Internet Engineering Task Force (IETF)                           R. Mahy
Request for Comments: 5850                                  Unaffiliated
Category: Informational                                        R. Sparks
ISSN: 2070-1721                                                  Tekelec
                                                            J. Rosenberg
                                                               D. Petrie
                                                        A. Johnston, Ed.
                                                                May 2010
Internet Engineering Task Force (IETF)                           R. Mahy
Request for Comments: 5850                                  Unaffiliated
Category: Informational                                        R. Sparks
ISSN: 2070-1721                                                  Tekelec
                                                            J. Rosenberg
                                                               D. Petrie
                                                        A. Johnston, Ed.
                                                                May 2010

A Call Control and Multi-Party Usage Framework for the Session Initiation Protocol (SIP)




This document defines a framework and the requirements for call control and multi-party usage of the Session Initiation Protocol (SIP). To enable discussion of multi-party features and applications, we define an abstract call model for describing the media relationships required by many of these. The model and actions described here are specifically chosen to be independent of the SIP signaling and/or mixing approach chosen to actually set up the media relationships. In addition to its dialog manipulation aspect, this framework includes requirements for communicating related information and events such as conference and session state and session history. This framework also describes other goals that embody the spirit of SIP applications as used on the Internet such as the definition of primitives (not services), invoker and participant oriented primitives, signaling and mixing model independence, and others.


Status of This Memo


This document is not an Internet Standards Track specification; it is published for informational purposes.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 5741.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。并非IESG批准的所有文件都适用于任何级别的互联网标准;见RFC 5741第2节。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2010 IETF信托基金和确定为文件作者的人员。版权所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

本文件受BCP 78和IETF信托有关IETF文件的法律规定的约束(自本文件出版之日起生效。请仔细阅读这些文件,因为它们描述了您对本文件的权利和限制。从本文件中提取的代码组件必须包括信托法律条款第4.e节中所述的简化BSD许可证文本,并提供简化BSD许可证中所述的无担保。

This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this material may not have granted the IETF Trust the right to allow modifications of such material outside the IETF Standards Process. Without obtaining an adequate license from the person(s) controlling the copyright in such materials, this document may not be modified outside the IETF Standards Process, and derivative works of it may not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English.


Table of Contents


   1.  Motivation and Background  . . . . . . . . . . . . . . . . . .  4
   2.  Key Concepts . . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.1.  Conversation Space Model . . . . . . . . . . . . . . . . .  7
     2.2.  Relationship between Conversation Space, SIP Dialogs,
           and SIP Sessions . . . . . . . . . . . . . . . . . . . . .  8
     2.3.  Signaling Models . . . . . . . . . . . . . . . . . . . . .  9
     2.4.  Mixing Models  . . . . . . . . . . . . . . . . . . . . . . 10
       2.4.1.  Tightly Coupled  . . . . . . . . . . . . . . . . . . . 11
       2.4.2.  Loosely Coupled  . . . . . . . . . . . . . . . . . . . 12
     2.5.  Conveying Information and Events . . . . . . . . . . . . . 13
     2.6.  Componentization and Decomposition . . . . . . . . . . . . 15
       2.6.1.  Media Intermediaries . . . . . . . . . . . . . . . . . 15
       2.6.2.  Text-to-Speech and Automatic Speech Recognition  . . . 17
       2.6.3.  VoiceXML . . . . . . . . . . . . . . . . . . . . . . . 17
     2.7.  Use of URIs  . . . . . . . . . . . . . . . . . . . . . . . 18
       2.7.1.  Naming Users in SIP  . . . . . . . . . . . . . . . . . 19
       2.7.2.  Naming Services with SIP URIs  . . . . . . . . . . . . 20
     2.8.  Invoker Independence . . . . . . . . . . . . . . . . . . . 22
     2.9.  Billing Issues . . . . . . . . . . . . . . . . . . . . . . 23
   1.  Motivation and Background  . . . . . . . . . . . . . . . . . .  4
   2.  Key Concepts . . . . . . . . . . . . . . . . . . . . . . . . .  6
     2.1.  Conversation Space Model . . . . . . . . . . . . . . . . .  7
     2.2.  Relationship between Conversation Space, SIP Dialogs,
           and SIP Sessions . . . . . . . . . . . . . . . . . . . . .  8
     2.3.  Signaling Models . . . . . . . . . . . . . . . . . . . . .  9
     2.4.  Mixing Models  . . . . . . . . . . . . . . . . . . . . . . 10
       2.4.1.  Tightly Coupled  . . . . . . . . . . . . . . . . . . . 11
       2.4.2.  Loosely Coupled  . . . . . . . . . . . . . . . . . . . 12
     2.5.  Conveying Information and Events . . . . . . . . . . . . . 13
     2.6.  Componentization and Decomposition . . . . . . . . . . . . 15
       2.6.1.  Media Intermediaries . . . . . . . . . . . . . . . . . 15
       2.6.2.  Text-to-Speech and Automatic Speech Recognition  . . . 17
       2.6.3.  VoiceXML . . . . . . . . . . . . . . . . . . . . . . . 17
     2.7.  Use of URIs  . . . . . . . . . . . . . . . . . . . . . . . 18
       2.7.1.  Naming Users in SIP  . . . . . . . . . . . . . . . . . 19
       2.7.2.  Naming Services with SIP URIs  . . . . . . . . . . . . 20
     2.8.  Invoker Independence . . . . . . . . . . . . . . . . . . . 22
     2.9.  Billing Issues . . . . . . . . . . . . . . . . . . . . . . 23
   3.  Catalog of Call Control Actions and Sample Features  . . . . . 23
     3.1.  Remote Call Control Actions on Early Dialogs . . . . . . . 24
       3.1.1.  Remote Answer  . . . . . . . . . . . . . . . . . . . . 24
       3.1.2.  Remote Forward or Put  . . . . . . . . . . . . . . . . 24
       3.1.3.  Remote Busy or Error Out . . . . . . . . . . . . . . . 24
     3.2.  Remote Call Control Actions on Single Dialogs  . . . . . . 24
       3.2.1.  Remote Dial  . . . . . . . . . . . . . . . . . . . . . 24
       3.2.2.  Remote On and Off Hold . . . . . . . . . . . . . . . . 25
       3.2.3.  Remote Hangup  . . . . . . . . . . . . . . . . . . . . 25
     3.3.  Call Control Actions on Multiple Dialogs . . . . . . . . . 25
       3.3.1.  Transfer . . . . . . . . . . . . . . . . . . . . . . . 25
       3.3.2.  Take . . . . . . . . . . . . . . . . . . . . . . . . . 26
       3.3.3.  Add  . . . . . . . . . . . . . . . . . . . . . . . . . 27
       3.3.4.  Local Join . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.5.  Insert . . . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.6.  Split  . . . . . . . . . . . . . . . . . . . . . . . . 29
       3.3.7.  Near-Fork  . . . . . . . . . . . . . . . . . . . . . . 29
       3.3.8.  Far-Fork . . . . . . . . . . . . . . . . . . . . . . . 29
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 30
   Appendix A.    Example Features  . . . . . . . . . . . . . . . . . 32
   Appendix A.1.  Attended Transfer . . . . . . . . . . . . . . . . . 32
   Appendix A.2.  Auto Answer . . . . . . . . . . . . . . . . . . . . 32
   Appendix A.3.  Automatic Callback  . . . . . . . . . . . . . . . . 32
   Appendix A.4.  Barge-In  . . . . . . . . . . . . . . . . . . . . . 32
   Appendix A.5.  Blind Transfer  . . . . . . . . . . . . . . . . . . 32
   Appendix A.6.  Call Forwarding . . . . . . . . . . . . . . . . . . 33
   Appendix A.7.  Call Monitoring . . . . . . . . . . . . . . . . . . 33
   Appendix A.8.  Call Park . . . . . . . . . . . . . . . . . . . . . 33
   Appendix A.9.  Call Pickup . . . . . . . . . . . . . . . . . . . . 33
   Appendix A.10. Call Return . . . . . . . . . . . . . . . . . . . . 34
   Appendix A.11. Call Waiting  . . . . . . . . . . . . . . . . . . . 34
   Appendix A.12. Click-to-Dial . . . . . . . . . . . . . . . . . . . 34
   Appendix A.13. Conference Call . . . . . . . . . . . . . . . . . . 34
   Appendix A.14. Consultative Transfer . . . . . . . . . . . . . . . 34
   Appendix A.15. Distinctive Ring  . . . . . . . . . . . . . . . . . 35
   Appendix A.16. Do Not Disturb  . . . . . . . . . . . . . . . . . . 35
   Appendix A.17. Find-Me . . . . . . . . . . . . . . . . . . . . . . 35
   Appendix A.18. Hotline . . . . . . . . . . . . . . . . . . . . . . 35
   Appendix A.19. IM Conference Alerts  . . . . . . . . . . . . . . . 35
   Appendix A.20. Inbound Call Screening  . . . . . . . . . . . . . . 35
   Appendix A.21. Intercom  . . . . . . . . . . . . . . . . . . . . . 36
   Appendix A.22. Message Waiting . . . . . . . . . . . . . . . . . . 36
   Appendix A.23. Music on Hold . . . . . . . . . . . . . . . . . . . 36
   Appendix A.24. Outbound Call Screening . . . . . . . . . . . . . . 36
   Appendix A.25. Pre-Paid Calling  . . . . . . . . . . . . . . . . . 37
   Appendix A.26. Presence-Enabled Conferencing . . . . . . . . . . . 37
   Appendix A.27. Single Line Extension/Multiple Line Appearance  . . 37
   Appendix A.28. Speakerphone Paging . . . . . . . . . . . . . . . . 38
   3.  Catalog of Call Control Actions and Sample Features  . . . . . 23
     3.1.  Remote Call Control Actions on Early Dialogs . . . . . . . 24
       3.1.1.  Remote Answer  . . . . . . . . . . . . . . . . . . . . 24
       3.1.2.  Remote Forward or Put  . . . . . . . . . . . . . . . . 24
       3.1.3.  Remote Busy or Error Out . . . . . . . . . . . . . . . 24
     3.2.  Remote Call Control Actions on Single Dialogs  . . . . . . 24
       3.2.1.  Remote Dial  . . . . . . . . . . . . . . . . . . . . . 24
       3.2.2.  Remote On and Off Hold . . . . . . . . . . . . . . . . 25
       3.2.3.  Remote Hangup  . . . . . . . . . . . . . . . . . . . . 25
     3.3.  Call Control Actions on Multiple Dialogs . . . . . . . . . 25
       3.3.1.  Transfer . . . . . . . . . . . . . . . . . . . . . . . 25
       3.3.2.  Take . . . . . . . . . . . . . . . . . . . . . . . . . 26
       3.3.3.  Add  . . . . . . . . . . . . . . . . . . . . . . . . . 27
       3.3.4.  Local Join . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.5.  Insert . . . . . . . . . . . . . . . . . . . . . . . . 28
       3.3.6.  Split  . . . . . . . . . . . . . . . . . . . . . . . . 29
       3.3.7.  Near-Fork  . . . . . . . . . . . . . . . . . . . . . . 29
       3.3.8.  Far-Fork . . . . . . . . . . . . . . . . . . . . . . . 29
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 30
   Appendix A.    Example Features  . . . . . . . . . . . . . . . . . 32
   Appendix A.1.  Attended Transfer . . . . . . . . . . . . . . . . . 32
   Appendix A.2.  Auto Answer . . . . . . . . . . . . . . . . . . . . 32
   Appendix A.3.  Automatic Callback  . . . . . . . . . . . . . . . . 32
   Appendix A.4.  Barge-In  . . . . . . . . . . . . . . . . . . . . . 32
   Appendix A.5.  Blind Transfer  . . . . . . . . . . . . . . . . . . 32
   Appendix A.6.  Call Forwarding . . . . . . . . . . . . . . . . . . 33
   Appendix A.7.  Call Monitoring . . . . . . . . . . . . . . . . . . 33
   Appendix A.8.  Call Park . . . . . . . . . . . . . . . . . . . . . 33
   Appendix A.9.  Call Pickup . . . . . . . . . . . . . . . . . . . . 33
   Appendix A.10. Call Return . . . . . . . . . . . . . . . . . . . . 34
   Appendix A.11. Call Waiting  . . . . . . . . . . . . . . . . . . . 34
   Appendix A.12. Click-to-Dial . . . . . . . . . . . . . . . . . . . 34
   Appendix A.13. Conference Call . . . . . . . . . . . . . . . . . . 34
   Appendix A.14. Consultative Transfer . . . . . . . . . . . . . . . 34
   Appendix A.15. Distinctive Ring  . . . . . . . . . . . . . . . . . 35
   Appendix A.16. Do Not Disturb  . . . . . . . . . . . . . . . . . . 35
   Appendix A.17. Find-Me . . . . . . . . . . . . . . . . . . . . . . 35
   Appendix A.18. Hotline . . . . . . . . . . . . . . . . . . . . . . 35
   Appendix A.19. IM Conference Alerts  . . . . . . . . . . . . . . . 35
   Appendix A.20. Inbound Call Screening  . . . . . . . . . . . . . . 35
   Appendix A.21. Intercom  . . . . . . . . . . . . . . . . . . . . . 36
   Appendix A.22. Message Waiting . . . . . . . . . . . . . . . . . . 36
   Appendix A.23. Music on Hold . . . . . . . . . . . . . . . . . . . 36
   Appendix A.24. Outbound Call Screening . . . . . . . . . . . . . . 36
   Appendix A.25. Pre-Paid Calling  . . . . . . . . . . . . . . . . . 37
   Appendix A.26. Presence-Enabled Conferencing . . . . . . . . . . . 37
   Appendix A.27. Single Line Extension/Multiple Line Appearance  . . 37
   Appendix A.28. Speakerphone Paging . . . . . . . . . . . . . . . . 38
   Appendix A.29. Speed Dial  . . . . . . . . . . . . . . . . . . . . 38
   Appendix A.30. Voice Message Screening . . . . . . . . . . . . . . 38
   Appendix A.31. Voice Portal  . . . . . . . . . . . . . . . . . . . 39
   Appendix A.32. Voicemail . . . . . . . . . . . . . . . . . . . . . 40
   Appendix A.33. Whispered Call Waiting  . . . . . . . . . . . . . . 40
   Appendix B.    Acknowledgments . . . . . . . . . . . . . . . . . . 40
   5.  Informative References . . . . . . . . . . . . . . . . . . . . 40
   Appendix A.29. Speed Dial  . . . . . . . . . . . . . . . . . . . . 38
   Appendix A.30. Voice Message Screening . . . . . . . . . . . . . . 38
   Appendix A.31. Voice Portal  . . . . . . . . . . . . . . . . . . . 39
   Appendix A.32. Voicemail . . . . . . . . . . . . . . . . . . . . . 40
   Appendix A.33. Whispered Call Waiting  . . . . . . . . . . . . . . 40
   Appendix B.    Acknowledgments . . . . . . . . . . . . . . . . . . 40
   5.  Informative References . . . . . . . . . . . . . . . . . . . . 40
1. Motivation and Background
1. 动机和背景

