Network Working Group G. Camarillo
Request for Comments: 5370 Ericsson
Category: Standards Track October 2008
Network Working Group G. Camarillo
Request for Comments: 5370 Ericsson
Category: Standards Track October 2008
The Session Initiation Protocol (SIP) Conference Bridge Transcoding Model
会话发起协议（SIP）会议桥转码模型
Status of This Memo
关于下段备忘
This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.
本文件规定了互联网社区的互联网标准跟踪协议，并要求进行讨论和提出改进建议。有关本协议的标准化状态和状态，请参考当前版本的“互联网官方协议标准”（STD 1）。本备忘录的分发不受限制。
Abstract
摘要
This document describes how to invoke transcoding services using the conference bridge model. This way of invocation meets the requirements for SIP regarding transcoding services invocation to support deaf, hard of hearing, and speech-impaired individuals.
本文档描述了如何使用会议桥模型调用代码转换服务。这种调用方式满足SIP关于转码服务调用的要求，以支持聋人、听力障碍者和言语障碍者。
Table of Contents
目录
1. Introduction ....................................................2
2. Terminology .....................................................3
3. Caller's Invocation .............................................3
3.1. Procedures at the User Agent ...............................3
3.2. Procedures at the Transcoder ...............................3
3.3. Example ....................................................4
3.4. Unsuccessful Session Establishment .........................6
4. Callee's Invocation .............................................7
5. Security Considerations .........................................7
6. Contributors ....................................................8
7. References ......................................................8
7.1. Normative References .......................................8
7.2. Informative References .....................................9
1. Introduction ....................................................2
2. Terminology .....................................................3
3. Caller's Invocation .............................................3
3.1. Procedures at the User Agent ...............................3
3.2. Procedures at the Transcoder ...............................3
3.3. Example ....................................................4
3.4. Unsuccessful Session Establishment .........................6
4. Callee's Invocation .............................................7
5. Security Considerations .........................................7
6. Contributors ....................................................8
7. References ......................................................8
7.1. Normative References .......................................8
7.2. Informative References .....................................9
RFC 5369 [RFC5369] describes how two SIP [RFC3261] UAs (User Agents) can discover incompatibilities that prevent them from establishing a session (e.g., lack of support for a common codec or for a common media type). When such incompatibilities are found, the UAs need to invoke transcoding services to successfully establish the session. The transcoding framework introduces two models to invoke transcoding services: the 3pcc (third-party call control) model [RFC4117] and the conference bridge model. This document specifies the conference bridge model.
RFC 5369[RFC5369]描述了两个SIP[RFC3261]UAs（用户代理）如何发现不兼容，从而阻止它们建立会话（例如，缺少对通用编解码器或通用媒体类型的支持）。当发现这种不兼容时，UAs需要调用代码转换服务来成功建立会话。代码转换框架引入了两种模型来调用代码转换服务：3pcc（第三方呼叫控制）模型[RFC4117]和会议桥模型。本文档指定了会议桥模型。
In the conference bridge model for transcoding invocation, a transcoding server that provides a particular transcoding service (e.g., speech-to-text) behaves as a B2BUA (Back-to-Back User Agent) between both UAs and is identified by a URI. As shown in Figure 1, both UAs, A and B, exchange signalling and media with the transcoder T. The UAs do not exchange any traffic (signalling or media) directly between them.
在用于转码调用的会议网桥模型中，提供特定转码服务（例如，语音到文本）的转码服务器作为两个ua之间的B2BUA（背对背用户代理），并由URI标识。如图1所示，UAs、A和B都与转码器T交换信令和媒体。UAs不直接在它们之间交换任何通信量（信令或媒体）。
+-------+
| |**
| T | **
| |\ **
+-------+ \\ **
^ * \\ **
| * \\ **
| * SIP **
SIP * \\ **
| * \\ **
| * \\ **
v * \ **
+-------+ +-------+
| | | |
| A | | B |
| | | |
+-------+ +-------+
+-------+
| |**
| T | **
| |\ **
+-------+ \\ **
^ * \\ **
| * \\ **
| * SIP **
SIP * \\ **
| * \\ **
| * \\ **
v * \ **
+-------+ +-------+
| | | |
| A | | B |
| | | |
+-------+ +-------+
<-SIP-> Signalling
******* Media
<-SIP-> Signalling
******* Media
Figure 1: Conference bridge model
图1：会议桥模型
Sections 3 and 4 specify how the caller A or the callee B, respectively, can use the conference bridge model to invoke transcoding services from T.
