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Session Initiation Protocol (sip)领导组主页

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 帝都老白
发布于 2015/03/03 11:05
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http://www.ietf.org/html.charters/sip-charter.html

Session Initiation Protocol (sip)


In addition to this official charter maintained by the IETF Secretariat, there is additional information about this working group on the Web at:

       Additional SIP Page

Last Modified: 2005-01-31

Chair(s):

Dean Willis <dean.willis@softarmor.com>
Rohan Mahy < rohan@ekabal.com>

Transport Area Director(s):

Allison Mankin <mankin@psg.com>
Jon Peterson <jon.peterson@neustar.biz>

Transport Area Advisor:

Allison Mankin <mankin@psg.com>

Technical Advisor(s):

Dan Romascanu <dromasca@avaya.com>

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: There 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 sessions between users.
Such sessions include voice, video, chat, interactive games, and
virtual reality. The main work 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, including the SIPPING working group, when those
requirements are within 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 whenever possible.

2. Extensions 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 successor 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 to
several 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 the
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. bis: 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 signaling, aka SIP "cookies".

7. priv: Completion of SIP extensions for security and privacy.

8. security: Assuring generally adequate security  and privacy
  mechanisms within SIP.

9. provrel: Completion of the SIP extensions needed for reliability of
  provisional messages.

10. servfeat: Completion of the SIP extensions needed for negotiation
    ofserver 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 negotiating 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-INVITE Transaction (22520 bytes)
Actions addressing identified issues with the Session Initiation Protocol's non-INVITE 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 Reason Header Field for the Session Initiation Protocol (SIP) (RFC 3326) (15695 bytes)
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 Inititation 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 Inititation 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 ietf-web@ietf.org.

Return to working group directory.

Return to IETF home page.

本文转载自:http://blog.csdn.net/ioke/article/details/288453

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