| Rfc | 4964 |
|---|
| Title | The P-Answer-State Header Extension to the Session Initiation
Protocol for the Open Mobile Alliance Push to Talk over Cellular |
|---|
| Author | A.
Allen, Ed., J. Holm, T. Hallin |
|---|
| Date | September 2007 |
|---|
| Format: | TXT, HTML |
|---|
| Updated by | RFC8996 |
|---|
| Status: | INFORMATIONAL |
|---|
|
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.
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
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-
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
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.
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
"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".
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.
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.
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".
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
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,
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.
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]:
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.
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
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)
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
(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)
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.
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.
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)
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>
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)
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
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)
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
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:
+----------------+--------------+-----------+
| 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.
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.
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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