第3节和第4节分别指定了调用方A或被调用方B如何使用会议网桥模型从T调用代码转换服务。
In this document, the key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" are to be interpreted as described in BCP 14, RFC 2119 [RFC2119], and indicate requirement levels for compliant implementations.
在本文件中，关键词“必须”、“不得”、“要求”、“应”、“不应”、“应”、“不应”、“建议”、“不建议”、“可”和“可选”将按照BCP 14、RFC 2119[RFC2119]中的描述进行解释，并指出合规实施的要求级别。
User agent A needs to perform two operations to invoke transcoding services from T for a session between user agent A and user agent B. User agent A needs to establish a session with T and provide T with user agent B's URI so that T can generate an INVITE towards user agent B.
用户代理A需要执行两个操作，以从T调用转码服务，用于用户代理A和用户代理B之间的会话。用户代理A需要与T建立会话，并向T提供用户代理B的URI，以便T可以生成对用户代理B的邀请。
User agent A uses the procedures for RFC 5366 [RFC5366] to provide T with B's URI using the same INVITE that establishes the session between A and T. That is, user agent A adds to the INVITE a body part whose disposition type is recipient-list [RFC5363]. This body part consists of a URI-list that contains a single URI: user agent B's URI.
用户代理A使用RFC 5366[RFC5366]的过程，使用在A和T之间建立会话的相同INVITE向T提供B的URI。也就是说，用户代理A向INVITE添加一个处置类型为收件人列表[RFC5363]的主体部分。这个主体部分由一个URI列表组成，该列表包含一个URI：用户代理B的URI。
Note that, as described in the transcoding framework [RFC5369], the transcoding model described in this document is modeled as a two-party conference server. Consequently, this document focuses on two-party sessions that need transcoding. Multi-party sessions can be established using INVITE requests with multiple URIs in their bodies, as specified in [RFC5366].
注意，如代码转换框架[RFC5369]中所述，本文中描述的代码转换模型被建模为一个两方会议服务器。因此，本文件侧重于需要转码的两党会议。如[RFC5366]所述，多方会话可以使用主体中包含多个URI的INVITE请求建立。
On receiving an INVITE with a URI-list body, the transcoder follows the procedures in [RFC5366] to generate an INVITE request towards the URI contained in the URI-list body. Note that the transcoder acts as a B2BUA, not as a proxy.
在接收到带有URI列表主体的INVITE时，转码器按照[RFC5366]中的过程生成针对URI列表主体中包含的URI的INVITE请求。请注意，转码器充当B2BUA，而不是代理。
Additionally, the transcoder MUST generate the From header field of the outgoing INVITE request using the same value as the From header field included in the incoming INVITE request, subject to the privacy requirements (see [RFC3323] and [RFC3325]) expressed in the incoming INVITE request. Note that this does not apply to the "tag" parameter.
此外，根据传入INVITE请求中表达的隐私要求（参见[RFC3323]和[RFC3325]），转码器必须使用与传入INVITE请求中包含的From报头字段相同的值生成传出INVITE请求的From报头字段。请注意，这不适用于“tag”参数。
The session description the transcoder includes in the outgoing INVITE request depends on the type of transcoding service that particular transcoder provides. For example, a transcoder resolving audio codec incompatibilities would generate a session description listing the audio codecs the transcoder supports.
转码器在传出INVITE请求中包含的会话描述取决于特定转码器提供的转码服务的类型。例如，解决音频编解码器不兼容的转码器将生成一个会话描述，列出转码器支持的音频编解码器。
When the transcoder receives a final response for the outgoing INVITE requests, it generates a new final response for the incoming INVITE request. This new final response SHOULD have the same status code as the one received in the response for the outgoing INVITE request.
当代码转换器收到传出INVITE请求的最终响应时，它会为传入INVITE请求生成新的最终响应。此新的最终响应应具有与传出INVITE请求的响应中接收到的状态代码相同的状态代码。
If a transcoder receives an INVITE request with a URI-list with more than one URI, it SHOULD return a 488 (Max 1 URI allowed in URI-list) response.
