Http://www.ietf.org/html.charters/sip-charter.html
Session Initiation Protocol (SIP)
In addition to this official chartained by the Ietf Secretariat, There IS Additional Information INFORMATION About this working group on the Web At:
Additional Sip Page
Last Modified: 2005-01-31
Chair (s):
Dean Willis
Rohan Mahy
Transport Area Director (s):
Allison mankin
Jon Peterson
TRANSPORT Area Advisor:
Allison mankin
Technical Advisor (s):
Dan Romascanu
Mailing Lists: General Discussion:
Sip@ietf.org
To Subscribe:
SIP-REQUEST@ietf.org
In Body: Subscribe
Archive:
http://www.ietf.org/mail-archive/web/sip/index.html
Description of Working Group: NOTE: THE IS Another
SIP Email List for General Information and
Implementations:
Discussion of EXISTING
SIP:
SIP-IMPLEMENTORS@cs.columbia.edu
To Subscribe:
SIP-IMPLEMENTORS-REQUEST@cs.columbia.edu
In Body: Subscribe Archive:
Http://lists.cs.columbia.edu/mailman/listinfo/
SIP-IMPLEMENTORS
============================================================================================================================================================================================================= =====================
The session initiation protocol
SIP) Working Group is chartered to
Continue the development of
SIP, Currently Specified as proposed
STANDARD RFC 2543.
SIP is a Text-based protocol, Similar to http and
SMTP, for Initiating Interactive Communication Session Between Uses.such Session Include Voice, Video, Chat, Interactive Games, and
Virtual reality. The main thing of the group involves bringing
SIP FROM
Proposed to Draft Standard, in Addition to Specifying and Developing
Proposed Extensions That Arise Out of Strong Requirements. THE
SIP
Working Group Will Concentrate On The Specification of
SIP and ITS
Extensions, And Will NOT Explore the use of
SIP for specific
Environments or Applications. It Will, However Respond To
General-Purpose Requirements for Changes To
SIP Provided by Other
Working groups, incruding the
Sipping Working Group, When Those
Requirements Are With the Scope and Charter of
SIP.
Throughout ITS Work, The Group Will Strive To Maintain The Basic Model
And Architecture Defined By
SIP. In particular:
1. Services and features are provided end-to-end wherever possible.
2. EXTENSION AND NEW FEATURES MUST BE Generally Applicable, And Not
Applicable Only to a Specific Set of Session Types.
3. Simplicity is key.
4. Reuse of existing ip protocols and architectures, and integrating
WITH OTHER IP Applications, IS CRUCIAL.
Sip Was First Developed within The Multiparty MultiMedia Session
Control (mmusic) Working Group, and the
SIP Working Group Will Continue
To Maintain Active Communications with mmusic. this is particularly
Important Since The Main Mime Type Carried in
SIP MESSAGES, THE SESSION
Description Protocol (SDP), Specified In RFC 2327, IS Developed by
Mmusic and Because Mmusic Is Developing a Chengpssor to SDP Which
SIP
Will Also Use.
The Group Will Work Very Closely with The (proposed)
Sipping wg, Which
Is Expected to Analyze The Requirements for Application of
SIP TOSEVERAL DIFFERENT TASKS, AND with THE SIMPLE WG, WHICH IS USING
SIP for
Messaging and presence.
The Group Will Also Maintain Open Dialogues with the ip telephony
(Iptel) WG, Whose Call Processing Language (CPL) RELATES To MANY
Features of
SIP; Will Continue to Consider the Requirements and
Specifications previously established by the Pstn and Internet
Internetworking (PINT) Working Group ;: And Will Consider Input from T
Distributed Call Signaling (DCS) Group of the PacketCable Consortium
For Distributed TELEPHONY SERVICES, AND FROM 3GPP, 3GPP2, And MWIF for
Third-Generation Wireless Network Requirements.
The Specific DeliveRables of the Group Are:
1. BI: a Draft Standard Version of
SIP.
2. CallControl: completion of the
SIP CALL Control Specifications,
Which Enables Multiparty Services, Such As Transfer and Bridged
SESSIONS.