The Session Initiation Protocol (SIP) [RFC3261] was defined for the initiation, maintenance, and termination of sessions or calls between one or more users. However, despite its origins as a large-scale multi-party conferencing protocol, SIP is used today primarily for point-to-point calls. This two-party configuration is the focus of the SIP specification and most of its extensions.


This document defines a framework and the requirements for call control and multi-party usage of SIP. Most multi-party operations manipulate SIP dialogs (also known as call legs) or SIP conference media policy to cause participants in a conversation to perceive specific media relationships. In other protocols that deal with the concept of calls, this manipulation is known as call control. In addition to its dialog or policy manipulation aspect, call control also includes communicating information and events related to manipulating calls, including information and events dealing with session state and history, conference state, user state, and even message state.


Based on input from the SIP community, the authors compiled the following set of goals for SIP call control and multi-party applications:


o Define primitives, not services. Allow for a handful of robust yet simple mechanisms that can be combined to deliver features and services. Throughout this document, we refer to these simple mechanisms as "primitives". Primitives should be sufficiently robust so that when they are combined with each other, they can be used to build lots of services. However, the goal is not to define a provably complete set of primitives. Note that while the IETF will NOT standardize behavior or services, it may define example services for informational purposes, as in service examples [RFC5359].

o 定义原语,而不是服务。允许使用一些健壮但简单的机制,这些机制可以组合起来提供功能和服务。在本文档中,我们将这些简单机制称为“原语”。原语应该足够健壮,这样当它们相互结合时,就可以用来构建大量的服务。然而,目标不是定义一组可证明的完整原语。请注意,尽管IETF不会标准化行为或服务,但它可能会定义示例服务以供参考,如服务示例[RFC5359]。

o Be participant oriented. The primitives should be designed to provide services that are oriented around the experience of the participants. The authors observe that end users of features and services usually don't care how a media relationship is set up.

o 以参与者为导向。原语应设计为提供面向参与者体验的服务。作者观察到,功能和服务的最终用户通常不关心如何建立媒体关系。

Their ultimate experience is only based on the resulting media and other externally visible characteristics.


o Be signaling model independent. Support both a central-control and a peer-to-peer feature invocation model (and combinations of the two). Baseline SIP already supports a centralized control model described in 3pcc (third party call control) [RFC3725], and the SIP community has expressed a great deal of interest in peer-to-peer or distributed call control using primitives such as those defined in REFER [RFC3515], Replaces [RFC3891], and Join [RFC3911].

o 独立于信令模型。支持中央控制和对等功能调用模型(以及两者的组合)。基线SIP已经支持3pcc(第三方呼叫控制)[RFC3725]中描述的集中式控制模型,SIP社区对使用原语(如参考[RFC3515]、替换[RFC3891]和加入[RFC3911]中定义的原语)的对等或分布式呼叫控制表示了极大的兴趣。

o Be mixing model independent. The bulk of interesting multi-party applications involve mixing or combining media from multiple participants. This mixing can be performed by one or more of the participants or by a centralized mixing resource. The experience of the participants should not depend on the mixing model used. While most examples in this document refer to audio mixing, the framework applies to any media type. In this context, a "mixer" refers to combining media of the same type in an appropriate, media-specific way. This is consistent with the model described in the SIP conferencing framework.

o 与模型无关。大量有趣的多方应用程序涉及混合或组合来自多个参与者的媒体。这种混合可以由一个或多个参与者或集中的混合资源执行。参与者的经验不应取决于所使用的混合模式。虽然本文档中的大多数示例都涉及音频混合,但该框架适用于任何媒体类型。在此上下文中,“混合器”指以适当的、特定于媒体的方式组合相同类型的媒体。这与SIP会议框架中描述的模型一致。

o Be invoker oriented. Only the user who invokes a feature or a service needs to know exactly which service is invoked or why. This is good because it allows new services to be created without requiring new primitives from all of the participants; and it allows for much simpler feature authorization policies, for example, when participation spans organizational boundaries. As discussed in Section 2.7, this also avoids exponential state explosion when combining features. The invoker only has to manage a user interface or application programming interface (API) to prevent local feature interactions. All the other participants simply need to manage the feature interactions of a much smaller number of primitives.

o 要面向调用方。只有调用功能或服务的用户才需要确切地知道调用了哪个服务或为什么。这很好,因为它允许创建新服务,而不需要来自所有参与者的新原语;并且它允许更简单的功能授权策略,例如,当参与跨越组织边界时。如第2.7节所述,这也避免了在组合特征时出现指数状态爆炸。调用方只需管理用户界面或应用程序编程接口(API),以防止本地功能交互。所有其他参与者只需要管理数量少得多的原语的特性交互。

o Primitives make full use of URIs (uniform resource identifiers). URIs are a very powerful mechanism for describing users and services. They represent a plentiful resource that can be extremely expressive and easily routed, translated, and manipulated -- even across organizational boundaries. URIs can contain special parameters and informational header fields that need only be relevant to the owner of the namespace (domain) of the URI. Just as a user who selects an http: URL need not understand the significance and organization of the web site it references, a user may encounter a SIP URI that translates into an email-style group alias, which plays a pre-recorded message or runs some complex call-handling logic. Note that while this may

o 原语充分利用URI(统一资源标识符)。URI是描述用户和服务的非常强大的机制。它们代表了一个丰富的资源,可以非常有表现力,并且可以轻松地路由、翻译和操纵——甚至可以跨越组织边界。URI可以包含特殊参数和信息头字段,这些字段只需要与URI的命名空间(域)的所有者相关。正如选择http:URL的用户不需要了解其引用的网站的意义和组织一样,用户可能会遇到一个SIP-URI,该URI转换为电子邮件样式的组别名,它播放预先录制的消息或运行一些复杂的呼叫处理逻辑。注意,虽然这可能

seem paradoxical to the previous goal, both goals can be satisfied by the same model.


o Make use of SIP header fields and SIP event packages to provide SIP entities with information about their environment. These should include information about the status/handling of dialogs on other user agents (UAs), information about the history of other contacts attempted prior to the current contact, the status of participants, the status of conferences, user presence information, and the status of messages.

o 使用SIP头字段和SIP事件包为SIP实体提供有关其环境的信息。这些信息应包括关于其他用户代理(UAs)对话框的状态/处理的信息、关于当前联系人之前尝试的其他联系人的历史记录的信息、参与者的状态、会议状态、用户状态信息以及消息状态。

o Encourage service decomposition, and design to make use of standard components using well-defined, simple interfaces. Sample components include a SIP mixer, recording service, announcement server, and voice-dialog server. (This is not an exhaustive list).

o 鼓励服务分解,并设计为使用定义良好、简单的接口使用标准组件。示例组件包括SIP混音器、录音服务、公告服务器和语音对话服务器。(这不是一份详尽的清单)。

o Include authentication, authorization, policy, logging, and accounting mechanisms to allow these primitives to be used safely among mutually untrusted participants. Some of these mechanisms may be used to assist in billing, but no specific billing system will be endorsed.

o 包括身份验证、授权、策略、日志和记帐机制,以允许在相互不信任的参与者之间安全地使用这些原语。其中一些机制可用于协助计费,但不会认可特定的计费系统。

o Permit graceful fallback to baseline SIP. Definitions for new SIP call control extensions/primitives must describe a graceful way to fallback to baseline SIP behavior. Support for one primitive must not imply support for another primitive.

o 允许正常回退到基线SIP。新SIP呼叫控制扩展/原语的定义必须描述回退到基线SIP行为的优雅方式。对一个原语的支持不能意味着对另一个原语的支持。

o Don't reinvent traditional models, such as the model used for the H.450 family of protocols, JTAPI (Java Telephony Application Programming Interface), or the CSTA (Computer-supported telecommunications applications) call model, as these other models do not share the design goals presented in this document.

o 不要重新发明传统模型,例如用于H.450协议系列的模型、JTAPI(Java电话应用程序编程接口)或CSTA(计算机支持的电信应用程序)调用模型,因为这些其他模型与本文档中介绍的设计目标不相同。

Note that the flexibility in this model does have some disadvantages in terms of interoperability. It is possible to build a call control feature in SIP using different combinations of primitives. For a discussion of the issues associated with this, see [BLISS-PROBLEM].


2. Key Concepts
2. 关键概念

This section introduces a number of key concepts that will be used to describe and explain various call control operations and services in the remainder of this document. This includes the conversation space model, signaling and mixing models, common components, and the use of URIs.


2.1. Conversation Space Model
2.1. 会话空间模型

This document introduces the concept of an abstract "conversation space" as a set of participants who believe they are all communicating among one another. Each conversation space contains one or more participants.


Participants are SIP UAs that send original media to or terminate and receive media from other members of the conversation space. Logically, every participant in the conversation space has access to all the media generated in that space (this is strictly true if all participants share a common media type). A SIP UA that does not contribute or consume any media is NOT a participant, nor is a UA that merely forwards, transcodes, mixes, or selects media originating elsewhere in the conversation space.

参与者是SIP UAs,向对话空间的其他成员发送原始媒体或终止并接收来自对话空间其他成员的媒体。从逻辑上讲,会话空间中的每个参与者都可以访问该空间中生成的所有媒体(如果所有参与者共享一种公共媒体类型,则严格来说这是正确的)。不贡献或使用任何媒体的SIP-UA不是参与者,也不是仅仅转发、转码、混合或选择源自会话空间中其他地方的媒体的UA。

Note that a conversation space consists of zero or more SIP calls or SIP conferences. A conversation space is similar to the definition of a "call" in some other call models.


Participants may represent human users or non-human users (referred to as robots or automatons in this document). Some participants may be hidden within a conversation space. Some examples of hidden participants include: robots that generate tones, images, or announcements during a conference to announce users arriving and departing, a human call center supervisor monitoring a conversation between a trainee and a customer, and robots that record media for training or archival purposes.


