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SIP信令协议消息（四） SIP消息分类请求消息消息含义 INVITE 发起会话请求，邀请用户加入一个会话，会话描述含于消息体中。对于两方呼叫来说，主叫方在会话描述中指示其能够接受的媒体类型及其参数。被叫方必需在成功响应消息的消息体中指明其希望接受哪些媒体，还可以指示其将发送的媒体。 ACK 证实已收到对于INVITE请求的最终响应。该消息仅和INVITE消息配套使用。 BYE 结束会话 CANCEL 取消尚未完成的请求，对于已完成的请求（即已收到最终响应的请求）则阅读详情

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

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)