如果代码转换器接收到具有多个URI的URI列表的INVITE请求，它应该返回488（URI列表中允许的最大URI为1）响应。
Figure 2 shows the message flow for the caller's invocation of a transcoder T. The caller A sends an INVITE (1) to the transcoder (T) to establish the session A-T. Following the procedures in [RFC5366], the caller A adds a body part whose disposition type is recipient-list [RFC5363].
图2显示了调用者调用转码器T的消息流。调用者a向转码器（T）发送INVITE（1）以建立会话a-T。按照[RFC5366]中的过程，调用者a添加一个处置类型为收件人列表[RFC5363]的主体部分。
A T B
| | |
|-----(1) INVITE SDP A----->| |
| | |
|<-(2) 183 Session Progress-| |
| |-----(3) INVITE SDP TB---->|
| | |
| |<-----(4) 200 OK SDP B-----|
| | |
| |---------(5) ACK---------->|
|<----(6) 200 OK SDP TA-----| |
| | |
|---------(7) ACK---------->| |
| | |
| ************************* | ************************* |
|** Media **|** Media **|
| ************************* | ************************* |
| | |
A T B
| | |
|-----(1) INVITE SDP A----->| |
| | |
|<-(2) 183 Session Progress-| |
| |-----(3) INVITE SDP TB---->|
| | |
| |<-----(4) 200 OK SDP B-----|
| | |
| |---------(5) ACK---------->|
|<----(6) 200 OK SDP TA-----| |
| | |
|---------(7) ACK---------->| |
| | |
| ************************* | ************************* |
|** Media **|** Media **|
| ************************* | ************************* |
| | |
Figure 2: Successful invocation of a transcoder by the caller
图2：调用者成功调用代码转换器
The following example shows an INVITE with two body parts: an SDP [RFC4566] session description and a URI-list.
下面的示例显示了一个包含两个主体部分的INVITE：SDP[RFC4566]会话描述和URI列表。
INVITE sip:transcoder@example.com SIP/2.0
Via: SIP/2.0/TCP client.chicago.example.com
;branch=z9hG4bKhjhs8ass83
Max-Forwards: 70
To: Transcoder <sip:transcoder@example.org>
From: A <sip:A@chicago.example.com>;tag=32331
Call-ID: d432fa84b4c76e66710
CSeq: 1 INVITE
Contact: <sip:A@client.chicago.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: dialog
Accept: application/sdp, message/sipfrag
Require: recipient-list-invite
Content-Type: multipart/mixed;boundary="boundary1"
Content-Length: 556
INVITE sip:transcoder@example.com SIP/2.0
Via: SIP/2.0/TCP client.chicago.example.com
;branch=z9hG4bKhjhs8ass83
Max-Forwards: 70
To: Transcoder <sip:transcoder@example.org>
From: A <sip:A@chicago.example.com>;tag=32331
Call-ID: d432fa84b4c76e66710
CSeq: 1 INVITE
Contact: <sip:A@client.chicago.example.com>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
SUBSCRIBE, NOTIFY
Allow-Events: dialog
Accept: application/sdp, message/sipfrag
Require: recipient-list-invite
Content-Type: multipart/mixed;boundary="boundary1"
Content-Length: 556
--boundary1
Content-Type: application/sdp
--boundary1
Content-Type: application/sdp
v=0
o=example 2890844526 2890842807 IN IP4 chicago.example.com
s=-
c=IN IP4 192.0.2.1
t=0 0
m=audio 50000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
v=0
o=example 2890844526 2890842807 IN IP4 chicago.example.com
s=-
c=IN IP4 192.0.2.1
t=0 0
m=audio 50000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
--boundary1 Content-Type: application/resource-lists+xml Content-Disposition: recipient-list
--boundary1内容类型：应用程序/资源列表+xml内容处置：收件人列表
<?xml version="1.0" encoding="UTF-8"?>
<resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<list>
<entry uri="sip:B@example.org" />
</list>
</resource-lists>
--boundary1--
<?xml version="1.0" encoding="UTF-8"?>
<resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<list>
<entry uri="sip:B@example.org" />
</list>
</resource-lists>
--boundary1--
On receiving the INVITE, the transcoder generates a new INVITE towards the callee. The transcoder acts as a B2BUA, not as a proxy. Therefore, this new INVITE (3) belongs to a different transaction than the INVITE (1) received by the transcoder.