Participants may also be active or passive. Active participants are expected to be intelligent enough to leave a conversation space when they no longer desire to participate. (An attentive human participant is obviously active.) Some robotic participants (such as a voice-messaging system, an instant-messaging agent, or a voice-dialog system) may be active participants if they can leave the conversation space when there is no human interaction. Other robots (for example, our tone-generating robot from the previous example) are passive participants. A human participant "on hold" is passive.


An example diagram of a conversation space can be shown as a "bubble" or ovals, or as a "set" in curly or square bracket notation. Each set, oval, or bubble represents a conversation space. Hidden participants are shown in lowercase letters. Examples are given in Figure 1.


Note that while the term "conversation" usually applies to oral exchange of information, we apply the conversation space model to any media exchange between participants.


{ A , B } [ A , b, C, D ]


      .-.                 .---.
     /   \               /     \
    /  A  \             / A   b \
   (       )           (         )
    \  B  /             \ C   D /
     \   /               \     /
      '-'                 '---'
      .-.                 .---.
     /   \               /     \
    /  A  \             / A   b \
   (       )           (         )
    \  B  /             \ C   D /
     \   /               \     /
      '-'                 '---'

Figure 1. Conversation Spaces


2.2. Relationship between Conversation Space, SIP Dialogs, and SIP Sessions

2.2. 会话空间、SIP对话框和SIP会话之间的关系

In [RFC3261], a call is "an informal term that refers to some communication between peers, generally set up for the purposes of a multimedia conversation". The concept of a conversation space is needed because the SIP definition of call is not sufficiently precise for the purpose of describing the user experience of multi-party features.


Do any other definitions convey the correct meaning? SIP and SDP (Session Description Protocol) [RFC4566] both define a conference as "a multimedia session identified by a common session description". A session is defined as "a set of multimedia senders and receivers and the data streams flowing from senders to receivers". The definition of "call" in some call models is more similar to our definition of a conversation space.


Some examples of the relationship between conversation spaces, SIP dialogs, and SIP sessions are listed below. In each example, a human user will perceive that there is a single call.


o A simple two-party call is a single conversation space, a single session, and a single dialog.

o 简单的两方通话是一个单独的对话空间、一个单独的会话和一个单独的对话。

o A locally mixed three-way call is two sessions and two dialogs. It is also a single conversation space.

o 本地混合三路呼叫是两个会话和两个对话框。它也是一个单一的对话空间。

o A simple dial-in audio conference is a single conversation space, but is represented by as many dialogs and sessions as there are human participants.

o 一个简单的拨入式音频会议是一个单独的对话空间,但它所代表的对话和会话数量与人类参与者的数量相同。

o A multicast conference is a single conversation space, a single session, and as many dialogs as participants.

o 多播会议是一个单独的对话空间、一个单独的会话以及与参与者一样多的对话。

2.3. Signaling Models
2.3. 信号模型

Obviously, to make changes to a conversation space, you must be able to use SIP signaling to cause these changes. Specifically, there must be a way to manipulate SIP dialogs (call legs) to move participants into and out of conversation spaces. Although this is not as obvious, there also must be a way to manipulate SIP dialogs to include non-participant UAs that are otherwise involved in a conversation space (e.g., back-to-back user agents or B2BUAs, third party call control (3pcc) controllers, mixers, transcoders, translators, or relays).


Implementations may setup the media relationships described in the conversation space model using a centralized control model. One common way to implement this using SIP is known as third party call control (3pcc) and is described in 3pcc [RFC3725]. The 3pcc approach relies on only the following three primitive operations:


o Create a new dialog (INVITE)

o 创建新对话框(邀请)

o Modify a dialog (reINVITE)

o 修改对话框(重新邀请)

o Destroy a dialog (BYE)

o 销毁对话框(再见)

The main advantage of the 3pcc approach is that it only requires very basic SIP support from end systems to support call control features. As such, third party call control is a natural way to handle protocol conversion and mid-call features. It also has the advantage and disadvantage that new features can/must be implemented in one place only (the controller), and it neither requires enhanced client functionality nor takes advantage of it.


In addition, a peer-to-peer approach is discussed at length in this document. The primary drawback of the peer-to-peer model is additional complexity in the end system and authentication and management models. The benefits of the peer-to-peer model include:


o state remains at the edges,

o 国家仍然处于边缘,

o call signaling need only go through participants involved (there are no additional points of failure), and

o 呼叫信号只需经过相关参与者(没有其他故障点),以及

o peers may take advantage of end-to-end message integrity or encryption

o 对等方可以利用端到端消息完整性或加密

The peer-to-peer approach relies on additional "primitive" operations, some of which are identified here.


o Replace an existing dialog

o 替换现有对话框

o Join a new dialog with an existing dialog

o 将新对话框与现有对话框联接

o Locally perform media forking (multi-unicast)

o 本地执行媒体分叉(多单播)

o Ask another user agent (UA) to send a request on your behalf

o 请另一个用户代理(UA)代表您发送请求

The peer-to-peer approach also only results in a single SIP dialog, directly between the two UAs. The 3pcc approach results in two SIP dialogs, between each UA and the controller. As a result, the SIP features and extensions that will be used during the dialog are limited to the those understood by the controller. As a result, in a situation where both the UAs support an advanced SIP feature but the controller does not, the feature will not be able to be used.


Many of the features, primitives, and actions described in this document also require some type of media mixing, combining, or selection as described in the next section.


2.4. Mixing Models
2.4. 混合模型

SIP permits a variety of mixing models, which are discussed here briefly. This topic is discussed more thoroughly in the SIP conferencing framework [RFC4353] and [RFC4579]. SIP supports both tightly coupled and loosely coupled conferencing, although more sophisticated behavior is available in tightly coupled conferences. In a tightly coupled conference, a single SIP user agent (called the focus) has a direct dialog relationship with each participant (and may control non-participant user agents as well). The focus can authoritatively publish information about the character and participants in a conference. In a loosely coupled conference, there are no coordinated signaling relationships among the participants.


For brevity, only the two most popular conferencing models are significantly discussed in this document (local and centralized mixing). Applications of the conversation spaces model to loosely coupled multicast and distributed full unicast mesh conferences are left as an exercise for the reader. Note that a distributed full mesh conference can be used for basic conferences, but does not easily allow for more complex conferencing actions like splitting, merging, and sidebars.


Call control features should be designed to allow a mixer (local or centralized) to decide when to reduce a conference back to a two-party call, or drop all the participants (for example, if only two automatons are communicating). The actual heuristics used to release calls are beyond the scope of this document, but may depend on properties in the conversation space, such as the number of active, passive, or hidden participants and the send-only, receive-only, or send-and-receive orientation of various participants.


2.4.1. Tightly Coupled
2.4.1. 紧密耦合

Tightly coupled conferences utilize a central point for signaling and authentication known as a focus [RFC4353]. The actual media can be centrally mixed or distributed.

紧密耦合的会议利用一个中心点进行信令和身份验证,称为焦点[RFC4353]。实际介质可以集中混合或分布。 (Single) End System Mixing (单)端系统混合

The first model we call "end system mixing". In this model, user A calls user B, and they have a conversation. At some point later, A decides to conference in user C. To do this, A calls C, using a completely separate SIP call. This call uses a different Call-ID, different tags, etc. There is no call set up directly between B and C. No SIP extension or external signaling is needed. A merely decides to locally join two dialogs.


      B     C
       \   /
        \ /
      B     C
       \   /
        \ /

Figure 2. End System Mixing Example


In Figure 2, A receives media streams from both B and C, and mixes them. A sends a stream containing A's and C's streams to B, and a stream containing A's and B's streams to C. Basically, user A handles both signaling and media mixing.

在图2中,A从B和C接收媒体流,并混合它们。A向B发送包含A和C流的流,向C发送包含A和B流的流。基本上,用户A处理信令和媒体混合。 Centralized Mixing 集中搅拌

In a centralized mixing model, all participants have a pairwise SIP and media relationship with the mixer. Common applications of centralized mixing include ad hoc conferences and scheduled dial-in or dial-out conferences. In Figure 3 below, the mixer M receives and sends media to participants A, B, C, D, and E.


      B     C
       \   /
        \ /
         M --- A
        / \
       /   \
      D     E
      B     C
       \   /
        \ /
         M --- A
        / \
       /   \
      D     E

Figure 3. Centralized Mixing Example

图3。集中混合示例 Centralized Signaling, Distributed Media 集中式信令、分布式媒体

In this conferencing model, there is a centralized controller, as in the dial-in and dial-out cases. However, the centralized server handles signaling only. The media is still sent directly between participants, using either multicast or multi-unicast. Participants perform their own mixing. Multi-unicast is when a user sends multiple packets (one for each recipient, addressed to that recipient). This is referred to as a "Decentralized Multipoint Conference" in [H.323]. Full mesh media with centralized mixing is another approach.


2.4.2. Loosely Coupled
2.4.2. 松散耦合

In these models, there is no point of central control of SIP signaling. As in the "Centralized Signaling, Distributed Media" case above, all endpoints send media to all other endpoints. Consequently, every endpoint mixes their own media from all the other sources and sends their own media to every other participant.

在这些模型中,没有SIP信令的集中控制点。在上面的“集中式信令,分布式媒体”案例中,所有端点都向所有其他端点发送媒体。因此,每个端点从所有其他来源混合他们自己的媒体,并将他们自己的媒体发送给每个其他参与者。 Large-Scale Multicast Conferences 大规模多播会议

Large-scale multicast conferences were the original motivation for both the Session Description Protocol (SDP) [RFC4566] and SIP. In a large-scale multicast conference, one or more multicast addresses are allocated to the conference. Each participant joins those multicast groups and sends their media to those groups. Signaling is not sent to the multicast groups. The sole purpose of the signaling is to inform participants of which multicast groups to join. Large-scale multicast conferences are usually pre-arranged, with specific start and stop times. However, multicast conferences do not need to be pre-arranged, so long as a mechanism exists to dynamically obtain a multicast address.

大规模多播会议是会话描述协议(SDP)[RFC4566]和SIP的最初动机。在大规模多播会议中,一个或多个多播地址被分配给会议。每个参与者加入这些多播组并将其媒体发送到这些组。信令不会发送到多播组。信令的唯一目的是通知参与者要加入哪些多播组。大型多播会议通常是预先安排的,有特定的开始和停止时间。然而,只要存在动态获取多播地址的机制,多播会议就不需要预先安排。 Full Distributed Unicast Conferencing 全分布式单播会议

In this conferencing model, each participant has both a pairwise media relationship and a pairwise signaling relationship with every other participant (a full mesh). This model requires a mechanism to maintain a consistent view of distributed state across the group. This is a classic, hard problem in computer science. Also, this model does not scale well for large numbers of participants. For <n> participants, the number of media and signaling relationships is approximately n-squared. As a result, this model is not generally available in commercial implementations; to the contrary, it is primarily the topic of research or experimental implementations. Note that this model assumes peer-to-peer signaling.


2.5. Conveying Information and Events
2.5. 传达信息和事件

Participants should have access to information about the other participants in a conversation space so that this information can be rendered to a human user or processed by an automaton. Although some of this information may be available from the Request-URI or To, From, Contact, or other SIP header fields, another mechanism of reporting this information is necessary.


Many applications are driven by knowledge about the progress of calls and conferences. In general, these types of events allow for the construction of distributed applications, where the application requires information on dialog and conference state, but is not necessarily a co-resident with an endpoint user agent or conference server. For example, a focus involved in a conversation space may wish to provide URIs for conference status and/or conference/floor control.


The SIP Events architecture [RFC3265] defines general mechanisms for subscription to and notification of events within SIP networks. It introduces the notion of a package that is a specific "instantiation" of the events mechanism for a well-defined set of events.


Event packages are needed to provide the status of a user's dialogs, the status of conferences and their participants, user-presence information, the status of registrations, and the status of a user's messages. While this is not an exhaustive list, these are sufficient to enable the sample features described in this document.


The conference event package [RFC4575] allows users to subscribe to information about an entire tightly coupled SIP conference. Notifications convey information about the participants such as the SIP URI identifying each user, their status in the space (active, declined, departed), URIs to invoke other features (such as sidebar


conversations), links to other relevant information (such as floor-control policies), and if floor-control policies are in place, the user's floor-control status. For conversation spaces created from cascaded conferences, conversation state can be gathered from relevant foci and merged into a cohesive set of state.


