RFC 5411: A Hitchhiker's Guide to the Session Initiation Protocol (SIP) 中文翻译

URL : https://datatracker.ietf.org/doc/html/rfc5411
标题 : RFC 5411
翻译类型 : 自动生成

Network Working Group                                       J. Rosenberg
Request for Comments: 5411                                         Cisco
Category: Informational                                     January 2009

Network Working Group                                       J. Rosenberg
Request for Comments: 5411                                         Cisco
Category: Informational                                     January 2009

A Hitchhiker's Guide to the Session Initiation Protocol (SIP)

会话启动协议（SIP）搭便车指南

Status of This Memo

关于下段备忘

This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited.

本备忘录为互联网社区提供信息。它没有规定任何类型的互联网标准。本备忘录的分发不受限制。

Abstract

摘要

The Session Initiation Protocol (SIP) is the subject of numerous specifications that have been produced by the IETF. It can be difficult to locate the right document, or even to determine the set of Request for Comments (RFC) about SIP. This specification serves as a guide to the SIP RFC series. It lists a current snapshot of the specifications under the SIP umbrella, briefly summarizes each, and groups them into categories.

会话启动协议（SIP）是IETF制定的众多规范的主题。很难找到正确的文档，甚至很难确定关于SIP的征求意见（RFC）集。本规范作为SIP RFC系列的指南。它列出了SIP保护伞下规范的当前快照，简要总结了每个规范，并将它们分类。

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2.  Scope of This Document . . . . . . . . . . . . . . . . . . . .  4
   3.  Core SIP Specifications  . . . . . . . . . . . . . . . . . . .  5
   4.  Public Switched Telephone Network (PSTN) Interworking  . . . .  8
   5.  General Purpose Infrastructure Extensions  . . . . . . . . . . 10
   6.  NAT Traversal  . . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Call Control Primitives  . . . . . . . . . . . . . . . . . . . 13
   8.  Event Framework  . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 15
   10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 16
   11. Operations and Management  . . . . . . . . . . . . . . . . . . 17
   12. SIP Compression  . . . . . . . . . . . . . . . . . . . . . . . 17
   13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 17
   14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 19
   15. Security Mechanisms  . . . . . . . . . . . . . . . . . . . . . 20
   16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 23
   17. Instant Messaging, Presence, and Multimedia  . . . . . . . . . 24
   18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 25
   19. Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   20. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   21. Informative References . . . . . . . . . . . . . . . . . . . . 26

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  2
   2.  Scope of This Document . . . . . . . . . . . . . . . . . . . .  4
   3.  Core SIP Specifications  . . . . . . . . . . . . . . . . . . .  5
   4.  Public Switched Telephone Network (PSTN) Interworking  . . . .  8
   5.  General Purpose Infrastructure Extensions  . . . . . . . . . . 10
   6.  NAT Traversal  . . . . . . . . . . . . . . . . . . . . . . . . 12
   7.  Call Control Primitives  . . . . . . . . . . . . . . . . . . . 13
   8.  Event Framework  . . . . . . . . . . . . . . . . . . . . . . . 14
   9.  Event Packages . . . . . . . . . . . . . . . . . . . . . . . . 15
   10. Quality of Service . . . . . . . . . . . . . . . . . . . . . . 16
   11. Operations and Management  . . . . . . . . . . . . . . . . . . 17
   12. SIP Compression  . . . . . . . . . . . . . . . . . . . . . . . 17
   13. SIP Service URIs . . . . . . . . . . . . . . . . . . . . . . . 17
   14. Minor Extensions . . . . . . . . . . . . . . . . . . . . . . . 19
   15. Security Mechanisms  . . . . . . . . . . . . . . . . . . . . . 20
   16. Conferencing . . . . . . . . . . . . . . . . . . . . . . . . . 23
   17. Instant Messaging, Presence, and Multimedia  . . . . . . . . . 24
   18. Emergency Services . . . . . . . . . . . . . . . . . . . . . . 25
   19. Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   20. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   21. Informative References . . . . . . . . . . . . . . . . . . . . 26

1. Introduction

1. 介绍

The Session Initiation Protocol (SIP) [RFC3261] is the subject of numerous specifications that have been produced by the IETF. It can be difficult to locate the right document, or even to determine the set of Request for Comments (RFC) about SIP. "Don't Panic!" [HGTTG] This specification serves as a guide to the SIP RFC series. It is a current snapshot of the specifications under the SIP umbrella at the time of publication. It is anticipated that this document itself will be regularly updated as SIP specifications mature. Furthermore, it references many specifications, which, at the time of publication of this document, were not yet finalized, and may eventually be completed or abandoned. Therefore, the enumeration of specifications here is a work-in-progress and subject to change.

会话启动协议（SIP）[RFC3261]是IETF制定的众多规范的主题。很难找到正确的文档，甚至很难确定关于SIP的征求意见（RFC）集。“不要惊慌！”[HGTTG]本规范作为SIP RFC系列的指南。它是发布时SIP保护伞下规范的最新快照。预计随着SIP规范的成熟，本文件本身将定期更新。此外，它引用了许多规范，这些规范在本文件发布时尚未最终确定，可能最终完成或放弃。因此，此处的规范枚举是一项正在进行的工作，可能会发生更改。

For each specification, a paragraph or so description is included that summarizes the purpose of the specification. Each specification also includes a letter that designates its category in the Standards Track [RFC2026]. These values are:

对于每个规范，都包含一段左右的描述，概述了规范的目的。每个规范还包括一个字母，用于在标准轨道[RFC2026]中指定其类别。这些值是：

S: Standards Track (Proposed Standard, Draft Standard, or Standard)

S：标准跟踪（建议标准、标准草案或标准）

E: Experimental

E:实验性的

B: Best Current Practice

B:目前的最佳做法

I: Informational

I:信息性

The specifications are grouped together by topic. The topics are:

规范按主题分组。主题包括：

Core: The SIP specifications that are expected to be utilized for each session or registration an endpoint participates in.

核心：端点参与的每个会话或注册预期使用的SIP规范。

Public Switched Telephone Network (PSTN) Interop: Specifications related to interworking with the telephone network.

公共交换电话网（PSTN）互操作：与电话网互操作相关的规范。

General Purpose Infrastructure: General purpose extensions to SIP, SDP (Session Description Protocol), and MIME, but ones that are not expected to always be used.

通用基础设施：SIP、SDP（会话描述协议）和MIME的通用扩展，但不希望总是使用这些扩展。

NAT Traversal: Specifications to deal with firewall and NAT traversal.

NAT穿越：处理防火墙和NAT穿越的规范。

Call Control Primitives: Specifications for manipulating SIP dialogs and calls.

调用控制原语：用于操作SIP对话框和调用的规范。

Event Framework: Definitions of the core specifications for the SIP event framework, providing for pub/sub capability.

事件框架：定义SIP事件框架的核心规范，提供发布/订阅功能。

Event Packages: Packages that utilize the SIP event framework.

事件包：利用SIP事件框架的包。

Quality of Service: Specifications related to multimedia quality of service (QoS).

服务质量：与多媒体服务质量（QoS）相关的规范。

Operations and Management: Specifications related to configuration and monitoring of SIP deployments.

操作和管理：与SIP部署的配置和监视相关的规范。

SIP Compression: Specifications to facilitate usage of SIP with the Signaling Compression (Sigcomp) framework.

SIP压缩：促进SIP与信令压缩（Sigcomp）框架一起使用的规范。

SIP Service URIs: Specifications on how to use SIP URIs to address multimedia services.

SIP服务URI：关于如何使用SIP URI处理多媒体服务的规范。

Minor Extensions: Specifications that solve a narrow problem space or provide an optimization.

次要扩展：解决狭窄问题空间或提供优化的规范。

Security Mechanisms: Specifications providing security functionality for SIP.

安全机制：为SIP提供安全功能的规范。

Conferencing: Specifications for multimedia conferencing.

会议：多媒体会议的规范。

Instant Messaging, Presence, and Multimedia: SIP extensions related to IM, presence, and multimedia. This covers only the SIP extensions related to these topics. See [SIMPLE] for a full treatment of SIP for IM and Presence (SIMPLE).

即时消息、状态和多媒体：与IM、状态和多媒体相关的SIP扩展。这仅涉及与这些主题相关的SIP扩展。关于SIP的IM和存在（简单）的完整处理，请参见[简单]。

Emergency Services: SIP extensions related to emergency services. See [ECRIT-FRAME] for a more complete treatment of additional functionality related to emergency services.

应急服务：与应急服务相关的SIP扩展。有关紧急服务相关附加功能的更完整处理，请参见[ECRIT-FRAME]。

Typically, SIP extensions fit naturally into topic areas, and implementors interested in a particular topic often implement many or all of the specifications in that area. There are some specifications that fall into multiple topic areas, in which case they are listed more than once.