3. Callerpref: completion of the
SIP CALLER Preferences Extensions,
Which Enables Intelligent Call Routing Services.
4. MIB: DEFINE A MIB FOR
SIP NODES.
5. Precon: Completion of the
SIP
Extensions Needed to Assure Satisfaction of External Preconditions
SUCH AS QoS Establishment.
6. State: Completion of the
SIP EXTENSIONS Needed to Manage State
Within signing, Aka
SIP "cookies".
7. PRIV: COMPLETION OF
SIP EXTENSIONS for Security and Privacy.
8. Security: Assuling Generally Adequate Security and Privacy
Mechanisms Within
SIP.
9. Provcl: Completion of the
SIP Extensions Needed for Reliability of
Province.
10. Servfeat: completion of the
SIP EXTENSIONS NEEDED for Negotiation
Of Server Features.
11. Sesstimer: completion of the
SIP session Timer Extension.
12. Events: completion of the
SIP Events Extensions (Subscribe / Notify).
13. Security: Requirements for privacy and security.
14. COMPRESSION:
SIP MECHANISMS for Negotiarating and Guidelines for
Using
Signaling compression as defined in rohc.
15. Content Indirection: a proposed Standard Mechanism to Reference
SIP Content Indirectly (by Reference, for Example Using An External
URI).
Other DeliveRables May Be Agreed Upon As Extensions Are proposed. New
Deliverables Must Be Approved by The Transport Area Directors Before
Inclusion on the agenda.
NOTE: MILESTONES WITHIN THE SAME MONTH Are Shown in Order of Planned
COMPLETION.
Goals and Milestones:
Done Server Features Negotiation submitted to IESG Done Complete IESG requested fixes to provrel and servfeat Done Revised proposed standard version of SIP (2543bis) submitted to IESG Done SIP Events specification to IESG Done The UPDATE Method submitted for Proposed Standard Done SIP extensions for media authorization ( call-auth) submitted as Informational Done Preconditions extensions (manyfolks) spec to IESG Done SIP Privacy specification to IESG Done SIP Privacy and Security Requirements to IESG Done The MESSAGE Method submitted for Proposed Standard Done The Replaces Header submitted for Proposed Standard Done Refer spec to IESG Done SIP NAT extension submitted to IESG Done SIP over SCTP specification and applicability statement Done Mechanism for Content Indirection in SIP submitted to IESG for Proposed Standard Done The SIP Referred-By Header submitted to IESG for Proposed Standard Done Session Timer spec,
revised to IESG Done Caller preferences specification submitted to IESG Done Submit SIP Identity documents to IESG for Proposed Standard Done The SIP Join Header submitted to IESG for Proposed Standard Done Replaces header to IESG (PS) Done Upgrade S / MIME requirement for AES in 3261 to IESG (PS) Mar 04 Application Interaction to IESG (BCP) Mar 04 Resource Priority signaling mechanism to IESG (PS) Done Presence Publication to IESG (PS) Apr 04 Connection reuse mechanism to IESG (PS) Apr 04 Enhancements for Authenticated Identity Management to IESG (BCP) Done Guidelines for Authors of SIP extensions submitted as Informational May 04 Mechanism for obtaining globally routable unique URIs to IESG (PS) Jun 04 MIB spec to IESG Sep 04 Review WG status (consider closing) and / or submit a future milestones Plan to IESG DONE Request History Mechanism To IESG (PS) Internet-Drafts:
Session Timers in The session Initiation Protocol (SIP) (67251 BYTES)
Management Information Base for Session Initiation Protocol (SIP) (207354 BYTES)
Guidelines for authors of extensions to the session Initiation Protocol (SIP) (55339 BYTES)
The Stream Control Transmission Protocol (SCTP) As a Transport for the Session Initiation Protocol (SIP) (23854 BYTES)
Compressing the session initiation protocol (23311 BYTES)
A Mechanism for Content Indirection In Session Initiation Protocol (SIP) Messages (38368 BYTES)