The dialog package [RFC4235] provides information about all the dialogs the target user is maintaining, in which conversations the user is participating, and how these are correlated. Likewise, the registration package [RFC3680] provides notifications when contacts have changed for a specific address-of-record (AOR). The combination of these allows a user agent to learn about all conversations occurring for the entire registered contact set for an address-of-record.


Note that user presence in SIP [RFC3856] has a close relationship with these latter two event packages. It is fundamental to the presence model that the information used to obtain user presence is constructed from any number of different input sources. Examples of other such sources include calendaring information and uploads of presence documents. These two packages can be considered another mechanism that allows a presence agent to determine the presence state of the user. Specifically, a user presence server can act as a subscriber for the dialog and registration packages to obtain additional information that can be used to construct a presence document.


The multi-party architecture may also need to provide a mechanism to get information about the status/handling of a dialog (for example, information about the history of other contacts attempted prior to the current contact). Finally, the architecture should provide ample opportunities to present informational URIs that relate to calls, conversations, or dialogs in some way. For example, consider the SIP Call-Info header or Contact header fields returned in a 300-class response. Frequently, additional information about a call or dialog can be fetched via non-SIP URIs. For example, consider a web page for package tracking when calling a delivery company or a web page with related documentation when joining a dial-in conference. The use of URIs in the multi-party framework is discussed in more detail in Section 3.7.


Finally, the interaction of SIP with stimulus-signaling-based applications, which allow a user agent to interact with an application without knowledge of the semantics of that application, is discussed in the SIP application interaction framework [RFC5629]. Stimulus signaling can occur with a user interface running locally with the client, or with a remote user interface, through media streams. Stimulus signaling encompasses a wide range of mechanisms,


from clicking on hyperlinks, to pressing buttons, to traditional Dual-Tone Multi Frequency (DTMF) input. In all cases, stimulus signaling is supported through the use of markup languages, which play a key role in that framework.


2.6. Componentization and Decomposition
2.6. 组件化和分解

This framework proposes a decomposed component architecture with a very loose coupling of services and components. This means that a service (such as a conferencing server or an auto-attendant) need not be implemented as an actual server. Rather, these services can be built by combining a few basic components in straightforward or arbitrarily complex ways.


Since the components are easily deployed on separate boxes, by separate vendors, or even with separate providers, we achieve a separation of function that allows each piece to be developed in complete isolation. We can also reuse existing components for new applications. This allows rapid service creation, and the ability for services to be distributed across organizational domains anywhere in the Internet.


For many of these components, it is also desirable to discover their capabilities, for example, querying the ability of a mixer to host a 10-dialog conference or to reserve resources for a specific time. These actions could be provided in the form of URIs, provided there is an a priori means of understanding their semantics. For example, if there is a published dictionary of operations, a way to query the service for the available operations and the associated URIs, the URI can be the interface for providing these service operations. This concept is described in more detail in the context of dialog operations in Section 3.


2.6.1. Media Intermediaries
2.6.1. 媒体中介

Media intermediaries are not participants in any conversation space, although an entity that is also a media translator may also have a co-located participant component (for example, a mixer that also announces the arrival of a new participant; the announcement portion is a participant, but the mixer itself is not). Media intermediaries should be as transparent as possible to the end users -- offering a useful, fundamental service without getting in the way of new features implemented by participants. Some common media intermediaries are described below.

媒体中间人不是任何对话空间中的参与者,尽管同时也是媒体翻译人员的实体也可能具有共同定位的参与者组件(例如,也宣布新参与者到达的混合器;宣布部分是参与者,但混合器本身不是)。媒体中介应该对最终用户尽可能透明——提供有用的基本服务,而不妨碍参与者实现新功能。下面介绍一些常见的媒体中介。 Mixer 搅拌机

A SIP mixer is a component that combines media from all dialogs in the same conversation in a media-specific way. For example, the default combining for an audio conference might be an N-1 configuration, while a text mixer might interleave text messages on a per-line basis. More details about how to manipulate the media policy used by mixers is discussed in [XCON-CCMP].

SIP混合器是一个组件,它以特定于媒体的方式将来自同一对话中所有对话的媒体组合在一起。例如,音频会议的默认组合可能是N-1配置,而文本混合器可能在每行的基础上交错文本消息。[XCON-CCMP]中讨论了有关如何操作混音器使用的媒体策略的更多详细信息。 Transcoder 转码器

A transcoder translates media from one encoding or format to another (for example, GSM (Global System for Mobile communications) voice to G.711, MPEG2 to H.261, or text/html to text/plain), or from one media type to another (for example, text to speech). A more thorough discussion of transcoding is described in the SIP transcoding services invocation [RFC5369].

转码器将媒体从一种编码或格式转换为另一种编码或格式(例如,GSM(全球移动通信系统)语音转换为G.711,MPEG2转换为H.261,或text/html转换为text/plain),或从一种媒体类型转换为另一种媒体类型(例如,文本转换为语音)。SIP代码转换服务调用[RFC5369]中描述了对代码转换的更深入的讨论。 Media Relay 媒体转播

A media relay terminates media and simply forwards it to a new destination without changing the content in any way. Sometimes, media relays are used to provide source IP address anonymity, to facilitate middlebox traversal, or to provide a trusted entity where media can be forcefully disconnected.

媒体中继终止媒体并将其转发到新的目的地,而不以任何方式更改内容。有时,媒体中继用于提供源IP地址匿名性、促进中间盒遍历或提供可强制断开媒体连接的可信实体。 Queue Server 队列服务器

A queue server is a location where calls can be entered into one of several FIFO (first-in, first-out) queues. A queue server would subscribe to the presence of groups or individuals who are interested in its queues. When detecting that a user is available to service a queue, the server redirects or transfers the last call in the relevant queue to the available user. On a queue-by-queue basis, authorized users could also subscribe to the call state (dialog information) of calls within a queue. Authorized users could use this information to effectively pluck (take) a call out of the queue (for example, by sending an INVITE with a Replaces header to one of the user agents in the queue).

队列服务器是一个可以将呼叫输入多个FIFO(先进先出)队列之一的位置。队列服务器将订阅对其队列感兴趣的组或个人的存在。当检测到一个用户可以为队列提供服务时,服务器会将相关队列中的最后一个呼叫重定向或转移给可用用户。在逐个队列的基础上,授权用户还可以订阅队列中呼叫的呼叫状态(对话框信息)。授权用户可以使用此信息有效地从队列中提取(接受)呼叫(例如,通过向队列中的一个用户代理发送带有Replaces头的INVITE)。 Parking Place 停车场

A parking place is a location where calls can be terminated temporarily and then retrieved later. While a call is "parked", it can receive media "on hold" such as music, announcements, or advertisements. Such a service could be further decomposed such that announcements or music are handled by a separate component.

停车场是一个可以暂时终止呼叫,然后稍后再检索的位置。当电话处于“暂停”状态时,它可以接收“暂停”媒体,如音乐、公告或广告。这样的服务可以进一步分解,以便公告或音乐由单独的组件处理。 Announcements and Voice Dialogs 公告和语音对话

An announcement server is a server that can play digitized media (frequently audio), such as music or recorded speech. These servers are typically accessible via SIP, HTTP (Hyper Text Transport Protocol), or RTSP (Real-Time Streaming Protocol). An analogous service is a recording service that stores digitized media. A convention for specifying announcements in SIP URIs is described in [RFC4240]. Likewise, the same server could easily provide a service that records digitized media.

公告服务器是可以播放数字化媒体(通常是音频)的服务器,如音乐或录制的语音。这些服务器通常可以通过SIP、HTTP(超文本传输协议)或RTSP(实时流协议)访问。类似服务是存储数字化媒体的录制服务。[RFC4240]中描述了在SIP URI中指定公告的约定。同样,同一台服务器可以轻松提供记录数字化媒体的服务。

A "voice dialog" is a model of spoken interactive behavior between a human and an automaton that can include synthesized speech, digitized audio, recognition of spoken and DTMF key input, a recording of spoken input, and interaction with call control. Voice dialogs frequently consist of forms or menus. Forms present information and gather input; menus offer choices of what to do next.


Spoken dialogs are a basic building block of applications that use voice. Consider, for example, that a voicemail system, the conference-id and passcode collection system for a conferencing system, and complicated voice-portal applications all require a voice-dialog component.


2.6.2. Text-to-Speech and Automatic Speech Recognition
2.6.2. 文本到语音和自动语音识别

Text-to-speech (TTS) is a service that converts text into digitized audio. TTS is frequently integrated into other applications, but when separated as a component, it provides greater opportunity for broad reuse. Automatic Speech Recognition (ASR) is a service that attempts to decipher digitized speech based on a proposed grammar. Like TTS, ASR services can be embedded, or exposed so that many applications can take advantage of such services. A standardized (decomposed) interface to access standalone TTS and ASR services is currently being developed as described in [RFC4313].


2.6.3. VoiceXML
2.6.3. VoiceXML

VoiceXML is a W3C (World Wide Web Consortium) recommendation that was designed to give authors control over the spoken dialog between users and applications. The application and user take turns speaking: the application prompts the user, and the user in turn responds. Its major goal is to bring the advantages of web-based development and content delivery to interactive voice-response applications. We believe that VoiceXML represents the ideal partner for SIP in the development of distributed IVR (interactive voice response) servers. VoiceXML is an XML-based scripting language for describing IVR services at an abstract level. VoiceXML supports DTMF recognition,


speech recognition, text-to-speech, and the playing out of recorded media files. The results of the data collected from the user are passed to a controlling entity through an HTTP POST operation. The controller can then return another script, or terminate the interaction with the IVR server.

语音识别、文本到语音转换以及播放录制的媒体文件。从用户收集的数据的结果通过HTTP POST操作传递给控制实体。然后,控制器可以返回另一个脚本,或终止与IVR服务器的交互。

A VoiceXML server also need not be implemented as a monolithic server. Figure 4 shows a diagram of a VoiceXML browser that is split into media and non-media handling parts. The VoiceXML interpreter handles SIP dialog state and state within a VoiceXML document, and sends requests to the media component over another protocol.


                       |             |
                       | VoiceXML    |
                       | Interpreter |
                       | (signaling) |
                         ^          ^
                         |          |
                     SIP |          | RTSP
                         |          |
                         |          |
                         v          v
            +-------------+        +-------------+
            |             |        |             |
            |  SIP UA     |   RTP  | RTSP Server |
            |             |<------>|   (media)   |
            |             |        |             |
            +-------------+        +-------------+
                       |             |
                       | VoiceXML    |
                       | Interpreter |
                       | (signaling) |
                         ^          ^
                         |          |
                     SIP |          | RTSP
                         |          |
                         |          |
                         v          v
            +-------------+        +-------------+
            |             |        |             |
            |  SIP UA     |   RTP  | RTSP Server |
            |             |<------>|   (media)   |
            |             |        |             |
            +-------------+        +-------------+

Figure 4. Decomposed VoiceXML Server


2.7. Use of URIs
2.7. URI的使用

All naming in SIP uses URIs. URIs in SIP are used in a plethora of contexts: the Request-URI; Contact, To, From, and *-Info header fields; application/uri bodies; and embedded in email, web pages, instant messages, and ENUM records. The Request-URI identifies the user or service for which the call is destined.


SIP URIs embedded in informational SIP header fields, SIP bodies, and non-SIP content can also specify methods, special parameters, header fields, and even bodies. For example:

嵌入信息性SIP头字段、SIP正文和非SIP内容中的SIP URI还可以指定方法、特殊参数、头字段甚至正文。例如:;method=REFER?Refer-To=

Throughout this document, we discuss call control primitive operations. One of the biggest problems is defining how these operations may be invoked. There are a number of ways to do this. One way is to define the primitives in the protocol itself such that SIP methods (for example, REFER) or SIP header fields (for example, Replaces) indicate a specific call control action. Another way to invoke call control primitives is to define a specific Request-URI naming convention. Either these conventions must be shared between the client (the invoker) and the server, or published by or on behalf of the server. The former involves defining URI construction techniques (e.g., URI parameters and/or token conventions) as proposed in [RFC4240]. The latter technique usually involves discovering the URI via a SIP event package, a web page, a business card, or an instant message. Yet, another means to acquire the URIs is to define a dictionary of primitives with well-defined semantics and provide a means to query the named primitives and corresponding URIs that may be invoked on the service or dialogs.