通常，SIP扩展自然适合主题领域，对特定主题感兴趣的实现者通常实现该领域中的许多或所有规范。有些规范属于多个主题领域，在这种情况下，它们会被多次列出。

Do not print all the specs cited here at once, as they might share the fate of the rules of Brockian Ultracricket when bound together: collapse under their own gravity and form a black hole [HGTTG].

不要一次打印这里引用的所有规格，因为它们在捆绑在一起时可能与布罗基亚超级蟋蟀的规则命运相同：在自身重力作用下崩溃并形成黑洞[HGTTG]。

This document itself is not an update to RFC 3261 or an extension to SIP. It is an informational document, meant to guide newcomers, implementors, and deployers to the many specifications associated with SIP.

本文档本身不是RFC 3261的更新或SIP的扩展。它是一个信息性文档，旨在指导新手、实现者和部署者了解与SIP相关的许多规范。

2. Scope of This Document

2. 本文件的范围

It is very difficult to enumerate the set of SIP specifications. This is because there are many protocols that are intimately related to SIP and used by nearly all SIP implementations, but are not formally SIP extensions. As such, this document formally defines a "SIP specification" as:

很难列举SIP规范集。这是因为有许多协议与SIP密切相关，几乎所有SIP实现都使用这些协议，但它们不是正式的SIP扩展。因此，本文件正式将“SIP规范”定义为：

o RFC 3261 and any specification that defines an extension to it, where an extension is a mechanism that changes or updates in some way a behavior specified there.

o RFC 3261和任何定义其扩展的规范，其中扩展是一种机制，以某种方式更改或更新其中指定的行为。

o The basic SDP specification [RFC4566] and any specification that defines an extension to SDP whose primary purpose is to support SIP.

o 基本SDP规范[RFC4566]和任何定义SDP扩展的规范，其主要目的是支持SIP。

o Any specification that defines a MIME object whose primary purpose is to support SIP.

o 任何定义MIME对象的规范，其主要目的是支持SIP。

Excluded from this list are requirements, architectures, registry definitions, non-normative frameworks, and processes. Best Current Practices are included when they normatively define mechanisms for accomplishing a task, or provide significant description of the usage of the normative specifications, such as call flows.

此列表中不包括需求、体系结构、注册表定义、非规范性框架和流程。当前最佳实践包括规范性地定义完成任务的机制，或提供规范性规范（如调用流）用法的重要描述。

The SIP change process [RFC3427] defines two types of extensions to SIP: normal extensions and the so-called P-headers (where P stands for "preliminary", "private", or "proprietary", and the "P-" prefix is included in the header field name), which are meant to be used in areas of limited applicability. P-headers cannot be defined in the Standards Track. For the most part, P-headers are not included in the listing here, with the exception of those that have seen general usage despite their P-header status.

SIP变更过程[RFC3427]定义了SIP的两种扩展类型：普通扩展和所谓的P头（其中P代表“初步”、“专用”或“专有”，头字段名称中包含“P-”前缀），用于有限适用性的区域。无法在标准跟踪中定义P标题。在大多数情况下，P-header不包括在这里的列表中，除了那些尽管处于P-header状态但仍被广泛使用的P-header。

This document includes specifications, which have already been approved by the IETF and granted an RFC number, in addition to Internet Drafts, which are still under development within the IETF and will eventually finish and get an RFC number. Inclusion of Internet Drafts here helps encourage early implementation and demonstrations of interoperability of the protocol, and thus aids in the standards-setting process. Inclusion of these also identifes where the IETF is targetting a solution at a particular problem space. Note that final IANA assignment of codepoints (such as option tags and header field names) does not take place until shortly before publication as an RFC, and thus codepoint assignments may change.

本文件包括已获IETF批准并授予RFC号的规范，以及仍在IETF内开发的、最终将完成并获得RFC号的互联网草案。此处包含互联网草案有助于鼓励尽早实施和演示协议的互操作性，从而有助于标准制定过程。如果IETF针对某一特定问题空间的解决方案，则还包括这些标识。请注意，代码点的最终IANA分配（如选项标记和标题字段名）直到作为RFC发布前不久才会发生，因此代码点分配可能会发生变化。

3. Core SIP Specifications

3. 核心SIP规范

The core SIP specifications represent the set of specifications whose functionality is broadly applicable. An extension is broadly applicable if it fits into one of the following categories:

核心SIP规范代表一组功能广泛适用的规范。如果扩展符合以下类别之一，则扩展适用范围很广：

o For specifications that impact SIP session management, the extension would be used for almost every session initiated by a user agent.

o 对于影响SIP会话管理的规范，扩展将用于几乎由用户代理启动的每个会话。

o For specifications that impact SIP registrations, the extension would be used for almost every registration initiated by a user agent.

o 对于影响SIP注册的规范，扩展将用于用户代理发起的几乎所有注册。

o For specifications that impact SIP subscriptions, the extension would be used for almost every subscription initiated by a user agent.

o 对于影响SIP订阅的规范，扩展将用于几乎所有由用户代理发起的订阅。

In other words, these are not specifications that are used just for some requests and not others; they are specifications that would apply to each and every request for which the extension is relevant. In the galaxy of SIP, these specifications are like towels [HGTTG].

换句话说，这些规范不是仅用于某些请求而不是其他请求的规范；它们是适用于与扩展相关的每个请求的规范。在SIP的银河系中，这些规范就像毛巾[HGTTG]。

RFC 3261, The Session Initiation Protocol (S): [RFC3261] is the core SIP protocol itself. RFC 3261 obsoletes [RFC2543]. It is the president of the galaxy [HGTTG] as far as the suite of SIP specifications is concerned.

RFC 3261，会话启动协议：[RFC3261]是核心SIP协议本身。RFC 3261淘汰产品[RFC2543]。就SIP规范套件而言，它是galaxy[HGTTG]的总裁。

RFC 3263, Locating SIP Servers (S): [RFC3263] provides DNS procedures for taking a SIP URI and determining a SIP server that is associated with that SIP URI. RFC 3263 is essential for any implementation using SIP with DNS. RFC 3263 makes use of both DNS SRV records [RFC2782] and NAPTR records [RFC3401].

RFC 3263，查找SIP服务器：[RFC3263]提供了获取SIP URI并确定与该SIP URI关联的SIP服务器的DNS过程。RFC 3263对于使用SIP和DNS的任何实现都是必不可少的。RFC 3263同时使用DNS SRV记录[RFC2782]和NAPTR记录[RFC3401]。

RFC 3264, An Offer/Answer Model with the Session Description Protocol (S): [RFC3264] defines how the Session Description Protocol (SDP) [RFC4566] is used with SIP to negotiate the parameters of a media session. It is in widespread usage and an integral part of the behavior of RFC 3261.

RFC 3264，一种具有会话描述协议的提供/应答模型：[RFC3264]定义了会话描述协议（SDP）[RFC4566]如何与SIP一起用于协商媒体会话的参数。它被广泛使用，是RFC 3261行为的一个组成部分。

RFC 3265, SIP-Specific Event Notification (S): [RFC3265] defines the SUBSCRIBE and NOTIFY methods. These two methods provide a general event notification framework for SIP. To actually use the framework, extensions need to be defined for specific event packages. An event package defines a schema for the event data and describes other aspects of event processing specific to that schema. An RFC 3265 implementation is required when any event package is used.

RFC 3265，SIP特定事件通知：[RFC3265]定义订阅和通知方法。这两种方法为SIP提供了通用的事件通知框架。要实际使用该框架，需要为特定的事件包定义扩展。事件包定义事件数据的模式，并描述特定于该模式的事件处理的其他方面。使用任何事件包时都需要RFC 3265实现。

RFC 3325, Private Extensions to SIP for Asserted Identity within Trusted Networks (I): Though its P-header status implies that it has limited applicability, [RFC3325], which defines the P-Asserted-Identity header field, has been widely deployed. It is used as the basic mechanism for providing network-asserted caller ID services. Its intended update, [UPDATE-PAI], clarifies its usage for connected party identification as well.

RFC 3325，《可信网络内断言身份的SIP专用扩展》（I）：尽管其P-header状态表明其适用性有限，但定义P-Asserted-Identity header字段的[RFC3325]已被广泛部署。它被用作提供网络主叫ID服务的基本机制。其预期更新[update-PAI]也阐明了其用于关联方识别的用途。

RFC 3327, SIP Extension Header Field for Registering Non-Adjacent Contacts (S): [RFC3327] defines the Path header field. This field is inserted by proxies between a client and their registrar. It allows inbound requests towards that client to traverse these proxies prior to being delivered to the user agent. It is essential in any SIP deployment that has edge proxies, which are proxies between the client and the home proxy or SIP registrar.

