" <sip:bob@example.com>;tag=d28119a
   P-Answer-State: Confirmed

   16 200 (OK) Alice -> Alice's PTTServer

   SIP/2.0 200 OK
   Via: SIP/2.0/UDP
        AlicesPre-establishedSession@AlicesPTTServer.example.org;
        branch=z9hG4bKnashds8
   To: <sip:AlicesConferenceFactoryURI.example.org>;tag=c70ef99
   From: "Alice" <sip:alice@example.org>;tag=1928301774
   Call-ID: a84b4c76e66710
   CSeq: 314162 NOTIFY


9.  Security Considerations

   The information returned in the P-Answer-State header is not viewed
   as particularly sensitive.  Rather, it is informational in nature,
   providing an indication to the UAC that delivery of any media sent as
   a result of an answer in this response is not guaranteed.  An
   eavesdropper cannot gain any useful information by obtaining the
   contents of this header.



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   End-to-end protection is not appropriate because the P-Answer-State
   header is used and added by proxies and intermediate UAs.  As a
   result, a "malicious" proxy between the UAs or attackers on the
   signaling path could add or remove the header or modify the contents
   of the header value.  This attack either denies the caller the
   knowledge that the callee has yet to be contacted or falsely
   indicates that the callee has yet to be contacted when they have
   already answered.  The falsely indicating that the callee has yet to
   be contacted when they have already answered attack could result in
   the caller deciding not transmit media because they do not wish to
   have their media stored by an intermediary even though in reality the
   callee has answered.  The denying the callee the additional knowledge
   that the callee has yet to be contacted attack does not appear to be
   a significant concern since this is the same as the situation when a
   B2BUA sends a 200 (OK) before the callee has answered without the use
   of this extension.

   It is therefore necessary to protect the messages between proxies and
   implementation SHOULD use a transport that provides integrity and
   confidentially between the signaling hops.  The Transport Layer
   Security (TLS) [9] based signaling in SIP can be used to provide this
   protection.

   Security issues have only been considered for networks which are
   trusted and use hop by hop security mechanisms with transitive trust
   and security issues with usage of this mechanism in the general
   internet have not been evaluated.


10.  IANA Considerations

10.1.  Registration of Header Fields

   This document defines a private SIP extension header field (beginning
   with the prefix "P-" ) based on the registration procedures defined
   in RFC 3427 [21].

    The following rows shall be added to the "Header Fields" section of
                        the SIP parameter registry:

               +----------------+--------------+-----------+
               | Header Name    | Compact Form | Reference |
               +----------------+--------------+-----------+
               | P-Answer-State |              | [RFCXXXX] |
               +----------------+--------------+-----------+

   Editor Note: [RFCXXXX] should be replaced with the designation of
   this document.



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11.  Acknowledgements

   The authors would like to thank Jon Peterson, Cullen Jennings, Jeroen
   van Bemmel, Paul Kyzivat, Dale Worley, Dean Willis, Rohan Mahay,
   Christian Schmidt, Mike Hammer, and Miguel Garcia-Martin for their
   comments that contributed to the progression of this work.  The
   authors would also like to thank the OMA POC Working Group members
   for their support of this document and in particular Tom Hiller for
   presenting the concept of the P-Answer-State header to SIPPING at
   IETF#62.


12.  References

12.1.  Normative References

   [1]   Bradner, S., "Key words for use in RFCs to Indicate Requirement
         Levels", BCP 14, RFC 2119, March 1997.

   [2]   Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
         Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
         Session Initiation Protocol", RFC 3261, June 2002.

   [3]   OMA, "Push to talk over Cellular - Architecture", OMA-AD-
         PoC V1_0_1, 20061128 A, November 2006.

   [4]   Sparks, R., "Internet Media Type message/sipfrag", RFC 3420,
         November 2002.

   [5]   Roach, A., "Session Initiation Protocol (SIP)-Specific Event
         Notification", RFC 3265, June 2002.

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

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

   [8]   Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
         Specifications: ABNF", RFC 4234, October 2005.

   [9]   Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS)
         Protocol Version 1.1", RFC 4346, April 2006.

12.2.  Informative References

   [10]  Rosenberg, J., "A Framework for Conferencing with the Session
         Initiation Protocol (SIP)", RFC 4353, February 2006.



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   [11]  Garcia-Martin, M., "A Session Initiation Protocol (SIP) Event
         Package and Data Format for Various Settings in Support for the
         Push-to-Talk over Cellular (PoC) Service", RFC 4354,
         January 2006.

   [12]  Willis, D. and A. Allen, "Requesting Answering Modes for the
         Session Initiation Protocol (SIP)",
         draft-ietf-sip-answermode-01 (work in progress), May 2006.

   [13]  Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional
         Responses in Session Initiation Protocol (SIP)", RFC 3262,
         June 2002.

   [14]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason Header
         Field for the Session Initiation Protocol (SIP)", RFC 3326,
         December 2002.

   [15]  Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating
         User Agent Capabilities in the Session Initiation Protocol
         (SIP)", RFC 3840, August 2004.

   [16]  Rosenberg, J., "Request Authorization through Dialog
         Identification in the Session Initiation Protocol (SIP)",
         RFC 4538, June 2006.

   [17]  Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.

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

   [19]  Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and
         D. Gurle, "Session Initiation Protocol (SIP) Extension for
         Instant Messaging", RFC 3428, December 2002.

   [20]  Niemi, A., "Session Initiation Protocol (SIP) Extension for
         Event State Publication", RFC 3903, October 2004.

   [21]  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.











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Authors' Addresses

   Andrew Allen (editor)
   Research in Motion (RIM)
   102 Decker Court, Suite 100
   Irving, Texas  75062
   USA

   Phone: unlisted
   Fax:   unlisted
   Email: aallen@rim.com


   Jan Holm
   Ericsson
   Gotalandsvagen 220
   Stockholm  612526
   Sweden

   Phone: unlisted
   Fax:   unlisted
   Email: Jan.Holm@ericsson.com


   Tom Hallin
   Motorola
   1501 W Shure Drive
   Arlington Heights  60004
   USA

   Phone: unlisted
   Fax:   unlisted
   Email: thallin@motorola.com


















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Full Copyright Statement

   Copyright (C) The IETF Trust (2007).

   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
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   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
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Acknowledgment

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