2.7.1. Naming Users in SIP
2.7.1. SIP中的用户命名

An address-of-record, or public SIP address, is a SIP (or Secure SIP (SIPS)) URI that points to a domain with a location service that can map the URI to set of Contact URIs where the user might be available. Typically, the Contact URIs are populated via registration.


Address-of-Record Contacts

记录联系人地址 ->

Callee Capabilities [RFC3840] define a set of additional parameters to the Contact header field that define the characteristics of the user agent at the specified URI. For example, there is a mobility parameter that indicates whether the UA is fixed or mobile. When a user agent registers, it places these parameters in the Contact header fields to characterize the URIs it is registering. This allows a proxy for that domain to have information about the contact addresses for that user.

被调用者功能[RFC3840]为Contact header字段定义了一组附加参数,用于定义指定URI处用户代理的特征。例如,存在指示UA是固定的还是移动的移动参数。当用户代理注册时,它将这些参数放置在Contact header字段中,以描述它正在注册的URI。这允许该域的代理拥有有关该用户联系地址的信息。

When a caller sends a request, it can optionally request Caller Preferences [RFC3841] by including the Accept-Contact, Request-Disposition, and Reject-Contact header fields that request certain handling by the proxy in the target domain. These header fields contain preferences that describe the set of desired URIs to which the caller would like their request routed. The proxy in the target domain matches these preferences with the Contact characteristics originally registered by the target user. The target user can also


choose to run arbitrarily complex "Find-me" feature logic on a proxy in the target domain.


There is a strong asymmetry in how preferences for callers and callees can be presented to the network. While a caller takes an active role by initiating the request, the callee takes a passive role in waiting for requests. This motivates the use of callee-supplied scripts and caller preferences included in the call request. This asymmetry is also reflected in the appropriate relationship between caller and callee preferences. A server for a callee should respect the wishes of the caller to avoid certain locations, while the preferences among locations has to be the callee's choice, as it determines where, for example, the phone rings and whether the callee incurs mobile telephone charges for incoming calls.


SIP User Agent implementations are encouraged to make intelligent decisions based on the type of participants (active/passive, hidden, human/robot) in a conversation space. This information is conveyed via the dialog package or in a SIP header field parameter communicated using an appropriate SIP header field. For example, a music on hold service may take the sensible approach that if there are two or more unhidden participants, it should not provide hold music; or that it will not send hold music to robots.


Multiple participants in the same conversation space may represent the same human user. For example, the user may use one participant device for video, chat, and whiteboard media on a PC and another for audio media on a SIP phone. In this case, the address-of-record is the same for both user agents, but the Contacts are different. In this case, there is really only one human participant. In addition, human users may add robot participants that act on their behalf (for example, a call recording service or a calendar announcement reminder). Call control features in SIP should continue to function as expected in such an environment.


2.7.2. Naming Services with SIP URIs
2.7.2. 使用SIPURI命名服务

A critical piece of defining a session-level service that can be accessed by SIP is defining the naming of the resources within that service. This point cannot be overstated.


In the context of SIP control of application components, we take advantage of the fact that the left-hand side of a standard SIP URI is a user part. Most services may be thought of as user automatons that participate in SIP sessions. It naturally follows that the user part should be utilized as a service indicator.


For example, media servers commonly offer multiple services at a single host address. Use of the user part as a service indicator enables service consumers to direct their requests without ambiguity. It has the added benefit of enabling media services to register their availability with SIP Registrars just as any "real" SIP user would. This maintains consistency and provides enhanced flexibility in the deployment of media services in the network.


There has been much discussion about the potential for confusion if media-service URIs are not readily distinguishable from other types of SIP UAs. The use of a service namespace provides a mechanism to unambiguously identify standard interfaces while not constraining the development of private or experimental services.

如果媒体服务URI不能很容易地与其他类型的SIP UA区分开来,那么就有可能出现混淆,对此已经进行了很多讨论。服务名称空间的使用提供了一种明确标识标准接口的机制,同时不限制私有或实验服务的开发。

In SIP, the Request-URI identifies the user or service for which the call is destined. The great advantage of using URIs (specifically, the SIP Request-URI) as a service identifier comes because of the combination of two facts. First, unlike in the PSTN (Public Switched Telephone Network), where the namespace (dialable telephone numbers) is limited, URIs come from an infinite space. They are plentiful, and they are free. Secondly, the primary function of SIP is call routing through manipulations of the Request-URI. In the traditional SIP application, this URI represents a person. However, the URI can also represent a service, as we propose here. This means we can apply the routing services SIP provides to the routing of calls to services. The result -- the problem of service invocation and service location becomes a routing problem, for which SIP provides a scalable and flexible solution. Since there is such a vast namespace of services, we can explicitly name each service in a finely granular way. This allows the distribution of services across the network. For further discussion about services and SIP URIs, see RFC 3087 [RFC3087].

在SIP中,请求URI标识呼叫目的地的用户或服务。使用URI(特别是SIP请求URI)作为服务标识符的最大优势在于两个事实的结合。首先,与PSTN(公共交换电话网)中的命名空间(可拨打电话号码)有限不同,URI来自无限空间。它们很丰富,而且是免费的。第二,SIP的主要功能是通过操作请求URI进行呼叫路由。在传统的SIP应用程序中,此URI表示一个人。然而,URI也可以表示服务,正如我们在这里提出的那样。这意味着我们可以将SIP提供的路由服务应用于呼叫到服务的路由。结果——服务调用和服务位置问题变成了一个路由问题,SIP为此提供了一个可扩展且灵活的解决方案。由于存在如此庞大的服务名称空间,我们可以以精细粒度的方式显式命名每个服务。这允许跨网络分发服务。有关服务和SIP URI的进一步讨论,请参阅RFC 3087[RFC3087]。

Consider a conferencing service, where we have separated the names of ad hoc conferences from scheduled conferences, we can program proxies to route calls for ad hoc conferences to one set of servers and calls for scheduled ones to another, possibly even in a different provider. In fact, since each conference itself is given a URI, we can distribute conferences across servers, and easily guarantee that calls for the same conference always get routed to the same server. This is in stark contrast to conferences in the telephone network, where the equivalent of the URI -- the phone number -- is scarce. An entire conferencing provider generally has one or two numbers. Conference IDs must be obtained through IVR interactions with the caller or through a human attendant. This makes it difficult to distribute conferences across servers all over the network, since the PSTN routing only knows about the dialed number.


For more examples, consider the URI conventions of RFC 4240 [RFC4240] for media servers and RFC 4458 [RFC4458] for voicemail and IVR systems.

对于更多的例子,考虑用于媒体服务器的RFC 4240 [RFC4240]的URI约定和用于语音邮件和IVR系统的RFC 4458 [RFC4358]。

In practical applications, it is important that an invoker does not necessarily apply semantic rules to various URIs it did not create. Instead, it should allow any arbitrary string to be provisioned, and map the string to the desired behavior. The administrator of a service may choose to provision specific conventions or mnemonic strings, but the application should not require it. In any large installation, the system owner is likely to have preexisting rules for mnemonic URIs, and any attempt by an application to define its own rules may create a conflict. Implementations should allow an arbitrary mix of URIs from these schemes, or any other scheme that renders valid SIP URIs, rather than enforce only one particular scheme.


As we have shown, SIP URIs represent an ideal, flexible mechanism for describing and naming service resources, regardless of whether the resources are queues, conferences, voice dialogs, announcements, voicemail treatments, or phone features.


2.8. Invoker Independence
2.8. 调用程序独立性

With functional signaling, only the invoker of features in SIP needs to know exactly which feature they are invoking. One of the primary benefits of this approach is that combinations of functional features work in SIP call control without requiring complex feature-interaction matrices. For example, let us examine the combination of a "transfer" of a call that is "conferenced".


Alice calls Bob. Alice silently "conferences in" her robotic assistant Albert as a hidden party. Bob transfers Alice to Carol. If Bob asks Alice to Replace her leg with a new one to Carol, then both Alice and Albert should be communicating with Carol (transparently).


Using the peer-to-peer model, this combination of features works fine if A is doing local mixing (Alice replaces Bob's dialog with Carol's), or if A is using a central mixer (the mixer replaces Bob's dialog with Carol's). A clever implementation using the 3pcc model can generate similar results.


New extensions to the SIP Call Control Framework should attempt to preserve this property.


2.9. Billing Issues
2.9. 账单问题

Billing in the PSTN is typically based on who initiated a call. At the moment, billing in a SIP network is neither consistent with itself nor with the PSTN. (A billing model for SIP should allow for both PSTN-style billing and non-PSTN billing.) The example below demonstrates one such inconsistency.


Alice places a call to Bob. Alice then blind transfers Bob to Carol through a PSTN gateway. In current usage of REFER, Bob may be billed for a call he did not initiate (his UA originated the outgoing dialog, however). This is not necessarily a terrible thing, but it demonstrates a security concern (Bob must have appropriate local policy to prevent fraud). Also, Alice may wish to pay for Bob's session with Carol. There should be a way to signal this in SIP.


Likewise, a Replacement call may maintain the same billing relationship as a Replaced call, so if Alice first calls Carol, then asks Bob to Replace this call, Alice may continue to receive a bill.


Further work in SIP billing should define a way to set or discover the direction of billing.


3. Catalog of Call Control Actions and Sample Features
3. 调用控制操作和示例功能的目录

Call control actions can be categorized by the dialogs upon which they operate. The actions may involve a single or multiple dialogs. These dialogs can be early or established. Multiple dialogs may be related in a conversation space to form a conference or other interesting media topologies.


It should be noted that it is desirable to provide a means by which a party can discover the actions that may be performed on a dialog. The interested party may be independent or related to the dialogs. One means of accomplishing this is through the ability to define and obtain URIs for these actions, as described in Section 2.7.2.


Below are listed several call control "actions" that establish or modify dialogs and relate the participants in a conversation space. The names of the actions listed are for descriptive purposes only (they are not normative). This list of actions is not meant to be exhaustive.


In the examples, all actions are initiated by the user "Alice" represented by UA "A".


3.1. Remote Call Control Actions on Early Dialogs
3.1. 早期对话框上的远程呼叫控制操作

The following are a set of actions that may be performed on a single early dialog. These actions can be thought of as a set of remote control operations. For example, an automaton might perform the operation on behalf of a user. Alternatively, a user might use the remote control in the form of an application to perform the action on the early dialog of a UA that may be out of reach. All of these actions correspond to telling the UA how to respond to a request to establish an early dialog. These actions provide useful functionality for PDA-, PC-, and server-based applications that desire the ability to control a UA. A proposed mechanism for this type of functionality is described in remote call control [FEATURE-REF].


3.1.1. Remote Answer
3.1.1. 远程应答

A dialog is in some early dialog state such as 180 Ringing. It may be desirable to tell the UA to answer the dialog. That is, tell it to send a 200 OK response to establish the dialog.

对话框处于某些早期对话框状态,如180响。可能需要通知UA回答该对话框。也就是说,告诉它发送200 OK响应以建立对话框。

3.1.2. Remote Forward or Put
3.1.2. 远程提出或提出

It may be desirable to tell the UA to respond with a 3xx class response to forward an early dialog to another UA.


3.1.3. Remote Busy or Error Out
3.1.3. 远程忙或出错

It may be desirable to instruct the UA to send an error response such as 486 Busy Here.

可能需要指示UA发送诸如486 Busy之类的错误响应。

3.2. Remote Call Control Actions on Single Dialogs
3.2. 单个对话框上的远程调用控制操作

There is another useful set of actions that operate on a single established dialog. These operations are useful in building productivity applications for aiding users in controlling their phones. For example, a Customer Relationship Management (CRM) application that sets up calls for a user eliminating the need for the user to actually enter an address. These operations can also be thought of as remote control actions. A proposed mechanism for this type of functionality is described in remote call control [FEATURE-REF].


3.2.1. Remote Dial
3.2.1. 远程拨号

This action instructs the UA to initiate a dialog. This action can be performed using the REFER method.


3.2.2. Remote On and Off Hold
3.2.2. 远程开/关保持

This action instructs the UA to put an established dialog on hold. Though this operation can conceptually be performed with the REFER method, there are no semantics defined as to what the referred party should do with the SDP. There is no way to distinguish between the desire to go on or off hold on a per-media stream basis.


3.2.3. Remote Hangup
3.2.3. 远程挂断

This action instructs the UA to terminate an early or established dialog. A REFER request with the following Refer-To URI and Target-Dialog header field [RFC4538] performs this action. Note: this example does not show the full set of header fields.