RFC 3327，用于注册非相邻联系人的SIP扩展标头字段：[RFC3327]定义路径标头字段。此字段由代理插入到客户机及其注册器之间。它允许对该客户端的入站请求在传递到用户代理之前遍历这些代理。它在任何具有边缘代理的SIP部署中都是必不可少的，边缘代理是客户端和主代理或SIP注册器之间的代理。

RFC 3581, An Extension to SIP for Symmetric Response Routing (S): [RFC3581] defines the rport parameter of the Via header. It allows SIP responses to traverse NAT. It is one of several specifications that are utilized for NAT traversal (see Section 6).

RFC 3581，对称响应路由的SIP扩展：[RFC3581]定义了Via头的rport参数。它允许SIP响应遍历NAT。它是用于NAT遍历的几个规范之一（参见第6节）。

RFC 3840, Indicating User Agent Capabilities in SIP (S): [RFC3840] defines a mechanism for carrying capability information about a user agent in REGISTER requests and in dialog-forming requests like INVITE. It has found use with conferencing (the isfocus parameter declares that a user agent is a conference server) and with applications like push-to-talk.

RFC 3840，指示SIP中的用户代理功能：[RFC3840]定义了一种机制，用于在注册请求和INVITE等对话框形成请求中携带有关用户代理的功能信息。它已用于会议（isfocus参数声明用户代理是会议服务器）和按键通话等应用程序。

RFC 4320, Actions Addressing Issues Identified with the Non-INVITE Transaction in SIP (S): [RFC4320] formally updates RFC 3261 and modifies some of the behaviors associated with non-INVITE transactions. This addresses some problems found in timeout and failure cases.

RFC 4320，解决SIP中非邀请事务识别问题的操作：[RFC4320]正式更新RFC 3261并修改与非邀请事务相关的一些行为。这解决了超时和故障情况下发现的一些问题。

RFC 4474, Enhancements for Authenticated Identity Management in SIP (S): [RFC4474] defines a mechanism for providing a cryptographically verifiable identity of the calling party in a SIP request. Known as "SIP Identity", this mechanism provides an alternative to RFC 3325. It has seen little deployment so far, but its importance as a key construct for anti-spam techniques and new security mechanisms makes it a core part of the SIP specifications.

RFC 4474，SIP中身份验证管理的增强功能：[RFC4474]定义了一种机制，用于在SIP请求中提供主叫方的可加密验证身份。这种机制称为“SIP标识”，它提供了RFC3325的替代方案。到目前为止，它的部署很少，但作为反垃圾邮件技术和新安全机制的关键构造，它的重要性使其成为SIP规范的核心部分。

GRUU, Obtaining and Using Globally Routable User Agent Identifiers (GRUU) in SIP (S): [GRUU] defines a mechanism for directing requests towards a specific UA instance. GRUU is essential for features like transfer and provides another piece of the SIP NAT traversal story.

GRUU，获取并使用SIP中的全局可路由用户代理标识符（GRUU）：[GRUU]定义了一种将请求定向到特定UA实例的机制。GRUU对于传输等功能是必不可少的，它提供了SIP NAT遍历故事的另一部分。

OUTBOUND, Managing Client Initiated Connections through SIP (S): [OUTBOUND], also known as SIP outbound, defines important changes to the SIP registration mechanism that enable delivery of SIP messages towards a UA when it is behind a NAT. This specification is the cornerstone of the SIP NAT traversal strategy.

出站，通过SIP管理客户端启动的连接：[OUTBOUND]，也称为SIP OUTBOUND，定义了对SIP注册机制的重要更改，当UA位于NAT之后时，可以将SIP消息传递给UA。该规范是SIP NAT穿越策略的基石。

RFC 4566, Session Description Protocol (S): [RFC4566] defines a format for representing multimedia sessions. SDP objects are carried in the body of SIP messages and, based on the offer/answer model, are used to negotiate the media characteristics of a session between users.

RFC 4566，会话描述协议：[RFC4566]定义了表示多媒体会话的格式。SDP对象包含在SIP消息体中，并基于提供/应答模型，用于协商用户之间会话的媒体特征。

SDP-CAP, SDP Capability Negotiation (S): [SDP-CAP] defines a set of extensions to SDP that allows for capability negotiation within SDP. Capability negotiation can be used to select between different profiles of RTP (secure vs. unsecure) or to negotiate codecs such that an agent has to select one amongst a set of supported codecs.

SDP-CAP，SDP能力协商：[SDP-CAP]定义了SDP的一组扩展，允许在SDP内进行能力协商。能力协商可用于在RTP的不同配置文件之间进行选择（安全与不安全），或协商编解码器，以便代理必须在一组受支持的编解码器中选择一个。

ICE, Interactive Connectivity Establishment (ICE) (S): [ICE] defines a technique for NAT traversal of media sessions for protocols that make use of the offer/answer model. This specification is the IETF-recommended mechanism for NAT traversal for SIP media streams, and is meant to be used even by endpoints that are themselves never behind a NAT. A SIP option tag and media feature tag [OPTION-TAG] (also a core specification) have been defined for use with ICE.

ICE，Interactive Connectivity Establish（ICE）：[ICE]为使用提供/应答模型的协议定义了一种NAT穿越媒体会话的技术。本规范是IETF推荐的SIP媒体流NAT穿越机制，即使是本身不在NAT后面的端点也可以使用。SIP选项标签和媒体功能标签[option-tag]（也是核心规范）已定义用于ICE。

RFC 3605, Real Time Control Protocol (RTCP) Attribute in the Session Description Protocol (SDP) (S): [RFC3605] defines a way to explicitly signal, within an SDP message, the IP address and port for RTCP, rather than using the port+1 rule in the Real Time Transport Protocol (RTP) [RFC3550]. It is needed for devices behind NAT, and the specification is required by ICE.

RFC 3605，会话描述协议（SDP）（S）中的实时控制协议（RTCP）属性：[RFC3605]定义了在SDP消息中显式发送RTCP的IP地址和端口信号的方法，而不是使用实时传输协议（RTP）[RFC3550]中的端口+1规则。NAT后面的设备需要，ICE需要该规范。

RFC 4916, Connected Identity in the Session Initiation Protocol (SIP) (S): [RFC4916] formally updates RFC 3261. It defines an extension to SIP that allows a calling user to determine the identity of the final called user (connected party). Due to forwarding and retargeting services, this may not be the same as the user that the caller was originally trying to reach. The mechanism works in tandem with the SIP identity specification [RFC4474] to provide signatures over the connected party identity. It can also be used if a party identity changes mid-call due to third-party call control actions or PSTN behavior.

RFC 4916，会话启动协议（SIP）中的连接标识：[RFC4916]正式更新RFC 3261。它定义了SIP的扩展，允许主叫用户确定最终被叫用户（连接方）的身份。由于转发和重定目标服务，这可能与调用者最初试图联系的用户不同。该机制与SIP身份规范[RFC4474]协同工作，通过连接方身份提供签名。如果由于第三方呼叫控制操作或PSTN行为，在通话中当事方身份发生变化时，也可以使用此选项。

RFC 3311, The SIP UPDATE Method (S): [RFC3311] defines the UPDATE method for SIP. This method is meant as a means for updating session information prior to the completion of the initial INVITE transaction. It can also be used to update other information, such as the identity of the participant [RFC4916], without involving an updated offer/answer exchange. It was developed initially to support [RFC3312], but has found other uses. In particular, its usage with RFC 4916 means it will typically be used as part of every session, to convey a secure, connected identity.

RFC 3311，SIP更新方法：[RFC3311]定义SIP的更新方法。此方法用于在初始INVITE事务完成之前更新会话信息。它还可以用于更新其他信息，例如参与者的身份[RFC4916]，而不涉及更新的报价/应答交换。它最初是为了支持[RFC3312]而开发的，但也有其他用途。特别是，它与RFC 4916一起使用意味着它通常将被用作每个会话的一部分，以传递安全、连接的身份。

SIPS-URI, The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP) (S): [SIPS-URI] is intended to update RFC 3261. It revises the processing of the SIPS URI, originally defined in RFC 3261, to fix many errors and problems that have been encountered with that mechanism.

SIPS-URI，会话启动协议（SIP）（S）中SIPS URI方案的使用：[SIPS-URI]旨在更新RFC 3261。它修改了最初在RFC3261中定义的SIPS URI的处理，以修复该机制遇到的许多错误和问题。

RFC 3665, Session Initiation Protocol (SIP) Basic Call Flow Examples (B): [RFC3665] contains best-practice call flow examples for basic SIP interactions -- call establishment, termination, and registration.

RFC 3665，会话启动协议（SIP）基本呼叫流示例（B）：[RFC3665]包含基本SIP交互的最佳实践呼叫流示例——呼叫建立、终止和注册。

Essential Corrections to SIP: A collection of fixes to SIP that address important bugs and vulnerabilities. These include a fix requiring loop detection in any proxy that forks [LOOP-FIX], a clarification on how record-routing works [RECORD-ROUTE], and a correction to the IPv6 BNF [ABNF-FIX].