在接收到邀请后，转码器向被叫方生成一个新的邀请。转码器充当B2BUA，而不是代理。因此，这个新的INVITE（3）属于与转码器接收的INVITE（1）不同的事务。
When the transcoder receives a final response (4) from the callee, it generates a new final response (6) for INVITE (1). This new final response (6) has the same status code as the one received in the response from the callee (4).
当转码器从被叫方接收到最终响应（4）时，它为INVITE（1）生成新的最终响应（6）。这个新的最终响应（6）的状态代码与从被叫方（4）收到的响应中的状态代码相同。
Figure 3 shows a similar message flow as the one in Figure 3. Nevertheless, this time the callee generates a non-2xx final response (4). Consequently, the transcoder generates a non-2xx final response (6) towards the caller as well.
图3显示了与图3类似的消息流。然而，这次被调用方生成一个非2xx最终响应（4）。因此，转码器也向调用者生成非2xx最终响应（6）。
A T B
| | |
|-----(1) INVITE SDP A----->| |
| | |
|<-(2) 183 Session Progress-| |
| |-----(3) INVITE SDP TB---->|
| | |
| |<----(4) 603 Decline-------|
| | |
| |---------(5) ACK---------->|
|<----(6) 603 Decline-------| |
| | |
|---------(7) ACK---------->| |
| | |
A T B
| | |
|-----(1) INVITE SDP A----->| |
| | |
|<-(2) 183 Session Progress-| |
| |-----(3) INVITE SDP TB---->|
| | |
| |<----(4) 603 Decline-------|
| | |
| |---------(5) ACK---------->|
|<----(6) 603 Decline-------| |
| | |
|---------(7) ACK---------->| |
| | |
Figure 3: Unsuccessful session establishment
图3：会话建立失败
The ambiguity in this flow is that, if the provisional response (2) gets lost, the caller does not know whether the 603 (Decline) response means that the initial INVITE (1) was rejected by the transcoder or that the INVITE generated by the transcoder (4) was rejected by the callee. The use of the "History-Info" header field [RFC4244] between the transcoder and the caller resolves the previous ambiguity.
该流中的模糊性在于，如果临时响应（2）丢失，则调用者不知道603（拒绝）响应是指初始邀请（1）被转码器拒绝还是由转码器（4）生成的邀请被调用者拒绝。在转码器和调用者之间使用“历史信息”报头字段[RFC4244]解决了先前的歧义。
Note that this ambiguity problem could also have been resolved by having transcoders act as a pure conference bridge. The transcoder would respond with a 200 (OK) to the INVITE request from the caller, and it would generate an outgoing INVITE request towards the callee. The caller would get information about the result of the latter INVITE request by subscribing to the conference event package [RFC4575] at the transcoder. Although this flow would have resolved the ambiguity problem without requiring support for the "History-Info" header field, it is more complex, requires a higher number of messages, and introduces higher session setup delays. That is why it was not chosen to implement transcoding services.
请注意，这种模糊问题也可以通过将转码器用作纯粹的会议桥接器来解决。转码器将以200（OK）响应来自调用者的INVITE请求，并将向被调用者生成传出INVITE请求。调用者将通过在转码器上订阅会议事件包[RFC4575]来获得关于后一个INVITE请求结果的信息。虽然此流在不需要支持“History Info”头字段的情况下解决了歧义问题，但它更复杂，需要更多的消息，并且引入了更高的会话设置延迟。这就是为什么没有选择它来实现代码转换服务。
If a UA receives an INVITE with a session description that is not acceptable, it can redirect it to the transcoder by using a 302 (Moved Temporarily) response. The Contact header field of the 302 (Moved Temporarily) response contains the URI of the transcoder plus a "?body=" parameter. This parameter contains a recipient-list body with B's URI. Note that some escaping (e.g., for Carriage Returns and Line Feeds) is needed to encode a recipient-list body in such a parameter. Figure 4 shows the message flow for this scenario.