   REFER SIP/2.0
   Target-Dialog: 13413098;local-tag=879738;remote-tag=023214
   REFER SIP/2.0
   Target-Dialog: 13413098;local-tag=879738;remote-tag=023214
3.3. Call Control Actions on Multiple Dialogs
3.3. 在多个对话框上调用控制操作

These actions apply to a set of related dialogs.


3.3.1. Transfer
3.3.1. 转移

This section describes how call transfer can be achieved using centralized (3pcc) and peer-to-peer (REFER) approaches.


The conversation space changes as follows:


    before            after
   { A , B }  -->   { C , B }
    before            after
   { A , B }  -->   { C , B }

A replaces itself with C.


To make this happen using the peer-to-peer approach, "A" would send two SIP requests. A shorthand for those requests is shown below:




To make this happen using the 3pcc approach instead, the controller sends requests represented by the shorthand below:




Features enabled by this action:


- blind transfer - transfer to a central mixer (some type of conference or forking) - transfer to park server (park) - transfer to music on hold or announcement server - transfer to a "queue" - transfer to a service (such as voice-dialog service) - transition from local mixer to central mixer

- 盲传输-传输到中央混音器(某种类型的会议或分叉)-传输到公园服务器(公园)-传输到音乐保留或公告服务器-传输到“队列”-传输到服务(如语音对话服务)-从本地混音器转换到中央混音器

This action is frequently referred to as "completing an attended transfer". It is described in more detail in [RFC5589].


Note that if a transfer requires URI hiding or privacy, then the 3pcc approach can more easily implement this. For example, if the URI of C needs to be hidden from B, then the use of 3pcc helps accomplish this.


3.3.2. Take
3.3.2. 拿

The conversation space changes as follows:


   { B , C } --> { B , A }
   { B , C } --> { B , A }

A forcibly replaces C with itself. In most uses of this primitive, A is just "un-replacing" itself.


Using the peer-to-peer approach, "A" sends:


    INVITE B  Replaces: <dialog between B and C>
    INVITE B  Replaces: <dialog between B and C>

Using the 3pcc approach (all requests sent from controller):




Features enabled by this action:


- transferee completes an attended transfer - retrieve from central mixer (not recommended) - retrieve from music on hold or park - retrieve from queue - call center take - voice portal resuming ownership of a call it originated - answering-machine style screening (pickup) - pickup of a ringing call (i.e., early dialog)

- 受让人完成有人值守转接-从中央调音台取回(不推荐)-从暂停音乐或停车场取回-从队列取回-呼叫中心取回-语音门户恢复其发起呼叫的所有权-应答机式筛选(取回)-接听来电(即,提前对话)

Note that pick up of a ringing call has perhaps some interesting additional requirements. First of all, it is an early dialog as opposed to an established dialog. Secondly, the party that is to pick up the call may only wish to do so only while it is an early dialog. That is in the race condition where the ringing UA accepts just before it receives signaling from the party wishing to take the call, the taking party wishes to yield or cancel the take. The goal is to avoid yanking an answered call from the called party.


This action is described in Replaces [RFC3891] and in [RFC5589].


3.3.3. Add
3.3.3. 添加

Note that the following four actions are described in [RFC4579].


This is merely adding a participant to a SIP conference. The conversation space changes as follows:


   { A , B } --> { A , B , C }
   { A , B } --> { A , B , C }

A adds C to the conversation.


Using the peer-to-peer approach, adding a party using local mixing requires no signaling. To transition from a two-party call or a locally mixed conference to central mixing, A could send the following requests:


REFER B Refer-To: conference-URI INVITE conference-URI BYE B


To add a party to a conference:


REFER C Refer-To: conference-URI or REFER conference-URI Refer-To: C


Using the 3pcc approach to transition to centrally mixed, the controller would send:


INVITE mixer leg 1 (w/SDP of A) INVITE mixer leg 2 (w/SDP of B) INVITE C (late SDP) reINVITE A (w/SDP of mixer leg 1) reINVITE B (w/SDP of mixer leg 2) INVITE mixer leg3 (w/SDP of C)


To add a party to a SIP conference:


INVITE C (late SDP) INVITE conference-URI (w/SDP of C)

邀请C(后期SDP)邀请会议URI(w/SDP of C)

Features enabled:


- standard conference feature - call recording - answering-machine style screening (screening)

- 标准会议功能-通话录音-答录机式屏蔽(屏蔽)

3.3.4. Local Join
3.3.4. 本地连接

The conversation space changes like this:


   { A , B } , { A , C }  -->  { A , B , C }
   { A , B } , { A , C }  -->  { A , B , C }

or like this


   { A , B } , { C , D }  -->  { A , B , C , D }
   { A , B } , { C , D }  -->  { A , B , C , D }

A takes two conversation spaces and joins them together into a single space.


Using the peer-to-peer approach, A can mix locally, or REFER the participants of both conversation spaces to the same central mixer (as in Section 3.3.5).


For the 3pcc approach, the call flows for inserting participants, and joining and splitting conversation spaces are tedious yet straightforward, so these are left as an exercise for the reader.


Features enabled:


- standard conference feature - leaving a sidebar to rejoin a larger conference

- 标准会议功能-留下侧边栏重新加入更大的会议

3.3.5. Insert
3.3.5. 插入

The conversation space changes like this:


   { B , C } --> { A , B , C }
   { B , C } --> { A , B , C }

A inserts itself into a conversation space.


A proposed mechanism for signaling this using the peer-to-peer approach is to send a new header field in an INVITE with "joining" [RFC3911] semantics. For example:


   INVITE B Join: <dialog id of B and C>
   INVITE B Join: <dialog id of B and C>

If B accepted the INVITE, B would accept responsibility to set up the dialogs and mixing necessary (for example, to mix locally or to transfer the participants to a central mixer).


Features enabled:


- barge-in - call center monitoring - call recording

- 驳船进港-呼叫中心监控-呼叫记录

3.3.6. Split
3.3.6. 分裂
   { A , B , C , D } --> { A , B } , { C , D }
   { A , B , C , D } --> { A , B } , { C , D }

If using a central conference with peer-to-peer


REFER C Refer-To: conference-URI (new URI) REFER D Refer-To: conference-URI (new URI) BYE C BYE D


Features enabled:


- sidebar conversations during a larger conference

- 大型会议期间的侧边栏对话

3.3.7. Near-Fork
3.3.7. 近叉

A participates in two conversation spaces simultaneously:


   { A, B } --> { B , A } & { A , C }
   { A, B } --> { B , A } & { A , C }

A is a participant in two conversation spaces such that A sends the same media to both spaces, and renders media from both spaces, presumably by mixing or rendering the media from both. We can define that A is the "anchor" point for both forks, each of which is a separate conversation space.


This action is purely local implementation (it requires no special signaling). Local features such as switching calls between the background and foreground are possible using this media relationship.


3.3.8. Far-Fork
3.3.8. 远叉

The conversation space diagram.


   { A, B } --> { A , B } & { B , C }
   { A, B } --> { A , B } & { B , C }

A requests B to be the "anchor" of two conversation spaces.


This is easily set up by creating a conference with two sub-conferences and setting the media policy appropriately such that B is a participant in both. Media forking can also be set up using 3pcc, as described in Section 5.1 of RFC 3264 [RFC3264] (an offer/answer model for SDP). The session descriptions for forking are quite complex. Controllers should verify that endpoints can handle forked media, for example, using prior configuration.

通过创建一个包含两个子会议的会议,并适当设置媒体策略,使B同时参与这两个会议,就可以轻松地进行设置。也可以使用3pcc设置媒体分叉,如RFC 3264[RFC3264](SDP的提供/应答模型)第5.1节所述。分叉的会话描述相当复杂。控制器应验证端点是否可以处理分叉介质,例如,使用先前的配置。

Features enabled:


- barge-in - voice-portal services - whisper - key word detection - sending DTMF somewhere else

- 驳入-语音门户服务-耳语-关键字检测-将DTMF发送到其他地方

4. Security Considerations
4. 安全考虑

Call control primitives provide a powerful set of features that can be dangerous in the hands of an attacker. To complicate matters, call control primitives are likely to be automatically authorized without direct human oversight.


The class of attacks that are possible using these tools includes the ability to eavesdrop on calls, disconnect calls, redirect calls, render irritating content (including ringing) at a user agent, cause an action that has billing consequences, subvert billing (theft-of-service), and obtain private information. Call control extensions must take extra care to describe how these attacks will be prevented.


We can also make some general observations about authorization and trust with respect to call control. The security model is dramatically dependent on the signaling model chosen (see Section 2.3)


Let us first examine the security model used in the 3pcc approach. All signaling goes through the controller, which is a trusted entity. Traditional SIP authentication and hop-by-hop encryption and message integrity work fine in this environment, but end-to-end encryption and message integrity may not be possible.


When using the peer-to-peer approach, call control actions and primitives can be legitimately initiated by a) an existing participant in the conversation space, b) a former participant in the conversation space, or c) an entity trusted by one of the participants. For example, a participant always initiates a


transfer; a retrieve from park (a take) is initiated on behalf of a former participant, and a barge-in (insert or far-fork) is initiated by a trusted entity (an operator, for example).

转移从停车场取回(take)是代表前参与者发起的,而驳船进港(insert或far fork)是由可信实体(例如,运营商)发起的。

Authenticating requests by an existing participant or a trusted entity can be done with baseline SIP mechanisms. In the case of features initiated by a former participant, these should be protected against replay attacks, e.g., by using a unique name or identifier per invocation. The Replaces header field exhibits this behavior as a by-product of its operation (once a Replaces operation is successful, the dialog being Replaced no longer exists). These credentials may, for example, need to be passed transitively or fetched in an event body.

现有参与者或受信任实体的身份验证请求可以通过基线SIP机制完成。对于前参与者发起的功能,应保护这些功能免受重播攻击,例如,每次调用使用唯一的名称或标识符。Replaces header字段将此行为显示为其操作的副产品(一旦Replaces操作成功,被替换的对话框将不再存在)。例如,这些凭证可能需要传递或在事件体中获取。

To authorize call control primitives that trigger special behavior (such as an INVITE with Replaces or Join semantics), the receiving user agent may have trouble finding appropriate credentials with which to challenge or authorize the request, as the sender may be completely unknown to the receiver, except through the introduction of a third party. These credentials need to be passed transitively in some way or fetched in an event body, for example.


Standard SIP privacy and anonymity mechanisms such as [RFC3323] and [RFC3325] used during SIP session establishment apply equally well to SIP call control operations. SIP call control mechanisms should address privacy and anonymity issues associated with that operation. For example, privacy during a transfer operation using REFER is discussed in Section 7.2 of [RFC5589]


Appendix A. Example Features

Primitives are defined in terms of their ability to provide features. These example features should require an amply robust set of services to demonstrate a useful set of primitives. They are described here briefly. Note that the descriptions of these features are non-normative. Note also that this document describes a mixture of both features originating in the world of telephones and features that are clearly Internet oriented.


Appendix A.1. Attended Transfer


In Attended Transfer [RFC5589], the transferring party establishes a session with the transfer target before completing the transfer.


Appendix A.2. Auto Answer


In Auto Answer, calls to a certain address or URI answer immediately via a speakerphone. The Answer-Mode header field [RFC5373] can be used for this feature.


Appendix A.3. Automatic Callback


In Automatic Callback [RFC5359], Alice calls Bob, but Bob is busy. Alice would like Bob to call her automatically when he is available. When Bob hangs up, Alice's phone rings. When Alice answers, Bob's phone rings. Bob answers and they talk.


Appendix A.4. Barge-In


In Barge-in, Carol interrupts Alice who has an in-progress call with Bob. In some variations, Alice forcibly joins a new conversation with Carol, in other variations, all three parties are placed in the same conversation (basically a three-way conference). Barge-in works the same as call monitoring except that it must indicate that the send media stream be mixed so that all of the other parties can hear the stream from the UA that is barging in.


Appendix A.5. Blind Transfer


In Blind Transfer [RFC5589], Alice is in a conversation with Bob. Alice asks Bob to contact Carol, but makes no attempt to contact Carol independently. In many implementations, Alice does not verify Bob's success or failure in contacting Carol.


Appendix A.6. Call Forwarding


In call forwarding [RFC5359], before a dialog is accepted, it is redirected to another location, for example, because the originally intended recipient is busy, does not answer, is disconnected from the network, or has configured all requests to go elsewhere.