SIP的基本更正：SIP的修复集合，解决了重要的bug和漏洞。其中包括需要在任何分叉的代理中进行循环检测的修复[loop-fix]，关于记录路由如何工作的澄清[record-ROUTE]，以及对IPv6 BNF[ABNF-fix]的更正。

4. Public Switched Telephone Network (PSTN) Interworking

4. 公共交换电话网（PSTN）互通

Numerous extensions and usages of SIP are related to interoperability and communications with or through the PSTN.

SIP的许多扩展和使用与PSTN或通过PSTN的互操作性和通信有关。

RFC 2848, The PINT Service Protocol (S): [RFC2848] is one of the earliest extensions to SIP. It defines procedures for using SIP to invoke services that actually execute on the PSTN. Its main application is for third-party call control, allowing an IP host to set up a call between two PSTN endpoints. PINT (PSTN/Internet Interworking) has a relatively narrow focus and has not seen widespread deployment.

RFC 2848，PINT服务协议：[RFC2848]是SIP最早的扩展之一。它定义了使用SIP调用在PSTN上实际执行的服务的过程。其主要应用程序用于第三方呼叫控制，允许IP主机在两个PSTN端点之间建立呼叫。PINT（PSTN/互联网互通）的重点相对较窄，尚未得到广泛部署。

RFC 3910, The SPIRITS Protocol (S): Continuing the trend of naming PSTN-related extensions with alcohol references, SPIRITS (Services in PSTN Requesting Internet Services) [RFC3910] defines the inverse of PINT. It allows a switch in the PSTN to ask an IP element how to proceed with call waiting. It was developed primarily to support Internet Call Waiting (ICW). Perhaps the next specification will be called the Pan Galactic Gargle Blaster

RFC 3910，SPIRITS协议：继续使用酒精引用命名PSTN相关扩展的趋势，SPIRITS（PSTN中请求Internet服务的服务）[RFC3910]定义了PINT的相反形式。它允许PSTN中的交换机询问IP元素如何继续呼叫等待。它的开发主要是为了支持Internet呼叫等待（ICW）。也许下一个规格将被称为泛银河漱口爆破机

[HGTTG].

[HGTTG]。

RFC 3372, SIP for Telephones (SIP-T): Context and Architectures (I): SIP-T [RFC3372] defines a mechanism for using SIP between pairs of PSTN gateways. Its essential idea is to tunnel ISDN User Part (ISUP) signaling between the gateways in the body of SIP messages. SIP-T motivated the development of INFO [RFC2976]. SIP-T has seen widespread implementation for the limited deployment model that it addresses. As ISUP endpoints disappear from the network, the need for this mechanism will decrease.

RFC 3372，电话SIP（SIP-T）：上下文和体系结构（I）：SIP-T[RFC3372]定义了在PSTN网关对之间使用SIP的机制。其基本思想是在SIP消息体中的网关之间传输ISDN用户部分（ISUP）信令。SIP-T推动了INFO的发展[RFC2976]。SIP-T已经看到了它所解决的有限部署模型的广泛实现。随着ISUP端点从网络中消失，对该机制的需求将减少。

RFC 3398, ISUP to SIP Mapping (S): [RFC3398] defines how to do protocol mapping from the SS7 ISDN User Part (ISUP) signaling to SIP. It is widely used in SS7 to SIP gateways and is part of the SIP-T framework.

RFC 3398，ISUP到SIP映射：[RFC3398]定义如何从SS7 ISDN用户部分（ISUP）信令到SIP进行协议映射。它广泛用于SS7到SIP网关，是SIP-T框架的一部分。

RFC 4497, Interworking between the Session Initiation Protocol (SIP) and QSIG (B): [RFC4497] defines how to do protocol mapping from Q.SIG, used for Private Branch Exchange (PBX) signaling, to SIP.

RFC 4497，会话启动协议（SIP）和QSIG（B）之间的互通：[RFC4497]定义了如何将用于专用分支交换（PBX）信令的Q.SIG映射到SIP。

RFC 3578, Mapping of ISUP Overlap Signaling to SIP (S): [RFC3578] defines a mechanism to map overlap dialing into SIP. This specification is widely regarded as the ugliest SIP specification, as the introduction to the specification itself advises that it has many problems. Overlap signaling (the practice of sending digits into the network as dialed instead of waiting for complete collection of the called party number) is largely incompatible with SIP at some fairly fundamental levels. That said, RFC 3578 is mostly harmless and has seen some usage.

RFC 3578，ISUP重叠信令到SIP的映射：[RFC3578]定义了将重叠拨号映射到SIP的机制。该规范被广泛认为是最丑陋的SIP规范，因为对该规范本身的介绍表明它存在许多问题。重叠信令（以拨号方式向网络发送数字的做法，而不是等待被叫方号码的完全收集）在某些相当基本的层面上与SIP基本不兼容。也就是说，RFC3578基本上是无害的，并且已经有了一些用途。

RFC 3960, Early Media and Ringtone Generation in SIP (I): [RFC3960] defines some guidelines for handling early media -- the practice of sending media from the called party or an application server towards the caller prior to acceptance of the call. Early media is often generated from the PSTN. Early media is a complex topic, and this specification does not fully address the problems associated with it.

RFC 3960，SIP（I）中的早期媒体和铃声生成：[RFC3960]定义了处理早期媒体的一些准则——在接受呼叫之前从被叫方或应用程序服务器向呼叫者发送媒体的实践。早期的媒体通常是从PSTN生成的。早期媒体是一个复杂的主题，本规范并未完全解决与之相关的问题。

RFC 3959, Early Session Disposition Type for the Session Initiation Protocol (SIP) (S): [RFC3959] defines a new session disposition type for use with early media. It indicates that the SDP in the body is for a special early media session. This has seen little usage.

RFC 3959，会话启动协议（SIP）（S）的早期会话处置类型：[RFC3959]定义了用于早期媒体的新会话处置类型。它表示正文中的SDP用于特殊的早期媒体会话。这几乎没有什么用处。

RFC 3204, MIME Media Types for ISUP and QSIG Objects (S): [RFC3204] defines MIME objects for representing SS7 and QSIG signaling messages. SS7 signaling messages are carried in the body of SIP messages when SIP-T is used. QSIG signaling messages can be carried in a similar way.

RFC 3204，ISUP和QSIG对象的MIME媒体类型：[RFC3204]定义用于表示SS7和QSIG信令消息的MIME对象。当使用SIP-T时，SS7信令消息携带在SIP消息体中。QSIG信令消息可以以类似的方式传输。

RFC3666, Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows (B): [RFC3666] provides best practice call flows around interworking with the PSTN.

RFC3666，会话发起协议（SIP）公共交换电话网（PSTN）呼叫流（B）：[RFC3666]提供与PSTN互通的最佳实践呼叫流。

5. General Purpose Infrastructure Extensions

5. 通用基础设施扩展

These extensions are general purpose enhancements to SIP, SDP, and MIME that can serve a wide variety of uses. However, they are not used for every session or registration, as the core specifications are.

这些扩展是对SIP、SDP和MIME的通用增强，可用于多种用途。但是，它们并不像核心规范那样用于每个会话或注册。

RFC 3262, Reliability of Provisional Responses in SIP (S): SIP defines two types of responses to a request: final and provisional. Provisional responses are numbered from 100 to 199. In SIP, these responses are not sent reliably. This choice was made in RFC 2543 since the messages were meant to just be truly informational and rendered to the user. However, subsequent work on PSTN interworking demonstrated a need to map provisional responses to PSTN messages that needed to be sent reliably. [RFC3262] was developed to allow reliability of provisional responses. The specification defines the PRACK method, used for indicating that a provisional response was received. Though it provides a generic capability for SIP, RFC 3262 implementations have been most common in PSTN interworking devices. However, PRACK brings a great deal of complication for relatively small benefit. As such, it has seen only moderate levels of deployment.

RFC 3262，《SIP中临时响应的可靠性》：SIP定义了两种类型的请求响应：最终响应和临时响应。临时回复的编号从100到199。在SIP中，这些响应不能可靠地发送。这一选择是在RFC2543中做出的，因为消息只是为了真正提供信息并呈现给用户。然而，PSTN互通的后续工作表明，需要将临时响应映射到需要可靠发送的PSTN消息。[RFC3262]旨在确保临时响应的可靠性。规范定义了PRACK方法，用于指示收到临时响应。尽管RFC3262为SIP提供了通用功能，但它在PSTN互通设备中最为常见。然而，PRACK带来了大量的复杂性，但收益相对较小。因此，它的部署水平只有中等水平。

RFC 3323, A Privacy Mechanism for the Session Initiation Protocol (SIP) (S): [RFC3323] defines the Privacy header field, used by clients to request anonymity for their requests. Though it defines several privacy services, the only one broadly used is the one that supports privacy of the P-Asserted-Identity header field [RFC3325].

