path: root/doc/rfc/rfc4964.txt

Network Working Group                                      A. Allen, Ed.
Request for Comments: 4964                      Research in Motion (RIM)
Category: Informational                                          J. Holm
                                                                Ericsson
                                                               T. Hallin
                                                                Motorola
                                                          September 2007


 The P-Answer-State Header Extension to the Session Initiation Protocol
        for the Open Mobile Alliance Push to Talk over Cellular

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.

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

























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Table of Contents

   1. Introduction ....................................................3
   2. Overall Applicability ...........................................3
   3. Terminology .....................................................3
   4. Background for the Extension ....................................4
   5. Overview ........................................................5
   6. The P-Answer-State Header .......................................6
      6.1. Requirements ...............................................8
      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 .....................14
   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 ....................................15
      8.1. Pre-Arranged Group Call Using On-Demand Session ...........15
      8.2. 1-1 Call Using Pre-Established Session ....................21
   9. Security Considerations ........................................28
   10. IANA Considerations ...........................................28
      10.1. Registration of Header Fields ............................28
   11. Acknowledgements ..............................................29
   12. References ....................................................29
      12.1. Normative References .....................................29
      12.2. Informative References ...................................30























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1.  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.

2.  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
   statement in section 6.3.

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-





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   to-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
   speakerphone 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 speakerphone, 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
   answer mode to a PTT Server in order to enable buffering are defined



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   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 is defined in
   [12] that allows a terminal that acts as a SIP UA (or as 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.  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 user 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 that 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.  If the extension is not
   understood, the header field is simply ignored by the recipient.  The
   extension is described below.



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   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 it has information that hints that the terminating UA identified
   in the Request-URI of the request will likely answer automatically.
   Thus, it enables 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
   "Unconfirmed" is included in a provisional response that contains an
   answer, the PTT Server is leaving the decision of where to do
   buffering to other PTT Servers upstream and will forward upstream a



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   "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 forwards the media and allows 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 acting as a B2BUA that understands this extension in the
   signaling path between itself and the originating UAC.  This PTT
   Server between the sending node and the originating UAC 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 time
   out 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.  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.  Therefore, sending provisional responses
   reliably could potentially delay the sending of the "Confirmed
   Response".









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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
   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 UASs 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 UAS 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 UAS
      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 UAS is likely
      to automatically accept the session invitation.

   REQ-4:  An intermediate server that is willing to spool media from
      the inviting SIP UAC until one or more invited SIP UASs 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's 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.





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   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
   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 or
   just an intermediary has accepted the SIP INVITE request.

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
   that are trusted and use hop-by-hop security mechanisms with
   transitive trust.  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 UAS 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 UAS 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 is 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 or not the media was 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 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 that 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 that 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 receives a "Confirmed Response" at the UAC part of
   its back-to-back UA, 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].  Also, it 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 in the
   sipfrag of the response contained in a SIP NOTIFY request.  The
   P-Answer-State header field that is contained in the sipfrag,



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   contains the parameters from the P-Answer-State from the original
   response unmodified.  This SIP NOTIFY request is the SIP NOTIFY
   request that the UAS part of the back-to-back UA of the PTT Server
   sends in response to the original SIP REFER request based upon
   receiving the SIP 18x or 2xx response.  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.

   In the case where a PTT Server that's UAS part of its back-to-back UA
   previously sent a SIP NOTIFY request as a result of the SIP REFER
   request:

   1) the SIP NOTIFY request contains a P-Answer-State header field with
      the parameter "Unconfirmed" in the sipfrag of a response, and

   2) the PTT Server subsequently receives at the UAC part of its back-
      to-back UA a "Confirmed Response" to the SIP INVITE request.

   Such a PTT Server 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 is 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].  Also, it 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.







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RFC 4964               The P-Answer-State Header          September 2007


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 (in a manner compliant with [2]) if it has information
   that hints that the final destination UA for the SIP INVITE request
   is likely to automatically accept the session.

   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]:











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      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            -   -   -   -   -   -   -

      Table 1: Additional Table Entries for the P-Answer-State Header

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.

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 Alice'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.






















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      Alice's        Alice'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 1: Pre-Arranged Group Call Using On-Demand Session

   1 INVITE Alice -> Alice'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>
   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



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RFC 4964               The P-Answer-State Header          September 2007


   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

   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)




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RFC 4964               The P-Answer-State Header          September 2007


   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
   (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




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   (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.

   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)




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   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
   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.






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RFC 4964               The P-Answer-State Header          September 2007


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 Alice'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 as 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 2: 1-1 Call Using Pre-Established Session

   1 INVITE Alice -> Alice'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

   Alice'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 -> Alice'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>






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RFC 4964               The P-Answer-State Header          September 2007


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

   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
   Referred-By: <sip:Alice@example.org>
   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
   Referred-By: <sip:Alice@example.org>
   Call-ID: 6eb4c66a847710
   CSeq: 478209 INVITE
   Contact: <sip:BobsPTTServer.example.com>
   Content-Type: application/sdp
   Content-Length: 142

   (SDP not shown)







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

   9 NOTIFY Alice'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






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   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
   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)




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   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
   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 Alice'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




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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.

   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 attack that falsely indicates that the callee
   has yet to be contacted when they have already answered attack could
   result in the caller deciding not to transmit media because they do
   not wish to have their media stored by an intermediary even though in
   reality the callee has answered.  The attack that denies the callee
   the additional knowledge that the callee has yet to be contacted 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 that are
   trusted and use hop-by-hop security mechanisms with transitive trust.
   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 row has been added to the "Header Fields" section of
   the SIP parameter registry:




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               +----------------+--------------+-----------+
               | Header Name    | Compact Form | Reference |
               +----------------+--------------+-----------+
               | P-Answer-State |              | [RFC4964] |
               +----------------+--------------+-----------+

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.



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12.2.  Informative References

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

   [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., Ed., and A. Allen, "Requesting Answering Modes for
         the Session Initiation Protocol (SIP)", Work in Progress, June
         2007.

   [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

   EMail: aallen@rim.com


   Jan Holm
   Ericsson
   Tellusborgsvagen 83-87
   Stockholm  12526
   Sweden

   EMail: Jan.Holm@ericsson.com


   Tom Hallin
   Motorola
   1501 W Shure Drive
   Arlington Heights, IL  60004
   USA

   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
   retain all their rights.

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   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
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   The IETF invites any interested party to bring to its attention any
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   ietf-ipr@ietf.org.












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