Appendix A.7. Call Monitoring


Call monitoring is a Join operation [RFC3911]. For example, a call center supervisor joins an in-progress call for monitoring purposes. The monitoring UA sends a Join to the dialog to which it wants to listen. It is able to discover the dialog via the dialog state on the monitored UA. The monitoring UA sends SDP in the INVITE that indicates receive-only media. As the UA is only monitoring, it does not matter whether the UA indicates it wishes the send stream to be mixed or point to point.


Appendix A.8. Call Park


In Call Park [RFC5359], a participant parks a call (essentially puts the call on hold), and then retrieves it at a later time (typically from another location). Call park requires the ability to put a dialog some place, advertise it to users in a pickup group, and to uniquely identify it in a means that can be communicated (including human voice). The dialog can be held locally on the UA parking the dialog or alternatively transferred to the park service for the pickup group. The parked dialog then needs to be labeled (e.g., orbit 12) in a way that can be communicated to the party that is to pick up the call. The UAs in the pickup group discover the parked dialog(s) via the dialog package from the park service. If the dialog is parked locally, the park service merely aggregates the parked call states from the set of UAs in the pickup group.


Appendix A.9. Call Pickup


There are two different features that are called Call Pickup [RFC5359]. The first is the pickup of a parked dialog. The UA from which the dialog is to be picked up subscribes to the dialog state of the park service or the UA that has locally parked the dialog. Dialogs that are parked should be labeled with an identifier. The labels are used by the UA to allow the user to indicate which dialog is to be picked up. The UA picking up the call invoked the URI in the call state that is labeled as replace-remote.

有两种不同的功能称为呼叫拾取[RFC5359]。第一个是拾取一个停驻的对话框。从中拾取对话框的UA订阅停车服务的对话框状态或已在本地停放该对话框的UA。停驻的对话框应标有标识符。UA使用标签来允许用户指示要拾取的对话框。拾取呼叫的UA在标记为replace remote的呼叫状态下调用URI。

The other call pickup feature involves picking up an early dialog (typically ringing). A party picks up a call that was ringing at another location. One variation allows the caller to choose which


location, another variation just picks up any call in that user's "pickup group". This feature uses some of the same primitives as the pickup of a parked call. The call state of the UA ringing phone is advertised using the dialog package. The UA that is to pick up the early dialog subscribes either directly to the ringing UA or to a service aggregating the states for UAs in the pickup group. The call state identifies early dialogs. The UA uses the call state(s) to help the user choose which early dialog is to be picked up. The UA then invokes the URI in the call state labeled as replace-remote.

位置,另一个变体只是拾取该用户“拾取组”中的任何呼叫。此功能使用与接听停驻呼叫相同的一些原语。UA振铃电话的呼叫状态通过对话框包公布。要拾取早期对话框的UA直接订阅振铃UA或订阅聚合拾取组中UAs状态的服务。调用状态标识早期对话框。UA使用呼叫状态帮助用户选择要拾取的早期对话。UA然后在标记为replace remote的调用状态中调用URI。

Appendix A.10. Call Return


In Call Return, Alice calls Bob. Bob misses the call or is disconnected before he is finished talking to Alice. Bob invokes Call return, which calls Alice, even if Alice did not provide her real identity or location to Bob.

在回叫时,爱丽丝打电话给鲍勃。鲍勃没有接到电话,或者在与爱丽丝通话结束之前就断开了连接。Bob调用Call return,调用Alice,即使Alice没有向Bob提供她的真实身份或位置。

Appendix A.11. Call Waiting


In Call Waiting, Alice is in a call, then receives another call. Alice can place the first call on hold, and talk with the other caller. She can typically switch back and forth between the callers.

在Call Waiting中,Alice正在接听一个电话,然后接到另一个电话。Alice可以将第一个呼叫挂起,并与另一个呼叫方通话。她通常可以在呼叫者之间来回切换。

Appendix A.12. Click-to-Dial


In Click-to-Dial [RFC5359], Alice looks in her company directory for Bob. When she finds Bob, she clicks on a URI to call him. Her phone rings (or possibly answers automatically), and when she answers, Bob's phone rings. The application or server that hosts the Click-to-Dial application captures the URI to be dialed and can set up the call using 3pcc or can send a REFER request to the UA that is to dial the address. As users sometimes change their mind or wish to give up listing to a ringing or voicemail answered phone, this application illustrates the need to also have the ability to remotely hangup a call.


Appendix A.13. Conference Call


In a Conference Call [RFC4579], there are three or more active, visible participants in the same conversation space.


Appendix A.14. Consultative Transfer


In Consultative Transfer [RFC5589], the transferring party establishes a session with the target and mixes both sessions together so that all three parties can participate, then disconnects leaving the transferee and transfer target with an active session.


Appendix A.15. Distinctive Ring


In Distinctive Ring, incoming calls have different ring cadences or sample sounds depending on the From party, the To party, or other factors. The target UA either makes a local decision based on information in an incoming INVITE (To, From, Contact, Request-URI) or trusts an Alert-Info header field [RFC3261] provided by the caller or inserted by a trusted proxy. In the latter case, the UA fetches the content described in the URI (typically via HTTP) and renders it to the user.


Appendix A.16. Do Not Disturb


In Do Not Disturb, Alice selects the Do Not Disturb option. Calls to her either ring briefly or not at all and are forwarded elsewhere. Some variations allow specially authorized callers to override this feature and ring Alice anyway. Do Not Disturb is best implemented in SIP using presence [RFC3856].


Appendix A.17. Find-Me


In Find-Me, Alice sets up complicated rules for how she can be reached (possibly using CPL (Call Processing Language) [RFC3880], presence [RFC3856], or other factors). When Bob calls Alice, his call is eventually routed to a temporary Contact where Alice happens to be available.

在Find Me中,Alice为如何联系她建立了复杂的规则(可能使用CPL(呼叫处理语言)[RFC3880]、状态[RFC3856]或其他因素)。当鲍勃给爱丽丝打电话时,他的电话最终被转接到一个临时联系人那里,爱丽丝正好在那里。

Appendix A.18. Hotline


In Hotline, Alice picks up a phone and is immediately connected to the technical support hotline, for example. Hotline is also sometimes known as a Ringdown line.


Appendix A.19. IM Conference Alerts


In IM Conference Alerts, a user receives a notification as an instant message whenever someone joins a conference in which they are already a participant.


Appendix A.20. Inbound Call Screening


In Inbound Call Screening, Alice doesn't want to receive calls from Matt. Inbound Screening prevents Matt from disturbing Alice. In some variations, this works even if Matt hides his identity.


Appendix A.21. Intercom


In Intercom, Alice typically presses a button on a phone that immediately connects to another user or phone and causes that phone to play her voice over its speaker. Some variations immediately set up two-way communications, other variations require another button to be pressed to enable a two-way conversation. The UA initiates a dialog using INVITE and the Answer-Mode: Auto header field as described in [RFC5373]. The called UA accepts the INVITE with a 200 OK and automatically enables the speakerphone.

在对讲机中,Alice通常会按下手机上的一个按钮,立即连接到另一个用户或手机,并使该手机通过扬声器播放她的声音。有些变体会立即建立双向通信,而另一些变体则需要按下另一个按钮以实现双向对话。UA使用[RFC5373]中所述的邀请和应答模式:自动标题字段启动对话框。被呼叫的UA以200 OK接受邀请,并自动启用扬声器。

Alternatively, this can be a local decision for the UA to auto answer based upon called-party identification.


Appendix A.22. Message Waiting


In Message Waiting [RFC3842], Bob calls Alice when she has stepped away from her phone. When she returns, a visible or audible indicator conveys that someone has left her a voicemail message. The message waiting indication may also convey how many messages are waiting, from whom, at what time, and other useful pieces of information.


Appendix A.23. Music on Hold


In Music on Hold [RFC5359], when Alice places a call with Bob on hold, it replaces its audio with streaming content such as music, announcements, or advertisements. Music on hold can be implemented a number of ways. One way is to transfer the held call to a holding service. When the UA wishes to take the call off hold, it basically performs a take on the call from the holding service. This involves subscribing to call state on the holding service and then invoking the URI in the call state labeled as replace-remote.

在Music on Hold[RFC5359]中,当Alice与Bob通话时,它会将其音频替换为流媒体内容,如音乐、公告或广告。暂停音乐可以通过多种方式实现。一种方法是将保留的呼叫转移到保留服务。当UA希望接收呼叫保持时,它基本上执行从等待服务接收呼叫。这涉及到订阅保持服务上的调用状态,然后在标记为replace remote的调用状态中调用URI。

Alternatively, music on hold can be performed as a local mixing operation. The UA holding the call can mix in the music from the music service via RTP (i.e., an additional dialog) or RTSP or other streaming media source. This approach is simpler (i.e., the held dialog does not move so there is less chance of loosing them) from a protocol perspective, however it does use more LAN bandwidth and resources on the UA.


Appendix A.24. Outbound Call Screening


In Outbound Call Screening, Alice is paged and unknowingly calls a PSTN pay-service telephone number in the Caribbean, but local policy blocks her call, and possibly informs her why.


Appendix A.25. Pre-Paid Calling


In Pre-paid Calling, Alice pays for a certain currency or unit amount of calling value. When she places a call, she provides her account number somehow. If her account runs out of calling value during a call, her call is disconnected or redirected to a service where she can purchase more calling value.


For prepaid calling, the user's media always passes through a device that is trusted by the pre-paid provider. This may be the other endpoint (for example, a PSTN gateway). In either case, an intermediary proxy or B2BUA can periodically verify the amount of time available on the pre-paid account, and use the session-timer extension to cause the trusted endpoint (gateway) or intermediary (media relay) to send a reINVITE before that time runs out. During the reINVITE, the SIP intermediary can re-verify the account and insert another session-timer header field.


Note that while most pre-paid systems on the PSTN use an IVR to collect the account number and destination, this isn't strictly necessary for a SIP-originated prepaid call. SIP requests and SIP URIs are sufficiently expressive to convey the final destination, the provider of the prepaid service, the location from which the user is calling, and the prepaid account they want to use. If a pre-paid IVR is used, the mechanism described below (Voice Portals) can be combined as well.

请注意,虽然PSTN上的大多数预付费系统使用IVR来收集帐号和目的地,但对于SIP发起的预付费呼叫来说,这并不是绝对必要的。SIP请求和SIP URI具有足够的表达能力,能够传达最终目的地、预付费服务的提供者、用户呼叫的位置以及他们想要使用的预付费帐户。如果使用预付费IVR,也可以组合下面描述的机制(语音门户)。

Appendix A.26. Presence-Enabled Conferencing


In Presence-Enabled Conferencing, Alice wants to set up a conference call with Bob and Cathy when they all happen to be available (rather than scheduling a predefined time). The server providing the application monitors their status, and calls all three when they are all "online", not idle, and not in another call. This could be implemented using conferencing [RFC4579] and presence [RFC3264] primitives.


Appendix A.27. Single Line Extension/Multiple Line Appearance


In Single Line Extension/Multiple Line Appearances, groups of phones are all treated as "extensions" of a single line or AOR. A call for one rings them all. As soon as one answers, the others stop ringing. If any extension is actively in a conversation, another extension can "pick up" and immediately join the conversation. This emulates the behavior of a home telephone line with multiple phones. Incoming calls ring all the extensions through basic parallel forking. Each extension subscribes to dialog events from each other extension. While one user has an active call, any other UA extension can insert


itself into that conversation (it already knows the dialog information) in the same way as barge-in.


When implemented using SIP, this feature is known as Shared Appearances of an AOR [BLISS-SHARED]. Extensions to the dialog package are used to convey appearance numbers (line numbers).


Appendix A.28. Speakerphone Paging


In Speakerphone Paging, Alice calls the paging address and speaks. Her voice is played on the speaker of every idle phone in a preconfigured group of phones. Speakerphone paging can be implemented using either multicast or through a simple multipoint mixer. In the multicast solution, the paging UA sends a multicast INVITE with send-only media in the SDP (see also [RFC3264]). The automatic answer and enabling of the speakerphone is a locally configured decision on the paged UAs. The paging UA sends RTP via the multicast address indicated in the SDP.


The multipoint solution is accomplished by sending an INVITE to the multipoint mixer. The mixer is configured to automatically answer the dialog. The paging UA then sends REFER requests for each of the UAs that are to become paging speakers (the UA is likely to send out a single REFER that is parallel forked by the proxy server). The UAs performing as paging speakers are configured to automatically answer based upon caller identification (e.g., the To field, URI, or Referred-To header fields).


