Subject: [Sipping] Protocol Action: 'Session Initiation Protocol Call Control - Conferencing for User Agents' to BCP

From: The IESG <[email-masked]>
Date: Mon, 27 Jun 2005 15:38:28 -0400

The IESG has approved the following document:

- 'Session Initiation Protocol Call Control - Conferencing for User Agents '
   <draft-ietf-sipping-cc-conferencing-07.txt> as a BCP

This document is the product of the Session Initiation Proposal Investigation 
Working Group. 

The IESG contact persons are Allison Mankin and Jon Peterson.

A URL of this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-sipping-cc-conferencing-07.txt

Technical Summary

   This specification defines conferencing call control features for the
   Session Initiation Protocol (SIP).  This document builds on the
   Conferencing Requirements and Framework documents to define how a
   tightly coupled SIP conference works.  The approach is explored from
   different user agent (UA) types perspective: conference-unaware,
   conference-aware and focus UAs.  The use of URIs in conferencing,
   OPTIONS for capabilities discovery, and call control using REFER are
   covered in detail with example call flow diagrams.  The usage of the
   isfocus feature tag is defined.

   This specification uses the concepts and definitions from the WG's 
   "High Level Requirements for Tightly Coupled SIP Conferencing,"
    and "A Framework for Conferencing with the Session Initiation Protocol,"
    approved earlier.  In the tightly coupled architecture, a UA, known
    as participant, establishes a SIP dialog with another UA, known as
    focus.  The focus is the central point of control, authentication and
    authorization.  This specification defines the operations of a focus
    and participant UAs.  Not that only the signaling (SIP) needs to be
    centralized in this model - the media can be centrally mixed,
    distributed, or even multicast (by the nature of the media descriptions
    that the model establishes).  For a full discussion of this architecture,
    see the SIP conferencing Framework mentioned already.
   already.
       
   This document presents the basic call control (dial-in and dial-out)
   conferencing building blocks from the UA perspective. Possible 
   applications include ad-hoc conferences and scheduled conferences.

Working Group Summary
 
 The working group strongly supported advancing this document.

 3GPP and OMA have notified the IETF that this specification is a 
 critical dependency.
 
Protocol Quality
 
 Allison Mankin reviewed the specification for the IESG.  It was
 revised to add specific security considerations.  Due to a
 General Area Directorate Review, it was revised to add some
 additional context and introduction.  

 Gonzalo Camarillo is the working group shepherd.

Notes to the RFC Editor
 
 Section 3.1 Remove the sentence:
"A focus SHOULD utilize a GRUU as discussed in Section 4.2."

Section 4.2
OLD:
The Conference URI MUST have the GRUU (Globally Routable User Agent
URI) properties as detailed in [16].

NEW:
By their nature, the conferences supported by this specification are
centralized.  Therefore, typically a conferencing system needs to
allocate a SIP Conference URI such that SIP requests to this URI are not
forked and are routed to a dedicated conference focus. For example, for
a globally accessible SIP conference could be well constructed with
a Conference URI using a Globally Routable User Agent URI (GRUU) 
(defined in [16]), because of its ability to support the non-forking and 
global routability requirements.

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