RFC 3323，会话启动协议（SIP）（S）的隐私机制：[RFC3323]定义隐私头字段，客户端使用该字段为其请求请求匿名性。尽管它定义了几个隐私服务，但唯一广泛使用的是支持P-Asserted-Identity头字段[RFC3325]隐私的服务。

UA-PRIVACY, UA-Driven Privacy Mechanism for SIP (S): [UA-PRIVACY] defines a mechanism for achieving anonymous calls in SIP. It is an alternative to [RFC3323], and instead places more intelligence in the endpoint to craft anonymous messages by directly accessing network services.

UA-PRIVACY，UA驱动的SIP隐私机制：[UA-PRIVACY]定义了在SIP中实现匿名呼叫的机制。它是[RFC3323]的替代方案，而是在端点中放置更多的智能，通过直接访问网络服务来创建匿名消息。

RFC 2976, The INFO Method (S): [RFC2976] was defined as an extension to RFC 2543. It defines a method, INFO, used to transport mid-dialog information that has no impact on SIP itself. Its driving application was the transport of PSTN-related information when using SIP between a pair of gateways. Though originally conceived for broader use, it only found standardized usage with SIP-T [RFC3372]. It has been used to support numerous proprietary and non-interoperable extensions due to its poorly defined scope.

在RFC 2976中，INFO方法[RFC2976]被定义为RFC 2543的扩展。它定义了一个方法INFO，用于传输对SIP本身没有影响的mid对话信息。其驱动应用是在一对网关之间使用SIP传输PSTN相关信息。虽然最初的设想是为了更广泛的使用，但它只在SIP-T[RFC3372]中找到了标准化的用法。由于其范围定义不当，它已被用于支持许多专有的和不可互操作的扩展。

RFC 3326, The Reason Header Field for SIP (S): [RFC3326] defines the Reason header field. It is used in requests, such as BYE, to indicate the reason that the request is being sent.

RFC 3326，SIP的原因标头字段：[RFC3326]定义原因标头字段。它用于请求（如BYE）中，以指示发送请求的原因。

RFC 3388, Grouping of Media Lines in the Session Description Protocol (S): RFC 3388 [RFC3388] defines a framework for grouping together media streams in an SDP message. Such a grouping allows relationships between these streams, such as which stream is the audio for a particular video feed, to be expressed.

RFC 3388，会话描述协议中的媒体线分组：RFC 3388[RFC3388]定义了一个用于将SDP消息中的媒体流分组在一起的框架。这样的分组允许表达这些流之间的关系，例如哪个流是特定视频馈送的音频。

RFC 3420, Internet Media Type message/sipfrag (S): [RFC3420] defines a MIME object that contains a SIP message fragment. Only certain header fields and parts of the SIP message are present. For example, it is used to report back on the responses received to a request sent as a consequence of a REFER.

RFC 3420，Internet媒体类型消息/sipfrag:[RFC3420]定义包含SIP消息片段的MIME对象。只有某些头字段和SIP消息的一部分存在。例如，它用于报告由于引用而发送的请求收到的响应。

RFC 3608, SIP Extension Header Field for Service Route Discovery During Registration (S): [RFC3608] allows a client to determine, from a REGISTER response, a path of proxies to use in requests it sends outside of a dialog. It can also be used by proxies to verify the Route header in client-initiated requests. In many respects, it is the inverse of the Path header field, but has seen less usage since default outbound proxies have been sufficient in many deployments.

RFC 3608，注册期间服务路由发现的SIP扩展头字段：[RFC3608]允许客户端通过寄存器响应确定在其在对话框外部发送的请求中使用的代理路径。代理也可以使用它来验证客户端启动请求中的路由头。在许多方面，它与Path header字段相反，但使用较少，因为默认出站代理在许多部署中已经足够了。

RFC 3841, Caller Preferences for SIP (S): [RFC3841] defines a set of headers that a client can include in a request to control the way in which the request is routed downstream. It allows a client to direct a request towards a UA with specific capabilities, which a UA indicates using [RFC3840].

RFC 3841，SIP的呼叫方首选项：[RFC3841]定义了一组头，客户机可以在请求中包含这些头，以控制请求向下游路由的方式。它允许客户端将请求指向具有特定功能的UA，UA使用[RFC3840]表示这一点。

RFC 4028, Session Timers in SIP (S): [RFC4028] defines a keepalive mechanism for SIP signaling. It is primarily meant to provide a way to clean up old state in proxies that are holding call state for calls from failed endpoints that were never terminated normally. Despite its name, the session timer is not a mechanism for detecting a network failure mid-call. Session timers introduce a fair bit of complexity for relatively little gain, and have seen moderate deployment.

RFC4028，SIP中的会话计时器：[RFC4028]定义了SIP信令的保持机制。它的主要目的是提供一种方法来清除代理中的旧状态，这些代理为从未正常终止的失败端点的调用保留调用状态。尽管名称不同，会话计时器并不是在通话中检测网络故障的机制。会话计时器引入了相当多的复杂性，但获得的收益相对较少，并且已经实现了适度的部署。

RFC 4168, SCTP as a Transport for SIP (S): [RFC4168] defines how to carry SIP messages over the Stream Control Transmission Protocol (SCTP) [RFC4960]. SCTP has seen very limited usage for SIP transport.

RFC 4168，作为SIP传输的SCTP:[RFC4168]定义了如何通过流控制传输协议（SCTP）[RFC4960]传输SIP消息。SCTP对SIP传输的使用非常有限。

RFC 4244, An Extension to SIP for Request History Information (S): [RFC4244] defines the History-Info header field, which indicates information on how and why a call came to be routed to a particular destination.

RFC 4244是SIP对请求历史信息的扩展：[RFC4244]定义了历史信息报头字段，该字段指示关于如何以及为什么将呼叫路由到特定目的地的信息。

RFC 4145, TCP-Based Media Transport in the Session Description Protocol (SDP) (S): [RFC4145] defines an extension to SDP for setting up TCP-based sessions between user agents. It defines who sets up the connection and how its lifecycle is managed. It has seen relatively little usage due to the small number of media types to date that use TCP.

RFC 4145，会话描述协议（SDP）中基于TCP的媒体传输：[RFC4145]定义了SDP的扩展，用于在用户代理之间建立基于TCP的会话。它定义了谁建立连接以及如何管理连接的生命周期。由于到目前为止使用TCP的媒体类型很少，所以它的使用率相对较低。

RFC 4091, The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework (S): [RFC4091] defines a mechanism for including both IPv4 and IPv6 addresses for a media session as alternates. This mechanism has been deprecated in favor of ICE [ICE].

RFC 4091，会话描述协议（SDP）分组框架的替代网络地址类型（ANAT）语义：[RFC4091]定义了一种机制，用于将媒体会话的IPv4和IPv6地址都包含为替代地址。这种机制已被弃用，取而代之的是ICE。

SDP-MEDIA, SDP Media Capabilities Negotiation (S): [SDP-MEDIA] defines an extension to the SDP capability negotiation framework [SDP-CAP] for negotiating codecs, codec parameters, and media streams.

SDP-MEDIA、SDP媒体能力协商：[SDP-MEDIA]定义了SDP能力协商框架[SDP-CAP]的扩展，用于协商编解码器、编解码器参数和媒体流。

BODY-HANDLING, Message Body Handling in the Session Initiation Protocol (SIP): [BODY-HANDLING] clarifies handling of bodies in SIP, focusing primarily on multi-part behavior, which was under-specified in SIP.

会话启动协议（SIP）中的正文处理、消息正文处理：[正文处理]阐明了SIP中正文的处理，主要关注SIP中未指定的多部分行为。

6. NAT Traversal

6. 内网互联

These SIP extensions are primarily aimed at addressing NAT traversal for SIP.

这些SIP扩展主要旨在解决SIP的NAT遍历问题。

ICE-TCP, TCP Candidates with Interactive Connectivity Establishment (ICE) (S): [ICE-TCP] specifies the usage of ICE for TCP streams. This allows for selection of RTP-based voice on top of TCP only when NAT or firewalls would prevent UDP-based voice from working.

ICE-TCP，具有交互式连接建立（ICE）的TCP候选者：[ICE-TCP]指定ICE对TCP流的使用。只有当NAT或防火墙阻止基于UDP的语音工作时，才允许在TCP之上选择基于RTP的语音。

出站，通过SIP管理客户端启动的连接：[OUTBOUND]，也称为SIP OUTBOUND，定义了对SIP注册机制的重要更改，当UA位于NAT之后时，可以将SIP消息传递给UA。

RFC 3581, An Extension to SIP for Symmetric Response Routing (S): [RFC3581] defines the rport parameter of the Via header. It allows SIP responses to traverse NAT.