如果UA接收到会话描述不可接受的INVITE，它可以使用302（临时移动）响应将其重定向到转码器。302（临时移动）响应的联系人标头字段包含代码转换器的URI加上一个“？body=”参数。此参数包含具有B的URI的收件人列表正文。请注意，需要进行一些转义（例如，回车和换行），以便在此类参数中对收件人列表体进行编码。图4显示了此场景的消息流。
A T B
| | |
|-------------------(1) INVITE SDP A------------------->|
| | |
|<--------------(2) 302 Moved Temporarily---------------|
| | |
|-----------------------(3) ACK------------------------>|
| | |
|-----(4) INVITE SDP A----->| |
| | |
|<-(5) 183 Session Progress-| |
| |-----(6) INVITE SDP TB---->|
| | |
| |<-----(7) 200 OK SDP B-----|
| | |
| |---------(8) ACK---------->|
|<----(9) 200 OK SDP TA-----| |
| | |
|--------(10) ACK---------->| |
| | |
| ************************* | ************************* |
|** Media **|** Media **|
| ************************* | ************************* |
A T B
| | |
|-------------------(1) INVITE SDP A------------------->|
| | |
|<--------------(2) 302 Moved Temporarily---------------|
| | |
|-----------------------(3) ACK------------------------>|
| | |
|-----(4) INVITE SDP A----->| |
| | |
|<-(5) 183 Session Progress-| |
| |-----(6) INVITE SDP TB---->|
| | |
| |<-----(7) 200 OK SDP B-----|
| | |
| |---------(8) ACK---------->|
|<----(9) 200 OK SDP TA-----| |
| | |
|--------(10) ACK---------->| |
| | |
| ************************* | ************************* |
|** Media **|** Media **|
| ************************* | ************************* |
Figure 4: Callee's invocation of a transcoder
图4：被调用方对转码器的调用
Note that the syntax resulting from encoding a body into a URI as described earlier is quite complex. It is actually simpler for callees to invoke transcoding services using the 3pcc transcoding model [RFC4117] instead.
请注意，如前所述，将主体编码为URI所产生的语法相当复杂。实际上，被调用方使用3pcc代码转换模型[RFC4117]调用代码转换服务更简单。
Transcoders implementing this specification behave as a URI-list service as described in [RFC5366]. Therefore, the security considerations for URI-list services discussed in [RFC5363] apply here as well.
实现此规范的转码器的行为类似于[RFC5366]中所述的URI列表服务。因此，[RFC5363]中讨论的URI列表服务的安全注意事项也适用于此处。
In particular, the requirements related to list integrity and unsolicited requests are important for transcoding services. User agents SHOULD integrity protect URI-lists using mechanisms such as S/MIME [RFC3850] or TLS [RFC5246], which can also provide URI-list confidentiality if needed. Additionally, transcoders MUST authenticate and authorize users and MAY provide information about the identity of the original sender of the request in their outgoing requests by using the SIP identity mechanism [RFC4474].
特别是，与列表完整性和未经请求的请求相关的要求对于转码服务非常重要。用户代理应该使用S/MIME[RFC3850]或TLS[RFC5246]等机制保护URI列表的完整性，如果需要，这些机制还可以提供URI列表的机密性。此外，转码器必须对用户进行身份验证和授权，并可通过使用SIP身份机制[RFC4474]在其传出请求中提供有关请求原始发送者身份的信息。
The requirement in [RFC5363] to use opt-in lists (e.g., using RFC 5360 [RFC5360]) deserves special discussion. The type of URI-list service implemented by transcoders following this specification does not produce amplification (only one INVITE request is generated by the transcoder on receiving an INVITE request from a user agent) and does not involve a translation to a URI that may be otherwise unknown to the caller (the caller places the callee's URI in the body of its initial INVITE request). Additionally, the identity of the caller is present in the INVITE request generated by the transcoder. Therefore, there is no requirement for transcoders implementing this specification to use opt-in lists.
[RFC5363]中使用选择加入列表的要求（例如，使用RFC 5360[RFC5360]）值得特别讨论。由遵循该规范的转码器实现的URI列表服务的类型不会产生放大（转码器在接收到来自用户代理的INVITE请求时仅生成一个INVITE请求），并且不涉及到对URI的转换，否则调用方可能不知道该URI（调用者将被调用者的URI放在其初始INVITE请求的主体中）。此外，调用者的身份存在于代码转换器生成的INVITE请求中。因此，实现此规范的代码转换器不需要使用选择加入列表。
This document is the result of discussions amongst the conferencing design team. The members of this team include Eric Burger, Henning Schulzrinne, and Arnoud van Wijk.