Finally, as a third option, the user agent can send a mass-invitation request to a conference server, which would create a conference and send INVITEs containing the Answer-Mode: Auto header field to all user agents in the paging group.


Appendix A.29. Speed Dial


In Speed Dial, Alice dials an abbreviated number, enters an alias, or presses a special speed-dial button representing Bob. Her action is interpreted as if she specified the full address of Bob.


Appendix A.30. Voice Message Screening


In Voice Message Screening, Bob calls Alice. Alice is screening her calls, so Bob hears Alice's voicemail greeting. Alice can hear Bob leave his message. If she decides to talk to Bob, she can take the call back from the voicemail system; otherwise, she can let Bob leave a message. This emulates the behavior of a home telephone answering machine.


At first, this is the same as Call Monitoring (Appendix A.7). In this case, the voicemail service is one of the UAs. The UA screening the message monitors the call on the voicemail service, and also subscribes to dialog information. If the user screening their messages decides to answer, they perform a take from the voicemail system (for example, send an INVITE with Replaces to the UA leaving the message).


Appendix A.31. Voice Portal


Voice Portal is service that allows users to access a portal site using spoken dialog interaction. For example, Alice needs to schedule a working dinner with her co-worker Carol. Alice uses a voice portal to check Carol's flight schedule, find a restaurant near her hotel, make a reservation, get directions there, and page Carol with this information. A voice portal is essentially a complex collection of voice dialogs used to access interesting content. One of the most desirable call control features of a Voice Portal is the ability to start a new outgoing call from within the context of the Portal (to make a restaurant reservation, or return a voicemail message, for example). Once the new call is over, the user should be able to return to the Portal by pressing a special key, using some DTMF sequence (e.g., a very long pound or hash tone), or by speaking a key word (e.g., "Main Menu").


In order to accomplish this, the Voice Portal starts with the following media relationship:


{ User , Voice Portal }


The user then asks to make an outgoing call. The Voice Portal asks the user to perform a far-fork. In other words, the Voice Portal wants the following media relationship:


           { Target , User }  &  { User , Voice Portal }
           { Target , User }  &  { User , Voice Portal }

The Voice Portal is now just listening for a key word or the appropriate DTMF. As soon as the user indicates they are done, the Voice Portal takes the call from the old target, and we are back to the original media relationship.


This feature can also be used by the account number and phone number collection menu in a pre-paid calling service. A user can press a DTMF sequence that presents them with the appropriate menu again.


Appendix A.32. Voicemail


In Voicemail, Alice calls Bob who does not answer or is not available. The call forwards to a voicemail server that plays Bob's greeting and records Alice's message for Bob. An indication is sent to Bob that a new message is waiting, and he retrieves the message at a later date. This feature is implemented using features such as Call Forwarding (Appendix A.6) and the History-Info header field [RFC4244] or voicemail URI convention [RFC4458] and Message Waiting [RFC3842] features.


Appendix A.33. Whispered Call Waiting


In Whispered Call Waiting, Alice is in a conversation with Bob. Carol calls Alice. Either Carol can "whisper" to Alice directly ("Can you get lunch in 15 minutes?"), or an automaton whispers to Alice informing her that Carol is trying to reach her.


Appendix B. Acknowledgments

The authors would like to acknowledge Ben Campbell for his contributions to the document and thank AC Mahendran, John Elwell, and Xavier Marjou for their detailed Working-Group review of the document. The authors would like to thank Magnus Nystrom for his review of the document.

作者感谢Ben Campbell对该文件的贡献,并感谢AC Mahendran、John Elwell和Xavier Marjou工作组对该文件的详细审查。作者要感谢Magnus Nystrom对该文件的审查。

5. Informative References
5. 资料性引用

[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。

[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002.

[RFC3264]Rosenberg,J.和H.Schulzrinne,“具有会话描述协议(SDP)的提供/应答模型”,RFC 3264,2002年6月。

[RFC3265] Roach, A., "Session Initiation Protocol (SIP)- Specific Event Notification", RFC 3265, June 2002.


[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006.


[RFC5359] Johnston, A., Sparks, R., Cunningham, C., Donovan, S., and K. Summers, "Session Initiation Protocol Service Examples", BCP 144, RFC 5359, October 2008.

[RFC5359]Johnston,A.,Sparks,R.,Cunningham,C.,Donovan,S.,和K.Summers,“会话启动协议服务示例”,BCP 144,RFC 5359,2008年10月。

[RFC3725] Rosenberg, J., Peterson, J., Schulzrinne, H., and G. Camarillo, "Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)", BCP 85, RFC 3725, April 2004.

[RFC3725]Rosenberg,J.,Peterson,J.,Schulzrinne,H.,和G.Camarillo,“会话启动协议(SIP)中第三方呼叫控制(3pcc)的当前最佳实践”,BCP 85,RFC 37252004年4月。

[RFC3515] Sparks, R., "The Session Initiation Protocol (SIP) Refer Method", RFC 3515, April 2003.


[RFC3891] Mahy, R., Biggs, B., and R. Dean, "The Session Initiation Protocol (SIP) "Replaces" Header", RFC 3891, September 2004.

[RFC3891]Mahy,R.,Biggs,B.,和R.Dean,“会话启动协议(SIP)”取代了RFC 38912004年9月的“头”。

[RFC3911] Mahy, R. and D. Petrie, "The Session Initiation Protocol (SIP) "Join" Header", RFC 3911, October 2004.

[RFC3911]Mahy,R.和D.Petrie,“会话启动协议(SIP)”加入“头”,RFC 3911,2004年10月。

[BLISS-PROBLEM] Rosenberg, J., "Basic Level of Interoperability for Session Initiation Protocol (SIP) Services (BLISS) Problem Statement", Work in Progress, March 2009.


[RFC4235] Rosenberg, J., Schulzrinne, H., and R. Mahy, "An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol (SIP)", RFC 4235, November 2005.

[RFC4235]Rosenberg,J.,Schulzrinne,H.,和R.Mahy,“会话启动协议(SIP)的邀请启动对话事件包”,RFC 4235,2005年11月。

[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, "A Session Initiation Protocol (SIP) Event Package for Conference State", RFC 4575, August 2006.

[RFC4575]Rosenberg,J.,Schulzrinne,H.,和O.Levin,“会议状态的会话启动协议(SIP)事件包”,RFC 45752006年8月。

[RFC3680] Rosenberg, J., "A Session Initiation Protocol (SIP) Event Package for Registrations", RFC 3680, March 2004.


[RFC3856] Rosenberg, J., "A Presence Event Package for the Session Initiation Protocol (SIP)", RFC 3856, August 2004.


[RFC4353] Rosenberg, J., "A Framework for Conferencing with the Session Initiation Protocol (SIP)", RFC 4353, February 2006.

[RFC4353]Rosenberg,J.,“会话启动协议(SIP)会议框架”,RFC 4353,2006年2月。

[RFC5629] Rosenberg, J., "A Framework for Application Interaction in the Session Initiation Protocol (SIP)", RFC 5629, October 2009.

[RFC5629]Rosenberg,J.,“会话启动协议(SIP)中的应用程序交互框架”,RFC 5629,2009年10月。

[RFC5369] Camarillo, G., "Framework for Transcoding with the Session Initiation Protocol (SIP)", RFC 5369, October 2008.

[RFC5369]Camarillo,G.“会话启动协议(SIP)转码框架”,RFC 5369,2008年10月。

[XCON-CCMP] Barnes, M., Boulton, C., Romano, S., and H. Schulzrinne, "Centralized Conferencing Manipulation Protocol", Work in Progress, February 2010.


[RFC5589] Sparks, R., Johnston, A., and D. Petrie, "Session Initiation Protocol (SIP) Call Control - Transfer", BCP 149, RFC 5589, June 2009.

[RFC5589]Sparks,R.,Johnston,A.,和D.Petrie,“会话启动协议(SIP)呼叫控制-传输”,BCP 149,RFC 5589,2009年6月。

[RFC4579] Johnston, A. and O. Levin, "Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents", BCP 119, RFC 4579, August 2006.

[RFC4579]Johnston,A.和O.Levin,“会话发起协议(SIP)呼叫控制-用户代理会议”,BCP 119,RFC 4579,2006年8月。

[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating User Agent Capabilities in the Session Initiation Protocol (SIP)", RFC 3840, August 2004.

[RFC3840]Rosenberg,J.,Schulzrinne,H.,和P.Kyzivat,“指出会话启动协议(SIP)中的用户代理功能”,RFC 3840,2004年8月。

[RFC3841] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Caller Preferences for the Session Initiation Protocol (SIP)", RFC 3841, August 2004.

[RFC3841]Rosenberg,J.,Schulzrinne,H.,和P.Kyzivat,“会话启动协议(SIP)的呼叫方偏好”,RFC 38412004年8月。

[RFC3087] Campbell, B. and R. Sparks, "Control of Service Context using SIP Request-URI", RFC 3087, April 2001.

[RFC3087]Campbell,B.和R.Sparks,“使用SIP请求URI控制服务上下文”,RFC 3087,2001年4月。

[FEATURE-REF] Audet, F., Johnston, A., Mahy, R., and C. Jennings, "Feature Referral in the Session Initiation Protocol (SIP)", Work in Progress, February 2008.


[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic Network Media Services with SIP", RFC 4240, December 2005.

[RFC4240]Burger,E.,Van Dyke,J.,和A.Spitzer,“具有SIP的基本网络媒体服务”,RFC 42402005年12月。

[RFC4458] Jennings, C., Audet, F., and J. Elwell, "Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and Interactive Voice Response (IVR)", RFC 4458, April 2006.

[RFC4458]Jennings,C.,Audet,F.,和J.Elwell,“语音邮件和交互式语音应答(IVR)等应用程序的会话启动协议(SIP)URI”,RFC 4458,2006年4月。

[RFC4538] Rosenberg, J., "Request Authorization through Dialog Identification in the Session Initiation Protocol (SIP)", RFC 4538, June 2006.

[RFC4538]Rosenberg,J.,“通过会话启动协议(SIP)中的对话标识请求授权”,RFC 4538,2006年6月。

[RFC3880] Lennox, J., Wu, X., and H. Schulzrinne, "Call Processing Language (CPL): A Language for User Control of Internet Telephony Services", RFC 3880, October 2004.

[RFC3880]Lennox,J.,Wu,X.,和H.Schulzrinne,“呼叫处理语言(CPL):互联网电话服务的用户控制语言”,RFC 3880,2004年10月。

[RFC5373] Willis, D. and A. Allen, "Requesting Answering Modes for the Session Initiation Protocol (SIP)", RFC 5373, November 2008.

[RFC5373]Willis,D.和A.Allen,“请求会话启动协议(SIP)的应答模式”,RFC 53732008年11月。

[RFC3842] Mahy, R., "A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)", RFC 3842, August 2004.

[RFC3842]Mahy,R.“会话启动协议(SIP)的消息摘要和消息等待指示事件包”,RFC 3842,2004年8月。

[BLISS-SHARED] Johnston, A., Soroushnejad, M., and V. Venkataramanan, "Shared Appearances of a Session Initiation Protocol (SIP) Address of Record (AOR)", Work in Progress, October 2009.


[RFC4244] Barnes, M., "An Extension to the Session Initiation Protocol (SIP) for Request History Information", RFC 4244, November 2005.

[RFC4244]Barnes,M.,“请求历史信息会话启动协议(SIP)的扩展”,RFC 4244,2005年11月。

[RFC4313] Oran, D., "Requirements for Distributed Control of Automatic Speech Recognition (ASR), Speaker Identification/Speaker Verification (SI/SV), and Text-to-Speech (TTS) Resources", RFC 4313, December 2005.

[RFC4313]Oran,D.“自动语音识别(ASR)、说话人识别/说话人验证(SI/SV)和文本到语音(TTS)资源的分布式控制要求”,RFC 4313,2005年12月。

[RFC3323] Peterson, J., "A Privacy Mechanism for the Session Initiation Protocol (SIP)", RFC 3323, November 2002.


[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。

Authors' Addresses


Rohan Mahy Unaffiliated

Rohan Mahy非附属公司


Robert Sparks Tekelec



Jonathan Rosenberg

Jonathan Rosenberg


Dan Petrie SIPez



Alan Johnston (editor) Avaya