RFC 3581，对称响应路由的SIP扩展：[RFC3581]定义了Via头的rport参数。它允许SIP响应遍历NAT。

7. Call Control Primitives

7. 调用控制原语

Numerous SIP extensions provide a toolkit of dialog- and call-management techniques. These techniques have been combined together to build many SIP-based services.

许多SIP扩展提供了对话和呼叫管理技术的工具包。这些技术结合在一起构建了许多基于SIP的服务。

RFC 3515, The REFER Method (S): REFER [RFC3515] defines a mechanism for asking a user agent to send a SIP request. It's a form of SIP remote control, and is the primary tool used for call transfer in SIP. Beware that not all potential uses of REFER (neither for all methods nor for all URI schemes) are well defined. Implementors should only use the well-defined ones, and should not second guess or freely assume behavior for the others to avoid unexpected behavior of remote UAs, interoperability issues, and other bad surprises.

在RFC 3515中，REFER方法：REFER[RFC3515]定义了请求用户代理发送SIP请求的机制。它是SIP远程控制的一种形式，是SIP中用于呼叫转移的主要工具。请注意，并不是所有潜在的refere用法（既不适用于所有方法，也不适用于所有URI方案）都定义得很好。实现者应该只使用定义良好的，并且不应该猜测或随意假设其他人的行为，以避免远程UAs的意外行为、互操作性问题和其他不好的意外。

RFC 3725, Best Current Practices for Third Party Call Control (3pcc) (B): [RFC3725] defines a number of different call flows that allow one SIP entity, called the controller, to create SIP sessions amongst other SIP user agents.

RFC 3725，第三方呼叫控制（3pcc）（B）的最佳当前实践：[RFC3725]定义了许多不同的呼叫流，允许一个称为控制器的SIP实体在其他SIP用户代理之间创建SIP会话。

RFC 3911, The SIP Join Header Field (S): [RFC3911] defines the Join header field. When sent in an INVITE, it causes the recipient to join the resulting dialog into a conference with another dialog in progress.

RFC 3911，SIP连接头字段：[RFC3911]定义连接头字段。当发送邀请时，它会使收件人将生成的对话框加入到另一个正在进行的对话框的会议中。

RFC 3891, The SIP Replaces Header (S): [RFC3891] defines a mechanism that allows a new dialog to replace an existing dialog. It is useful for certain advanced transfer services.

RFC 3891，SIP替换标头：[RFC3891]定义了一种机制，允许新对话框替换现有对话框。它对于某些高级传输服务很有用。

RFC 3892, The SIP Referred-By Mechanism (S): [RFC3892] defines the Referred-By header field. It is used in requests triggered by REFER, and provides the identity of the referring party to the referred-to party.

RFC 3892，由机制引用的SIP:[RFC3892]定义引用的标头字段。它用于REFER触发的请求中，并将转介方的身份提供给转介方。

RFC 4117, Transcoding Services Invocation in SIP Using Third Party Call Control (I): [RFC4117] defines how to use 3pcc for the purposes of invoking transcoding services for a call.

RFC 4117，SIP中使用第三方呼叫控制的代码转换服务调用（I）：[RFC4117]定义了如何使用3pcc调用调用代码转换服务。

8. Event Framework

8. 事件框架

RFC 3903, SIP Extension for Event State Publication (S): [RFC3903] defines the PUBLISH method. It is not an event package, but is used by all event packages as a mechanism for pushing an event into the system.

RFC 3903，事件状态发布的SIP扩展：[RFC3903]定义发布方法。它不是事件包，但被所有事件包用作将事件推入系统的机制。

RFC 4662, A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists (S): [RFC4662] defines an extension to RFC 3265 that allows a client to subscribe to a list of resources using a single subscription. The server, called a Resource List Server (RLS), will "expand" the subscription and subscribe to each individual member of the list. It has found applicability primarily in the area of presence, but can be used with any event package.

RFC 4662，资源列表的会话启动协议（SIP）事件通知扩展：[RFC4662]定义了RFC 3265的扩展，该扩展允许客户端使用单个订阅订阅资源列表。该服务器称为资源列表服务器（RLS），将“扩展”订阅并订阅列表中的每个成员。它主要适用于现场，但可用于任何事件包。

SUBNOT-ETAGS, An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification (S): [SUBNOT-ETAGS] defines an extension to RFC 3265 to optimize the performance of notifications. When a client subscribes, it can indicate what version of a document it has so that the server can skip sending a notification if the client is up-to-date. It is applicable to any event package.

SUNBNOT-ETAGS，对用于条件事件通知的会话启动协议（SIP）事件的扩展：[SUNBNOT-ETAGS]定义对RFC 3265的扩展，以优化通知的性能。当客户端订阅时，它可以指示文档的版本，以便服务器可以在客户端是最新的情况下跳过发送通知。它适用于任何事件包。

9. Event Packages

9. 事件包

These are event packages defined to utilize the SIP events framework. Many of these are also listed elsewhere in their respective areas.

这些是定义为利用SIP事件框架的事件包。其中许多也列在各自领域的其他地方。

RFC 3680, A SIP Event Package for Registrations (S): [RFC3680] defines an event package for finding out about changes in registration state.

RFC 3680，注册的SIP事件包：[RFC3680]定义了一个事件包，用于查找注册状态的更改。

GRUU-REG (S): [GRUU-REG] is an extension to the registration event package [RFC3680] that allows user agents to learn about their GRUUs. It is particularly useful in helping to synchronize a client and its registrar with their currently valid temporary GRUU.

GRUU-REG:[GRUU-REG]是注册事件包[RFC3680]的扩展，允许用户代理了解其GRUU。它在帮助客户机及其注册器与其当前有效的临时GRUU同步时特别有用。

RFC 3842, A Message Summary and Message Waiting Indication Event Package for SIP (S): [RFC3842] defines a way for a user agent to find out about voicemails and other messages that are waiting for it. Its primary purpose is to enable the voicemail waiting lamp on most business telephones.

RFC 3842是SIP的消息摘要和消息等待指示事件包：[RFC3842]定义了用户代理查找语音邮件和其他正在等待它的消息的方法。其主要目的是在大多数商用电话上启用语音邮件等待灯。

RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an event package for indicating user presence through SIP.

RFC 3856，SIP的存在事件包：[RFC3856]定义了一个事件包，用于通过SIP指示用户存在。

RFC 3857, A Watcher Information Event Template Package for SIP (S): [RFC3857], also known as winfo, provides a mechanism for a user agent to find out what subscriptions are in place for a particular event package. Its primary usage is with presence, but it can be used with any event package.

RFC 3857是SIP的观察者信息事件模板包：[RFC3857]，也称为winfo，它为用户代理提供了一种机制，用于找出特定事件包的订阅。它的主要用途是显示，但它可以用于任何事件包。

RFC 4235, An INVITE-Initiated Dialog Event Package for SIP (S): [RFC4235] defines an event package for learning the state of the dialogs in progress at a user agent, and is one of several RFCs starting with the important number 42 [HGTTG].

RFC 4235是SIP的一个INVITE启动的对话事件包：[RFC4235]定义了一个事件包，用于了解用户代理正在进行的对话的状态，它是以重要数字42[HGTTG]开始的几个RFC之一。

RFC 4575, A SIP Event Package for Conference State (S): [RFC4575] defines a mechanism for learning about changes in conference state, including conference membership.

RFC 4575，会议状态的SIP事件包：[RFC4575]定义了一种了解会议状态变化的机制，包括会议成员资格。

RFC 4730, A SIP Event Package for Key Press Stimulus (KPML) (S): [RFC4730] defines a way for an application in the network to subscribe to the set of key presses made on the keypad of a traditional telephone. It, along with RFC 4733 [RFC4733], are the two mechanisms defined for handling DTMF. RFC 4730 is a signaling-path solution, and RFC 4733 is a media-path solution.

RFC 4730，按键刺激（KPML）（S）的SIP事件包：[RFC4730]定义了网络中的应用程序订阅传统电话键盘上的按键集的方法。它与RFC 4733[RFC4733]一起，是为处理DTMF而定义的两种机制。RFC 4730是信令路径解决方案，RFC 4733是媒体路径解决方案。

RTCP-SUM, SIP Event Package for Voice Quality Reporting (S): [RTCP-SUM] defines a SIP event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions.

RTCP-SUM，用于语音质量报告的SIP事件包：[RTCP-SUM]定义了一个SIP事件包，该事件包支持收集和报告用于测量互联网语音协议（VoIP）会话质量的度量标准。

SESSION-POLICY, A Framework for Session Initiation Protocol (SIP) Session Policies (S): [SESSION-POLICY] defines a framework for session policies. In this framework, policy servers are used to tell user agents about the media characteristics required for a particular session. The session policy framework has not been widely implemented.