本文件是会议设计团队讨论的结果。这个团队的成员包括埃里克·伯格、亨宁·舒尔兹林内和阿诺德·范威克。
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2119]Bradner，S.，“RFC中用于表示需求水平的关键词”，BCP 14，RFC 2119，1997年3月。
[RFC5246] Dierks, T. and E. Rescorla, "The Transport Layer Security (TLS) Protocol Version 1.2", RFC 5246, August 2008.
[RFC5246]Dierks，T.和E.Rescorla，“传输层安全（TLS）协议版本1.2”，RFC 5246，2008年8月。
[RFC3261] 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.
[RFC3261]Rosenberg，J.，Schulzrinne，H.，Camarillo，G.，Johnston，A.，Peterson，J.，Sparks，R.，Handley，M.，和E.Schooler，“SIP：会话启动协议”，RFC 3261，2002年6月。
[RFC3323] Peterson, J., "A Privacy Mechanism for the Session Initiation Protocol (SIP)", RFC 3323, November 2002.
[RFC3323]Peterson，J.，“会话启动协议（SIP）的隐私机制”，RFC3323，2002年11月。
[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, November 2002.
[RFC3325]Jennings，C.，Peterson，J.，和M.Watson，“在可信网络中声明身份的会话启动协议（SIP）的私有扩展”，RFC 33252002年11月。
[RFC3850] Ramsdell, B., Ed., "Secure/Multipurpose Internet Mail Extensions (S/MIME) Version 3.1 Certificate Handling", RFC 3850, July 2004.
[RFC3850]Ramsdell，B.，Ed.，“安全/多用途Internet邮件扩展（S/MIME）版本3.1证书处理”，RFC 38502004年7月。
[RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk, "Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)", RFC 4117, June 2005.
[RFC4117]Camarillo，G.，Burger，E.，Schulzrinne，H.，和A.van Wijk，“使用第三方呼叫控制（3pcc）的会话启动协议（SIP）中的代码转换服务调用”，RFC 41172005年6月。
[RFC5369] Camarillo, G., "Framework for Transcoding with the Session Initiation Protocol", RFC 5369, October 2008.
[RFC5369]Camarillo，G.“会话启动协议的代码转换框架”，RFC 5369，2008年10月。
[RFC5363] Camarillo, G. and A.B. Roach, "Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services", RFC 5363, October 2008.
[RFC5363]Camarillo，G.和A.B.Roach，“会话启动协议（SIP）URI列表服务的框架和安全注意事项”，RFC 5363，2008年10月。
[RFC5366] Camarillo, G. and A. Johnston, "Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)", RFC 5366, October 2008.
[RFC5366]Camarillo，G.和A.Johnston，“使用会话启动协议（SIP）中包含的请求列表建立会议”，RFC 5366，2008年10月。
[RFC4244] Barnes, M., Ed., "An Extension to the Session Initiation Protocol (SIP) for Request History Information", RFC 4244, November 2005.
[RFC4244]Barnes，M.，Ed.，“请求历史信息会话启动协议（SIP）的扩展”，RFC 4244，2005年11月。
[RFC4474] Peterson, J. and C. Jennings, "Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)", RFC 4474, August 2006.
[RFC4474]Peterson，J.和C.Jennings，“会话启动协议（SIP）中身份验证管理的增强”，RFC 4474，2006年8月。
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006.
[RFC4566]Handley，M.，Jacobson，V.，和C.Perkins，“SDP：会话描述协议”，RFC4566，2006年7月。
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A Session Initiation Protocol (SIP) Event Package for Conference State", RFC 4575, August 2006.
[RFC4575]Rosenberg，J.，Schulzrinne，H.，和O.Levin，Ed.，“会议状态的会话启动协议（SIP）事件包”，RFC 45752006年8月。
[RFC5360] Rosenberg, J., "A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)", RFC 5360, October 2008.
[RFC5360]Rosenberg，J.，“会话启动协议（SIP）中基于同意的通信框架”，RFC 5360，2008年10月。
Author's Address
作者地址
Gonzalo Camarillo Ericsson Hirsalantie 11 Jorvas 02420 Finland
Gonzalo Camarillo Ericsson Hirsalantie 11 Jorvas 02420芬兰
EMail: Gonzalo.Camarillo@ericsson.com
EMail: Gonzalo.Camarillo@ericsson.com
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