Enhancements for Authenticated Identity Management In The Session Initiation Protocol (SIP) (82783 BYTES)
An extension to the session initiation protocol for request history information (117689 bytes)
Communications Resource Priority for the session Initiation Protocol (SIP) (79627 BYTES)
Connection Reuse In the session Initiation Protocol (SIP) (30416 BYTES) Update to the session Initiation Protocol (SIP) Preconditions framework (21088 bytes)
Problems Identified Associated with the session initiation protocol's non-inviteral transaction (22520 bytes)
Actions addressing id1Ified Issues with the session initiation protocol's non-inviteral transaction (15080 bytes)
Usage of the session description Protocol (SDP) Alternative Network Address Types (Anat) Semantics In The Session Initiation Protocol (SIP) (13473 BYTES)
Suppression of Refer Implicit Subscription (13399 BYTES)
Request for Comments:
The Sip Info Method (RFC 2976) (17736 BYTES)
Mime Media Types for Isup and Qsig Objects (RFC 3204) (19712 BYTES)
SIP: Session Initiation Protocol (RFC 3261) (647976 BYTES)
Reliability of Provisional Responses in Sip (RFC 3262) (29643 BYTES)
SIP: Locating Sip Servers (RFC 3263) (42310 BYTES)
SIP-Specific Event Notification (RFC 3265) (89005 BYTES)
DHCP OPTION for SIP Servers (RFC 3361) (12549 BYTES)
Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA) (RFC 3310) (36985 BYTES)
The session initiation protocol update method (RFC 3311) (28125 BYTES)
Integration of Resource Management and Sip (RFC 3312) (65757 BYTES)
Internet Media Type Message / Sipfrag (RFC 3420) (14745 BYTES)
A Privacy Mechanism for the session Initiation Protocol (SIP) (RFC 3323) (54116 BYTES)
Private extensions to the session initiation protocol (sip) for asserted identity within trusted networks (RFC 3325) (36170 BYTES)
Session Initiation Protocol Extension for Instant Messaging (RFC 3428) (41475 BYTES)
THE REAON Header Field for the session Initiation Protocol (SIP) (RFC 3326) Session Initiation Protocol Extension for Registering Non-Adjacent Contacts (RFC 3327) (36493 BYTES)
Security Mechanism Agreement for the session Initiation Protocol (SIP) Sessions (RFC 3329) (51503 BYTES)
Private session Initiation Protocol (SIP) Extensions for Media Authorization (RFC 3313) (36866 BYTES)
The session initiation protocol (SIP) Refer method (RFC 3515) (47788 BYTES)
Dynamic Host Configuration Protocol (DHCPV6) Options for Session Initiation Protocol (SIP) Servers (RFC 3319) (14444 BYTES)
An extension to the session initiation protocol (SIP) for Symmetric Response Routing (RFC 3581) (29121 BYTES)
Session Initiation Protocol Extension Header Field for Service Route Discovery During Registration (RFC 3608) (35628 BYTES)
S / MIME AES Requirement for SIP (RFC 3853) (0 bytes)
Indicating User Agent Capabilities in The session Initiation Protocol (SIP) (RFC 3840) (0 bytes)
Caller Preferences for the session Initiation Protocol (SIP) (RFC 3841) (0 bytes)
The session initation protocol (SIP) 'Replaces' Header (RFC 3891) (0 bytes)
The Sip Referred-by Mechanism (RFC 3892) (0 bytes)
SIP Authenticated Identity Body (AIB) Format (RFC 3893) (0 BYTES)
The session initation protocol (SIP) 'Join' Header (RFC 3911) (0 bytes)
An Event State Publication Extension To The Session Initiation Protocol (SIP) (RFC 3903) (0 Bytes)
THE Internet Assigned Number Authority (IANA) Header Field Parameter Registry for the session Initiation Protocol (SIP) (RFC 3968) (0 bytes)
The Internet Assigned Number Authority (IANA) Universal Resource Identifier (URI) Parameter Registry for the Session Initiation Protocol (SIP) (RFC 3969) (0 bytes) IETF Secretariat - Please send questions, comments, and / or suggestions to
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