会话策略，会话启动协议（SIP）会话策略的框架：[会话策略]定义会话策略的框架。在此框架中，策略服务器用于告诉用户代理特定会话所需的媒体特征。会议政策框架尚未得到广泛实施。

POLICY-PACK, A Session Initiation Protocol (SIP) Event Package for Session-Specific Session Policies (S): [POLICY-PACK] defines a SIP event package used in conjunction with the session policy framework [SESSION-POLICY].

POLICY-PACK是会话特定会话策略的会话启动协议（SIP）事件包：[POLICY-PACK]定义了与会话策略框架[Session-POLICY]一起使用的SIP事件包。

RFC 5362, The Session Initiation Protocol (SIP) Pending Additions Event Package (S): [RFC5362] defines a SIP event package that allows a UA to learn whether consent has been given for the addition of an address to a SIP "mailing list". It is used in conjunction with the SIP framework for consent [RFC5360].

RFC 5362，会话启动协议（SIP）待添加事件包：[RFC5362]定义了一个SIP事件包，该事件包允许UA了解是否已同意将地址添加到SIP“邮件列表”中。它与SIP同意框架结合使用[RFC5360]。

10. Quality of Service

10. 服务质量

Several specifications concern themselves with the interactions of SIP with network Quality of Service (QoS) mechanisms.

一些规范关注SIP与网络服务质量（QoS）机制的交互。

RFC 3312, Integration of Resource Management and SIP (S): [RFC3312], updated by [RFC4032], defines a way to make sure that the phone of the called party doesn't ring until a QoS reservation has been installed in the network. It does so by defining a general preconditions framework, which defines conditions that must be true in order for a SIP session to proceed.

RFC 3312，资源管理和SIP的集成：[RFC3312]，由[RFC4032]更新，定义了一种确保被叫方的电话在网络中安装QoS预留之前不会响的方法。它通过定义一个通用的先决条件框架来实现，该框架定义了SIP会话继续进行所必须满足的条件。

QoS-ID, Quality of Service (QoS) Mechanism Selection in the Session Description Protocol (SDP) (S): [QoS-ID] defines a way for user agents to negotiate what type of end-to-end QoS mechanism to use for a session. At this time, there are two that can be used: the Resource Reservation Protocol (RSVP) and Next Steps in Signaling (NSIS). This negotiation is done through an SDP extension. Due to limited deployment of RSVP and even more limited deployment of NSIS, this extension has not been widely used.

QoS ID，会话描述协议（SDP）（S）中的服务质量（QoS）机制选择：[QoS ID]定义了用户代理协商会话使用何种类型的端到端QoS机制的方法。此时，可以使用两种协议：资源预留协议（RSVP）和信令中的下一步（NSIS）。此协商通过SDP扩展完成。由于RSVP的部署有限，甚至NSIS的部署更为有限，因此该扩展尚未得到广泛应用。

RFC 3313, Private SIP Extensions for Media Authorization (I): [RFC3313] defines a P-header that provides a mechanism for passing an authorization token between SIP and a network QoS reservation protocol like RSVP. Its purpose is to make sure network QoS is

RFC 3313，用于媒体授权的专用SIP扩展（I）：[RFC3313]定义了一个P报头，该报头提供了在SIP和网络QoS保留协议（如RSVP）之间传递授权令牌的机制。它的目的是确保网络QoS是可靠的

only granted if a client has made a SIP call through the same provider's network. This specification is sometimes referred to as the SIP walled-garden specification by the truly paranoid androids in the SIP community. This is because it requires coupling of signaling and the underlying IP network.

仅当客户端通过同一提供商的网络进行SIP呼叫时才授予。该规范有时被SIP社区中真正偏执的机器人称为SIP围墙花园规范。这是因为它需要信令和底层IP网络的耦合。

RFC 3524, Mapping of Media Streams to Resource Reservation Flows (S): [RFC3524] defines a usage of the SDP grouping framework for indicating that a set of media streams should be handled by a single resource reservation.

RFC 3524，媒体流到资源保留流的映射：[RFC3524]定义了SDP分组框架的用法，用于指示一组媒体流应由单个资源保留处理。

11. Operations and Management

11. 业务和管理

Several specifications have been defined to support operations and management of SIP systems. These include mechanisms for configuration and network diagnostics.

已经定义了几个规范来支持SIP系统的操作和管理。这些包括用于配置和网络诊断的机制。

CONFIG-FRAME, A Framework for SIP User Agent Profile Delivery (S): [CONFIG-FRAME] defines a mechanism that allows a SIP user agent to bootstrap its configuration from the network and receive updates to its configuration, should it change. This is considered an essential piece of deploying a usable SIP network.

CONFIG-FRAME是SIP用户代理配置文件传递的框架：[CONFIG-FRAME]定义了一种机制，允许SIP用户代理从网络引导其配置，并在配置发生更改时接收其配置的更新。这被认为是部署可用SIP网络的关键部分。

RTCP-SUM，用于语音质量报告的SIP事件包：[RTCP-SUM]定义了一个SIP事件包，该事件包支持收集和报告用于测量互联网语音协议（VoIP）会话质量的度量标准。

12. SIP Compression

12. SIP压缩

Sigcomp [RFC3320] [RFC4896] was defined to allow compression of SIP messages over low bandwidth links. Sigcomp is not formally part of SIP. However, usage of Sigcomp with SIP has required extensions to SIP.

Sigcomp[RFC3320][RFC4896]被定义为允许在低带宽链路上压缩SIP消息。Sigcomp不是SIP的正式组成部分。然而，Sigcomp与SIP的结合使用需要对SIP进行扩展。

RFC 3486, Compressing SIP (S): [RFC3486] defines a SIP URI parameter that can be used to indicate that a SIP server supports Sigcomp.

RFC 3486，压缩SIP:[RFC3486]定义一个SIP URI参数，该参数可用于指示SIP服务器支持Sigcomp。

RFC 5049, Applying Signaling Compression (SigComp) to the Session Initiation Protocol (SIP) (S): [RFC5049] defines how to apply Sigcomp to SIP.

RFC 5049，将信令压缩（SigComp）应用于会话启动协议（SIP）：[RFC5049]定义了如何将SigComp应用于SIP。

13. SIP Service URIs

13. SIP服务URI

Several extensions define well-known services that can be invoked by constructing requests with specific structures for the Request URI, resulting in specific behaviors at the User Agent Server (UAS).

几个扩展定义了众所周知的服务，可以通过为请求URI构造具有特定结构的请求来调用这些服务，从而在用户代理服务器（UAS）上产生特定的行为。

RFC 3087, Control of Service Context using Request URI (I): [RFC3087] introduced the context of using Request URIs, encoded appropriately, to invoke services.

RFC 3087，使用请求URI控制服务上下文（I）：[RFC3087]介绍了使用请求URI（经过适当编码）调用服务的上下文。

RFC 4662, A SIP Event Notification Extension for Resource Lists (S): [RFC4662] defines a resource called a Resource List Server (RLS). A client can send a subscribe to this server. The server will generate a series of subscriptions, compile the resulting information, and send it back to the subscriber. The set of resources that the RLS will subscribe to is a property of the request URI in the SUBSCRIBE request.

RFC 4662，资源列表的SIP事件通知扩展：[RFC4662]定义了一个称为资源列表服务器（RLS）的资源。客户端可以向该服务器发送订阅。服务器将生成一系列订阅，编译生成的信息，并将其发送回订阅服务器。RLS将订阅的资源集是订阅请求中的请求URI的属性。

RFC 5363, Framework and Security Considerations for Session Initiation Protocol (SIP) Uniform Resource Identifier (URI)-List Services (S): [RFC5363] defines the framework for list services in SIP. In this framework, a UA can include an XML list object in the body of various requests and the server will provide list-oriented services as a consequence. For example, a SUBSCRIBE with a list subscribes to the URI in the list.

RFC 5363，会话启动协议（SIP）统一资源标识符（URI）-列表服务的框架和安全注意事项：[RFC5363]定义了SIP中列表服务的框架。在此框架中，UA可以在各种请求体中包含XML列表对象，因此服务器将提供面向列表的服务。例如，具有列表的订阅将订阅列表中的URI。

RFC 5367, Subscriptions To Request-Contained Resource Lists in SIP (S): [RFC5367] uses the URI-list framework [RFC5363] and allows a client to subscribe to a resource called a Resource List Server. This server will generate subscriptions to the URI in the list, compile the resulting information, and send it back to the subscriber.

RFC 5367，订阅SIP中包含的请求资源列表：[RFC5367]使用URI列表框架[RFC5363]，并允许客户端订阅称为资源列表服务器的资源。此服务器将生成对列表中URI的订阅，编译生成的信息，并将其发送回订阅服务器。

RFC 5365, Multiple-Recipient MESSAGE Requests in SIP (S): [RFC5365] uses the URI-list framework [RFC5363] and allows a client to send a MESSAGE to a number of recipients.

RFC 5365，SIP中的多个收件人消息请求：[RFC5365]使用URI列表框架[RFC5363]，允许客户端向多个收件人发送消息。

RFC 5366, Conference Establishment Using Request-Contained Lists in SIP (S): [RFC5366] uses the URI-list framework [RFC5363]. It allows a client to ask the server to act as a conference focus and send an invitation to each recipient in the list.

RFC 5366，使用SIP中包含请求的列表建立会议：[RFC5366]使用URI列表框架[RFC5363]。它允许客户端请求服务器充当会议焦点，并向列表中的每个收件人发送邀请。

RFC 4240, Basic Network Media Services with SIP (I): [RFC4240] defines a way for SIP application servers to invoke announcement and conferencing services from a media server. This is accomplished through a set of defined URI parameters that tell the media server what to do, such as what file to play and what language to render it in.

RFC 4240，带SIP的基本网络媒体服务（I）：[RFC4240]定义了SIP应用服务器从媒体服务器调用公告和会议服务的方法。这是通过一组定义的URI参数来实现的，这些参数告诉媒体服务器要做什么，例如播放什么文件以及用什么语言呈现它。

RFC 4458, Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and Interactive Voice Response (IVR) (I): [RFC4458] defines a way to invoke voicemail and IVR services by using a SIP URI constructed in a particular way.

RFC 4458，语音邮件和交互式语音响应（IVR）等应用程序的会话启动协议（SIP）URI（I）：[RFC4458]定义了通过使用以特定方式构造的SIP URI调用语音邮件和IVR服务的方法。

14. Minor Extensions

14. 小扩展

These SIP extensions don't fit easily into a single specific use case. They have somewhat general applicability, but they solve a relatively small problem or provide an optimization.

这些SIP扩展不容易适应单个特定的用例。它们具有一定的普遍适用性，但可以解决相对较小的问题或提供优化。

RFC 4488, Suppression of the SIP REFER Implicit Subscription (S): [RFC4488] defines an enhancement to REFER. REFER normally creates an implicit subscription to the target of the REFER. This subscription is used to pass back updates on the progress of the referral. This extension allows that implicit subscription to be bypassed as an optimization.

RFC 4488，抑制SIP REFER隐式订阅：[RFC4488]定义了要引用的增强。refere通常创建对refere的目标的隐式订阅。此订阅用于传回有关推荐进度的更新。此扩展允许作为优化绕过隐式订阅。

RFC 4538, Request Authorization through Dialog Identification in SIP (S): [RFC4538] provides a mechanism that allows a UAS to authorize a request because the requestor proves it knows a dialog that is in progress with the UAS. The specification is useful in conjunction with the SIP application interaction framework [INTERACT-FRAME].

RFC 4538，通过SIP中的对话标识进行的请求授权：[RFC4538]提供了一种机制，允许UAS授权请求，因为请求者证明其知道与UAS正在进行的对话。该规范与SIP应用程序交互框架[INTERACT-FRAME]结合使用非常有用。

RFC 4508, Conveying Feature Tags with the REFER Method in SIP (S): [RFC4508] defines a mechanism for carrying RFC 3840 feature tags in REFER. It is useful for informing the target of the REFER about the characteristics of the intended target of the referred request.

RFC 4508，使用SIP中的REFER方法传送功能标签：[RFC4508]定义了在REFER中传送RFC 3840功能标签的机制。它有助于将所引用请求的预期目标的特征告知所引用的目标。

RFC 5373, Requesting Answer Modes for SIP (S): [RFC5373] defines an extension for indicating to the called party whether or not the phone should ring and/or be answered immediately. This is useful for push-to-talk and for diagnostic applications.

RFC 5373，为SIP请求应答模式：[RFC5373]定义了一个扩展，用于向被叫方指示电话是否应该响铃和/或立即应答。这对于按键通话和诊断应用非常有用。

RFC 5079, Rejecting Anonymous Requests in SIP (S): [RFC5079] defines a mechanism for a called party to indicate to the calling party that a call was rejected since the caller was anonymous. This is needed for implementation of the Anonymous Call Rejection (ACR) feature in SIP.

RFC 5079，拒绝SIP中的匿名请求：[RFC5079]定义了被叫方向主叫方指示呼叫被拒绝的机制，因为主叫方是匿名的。这是在SIP中实现匿名呼叫拒绝（ACR）功能所必需的。

RFC 5368, Referring to Multiple Resources in SIP (S): [RFC5368] allows a UA sending a REFER to ask the recipient of the REFER to generate multiple SIP requests, not just one. This is useful for conferencing, where a client would like to ask a conference server to eject multiple users.

RFC 5368，引用SIP中的多个资源：[RFC5368]允许UA发送REFER，要求REFER的接收者生成多个SIP请求，而不仅仅是一个。这对于会议非常有用，客户机希望请求会议服务器弹出多个用户。

RFC 4483, A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages (S): [RFC4483] defines a mechanism for content indirection. Instead of carrying an object within a SIP body, a URL reference is carried instead, and the recipient dereferences the URL to obtain the object. The specification has potential applicability for sending large instant messages, but

RFC 4483，会话初始化协议（SIP）消息中的内容间接寻址机制：[RFC4483]定义了内容间接寻址机制。不是在SIP主体中携带对象，而是携带URL引用，收件人取消引用URL以获取对象。该规范可能适用于发送大型即时消息，但

has yet to find much actual use.

还没有找到多少实际用途。

RFC 3890, A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP) (S): [RFC3890] specifies an SDP extension that allows for the description of the bandwidth for a media session that is independent of the underlying transport mechanism.

RFC 3890，会话描述协议（SDP）（S）的传输无关带宽修饰符：[RFC3890]指定SDP扩展，该扩展允许描述独立于底层传输机制的媒体会话的带宽。

RFC 4583, Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams (S): [RFC4583] defines a mechanism in SDP to signal floor control streams that use BFCP. It is used for push-to-talk and conference floor control.

RFC 4583，二进制地板控制协议（BFCP）流的会话描述协议（SDP）格式：[RFC4583]在SDP中定义了一种机制，用于向使用BFCP的地板控制流发送信号。它用于按键通话和会议楼层控制。

CONNECT-PRECON, Connectivity Preconditions for Session Description Protocol Media Streams (S): [CONNECT-PRECON] defines a usage of the precondition framework [RFC3312]. The connectivity precondition makes sure that the session doesn't get established until actual packet connectivity is checked.

CONNECT-PRECON，会话描述协议媒体流的连接先决条件：[CONNECT-PRECON]定义先决条件框架[RFC3312]的用法。连接前提条件确保在检查实际数据包连接之前不会建立会话。

RFC 4796, The SDP (Session Description Protocol) Content Attribute (S): [RFC4796] defines an SDP attribute for describing the purpose of a media stream. Examples include a slide view, the speaker, a sign language feed, and so on.

在RFC 4796中，SDP（会话描述协议）内容属性：[RFC4796]定义了用于描述媒体流用途的SDP属性。示例包括幻灯片视图、说话人、手语提要等。

IPv6-TRANS, IPv6 Transition in the Session Initiation Protocol (SIP) (S): [IPv6-TRANS] defines practices for interworking between IPv6 and IPv6 user agents. This is done through multi-homed proxies that interwork IPv4 and IPv6, along with ICE [ICE] for media traversal. The specification includes some minor extensions and clarifications to SDP in order to cover some additional cases.

会话启动协议（SIP）（S）中的IPv6传输和IPv6转换：[IPv6传输]定义了IPv6和IPv6用户代理之间互通的实践。这是通过将IPv4和IPv6与ICE[ICE]进行交互的多宿主代理完成的，ICE[ICE]用于媒体遍历。本规范包括对SDP的一些小扩展和澄清，以涵盖一些其他情况。

CONNECT-REUSE, Connection Reuse in the Session Initiation Protocol (SIP) (S): [CONNECT-REUSE] defines an extension to SIP that allows a Transport Layer Security (TLS) connection between servers to be reused for requests in both directions. Normally, two connections are set up between a pair of servers, one for requests in each direction.

连接重用，会话启动协议（SIP）（S）中的连接重用：[连接重用]定义了SIP的扩展，该扩展允许服务器之间的传输层安全（TLS）连接在两个方向上的请求中重用。通常，在一对服务器之间设置两个连接，每个方向一个用于请求。

15. Security Mechanisms

15. 安全机制

Several extensions provide additional security features to SIP.

一些扩展为SIP提供了额外的安全特性。

RFC 4916, Connected Identity in the Session Initiation Protocol (SIP) (S): [RFC4916] formally updates RFC 3261. It defines an extension to SIP that allows a calling user to determine the identity of the final called user (connected party). Due to forwarding and retargeting services, this may not be the same as the user that the caller was originally trying to reach. The mechanism works in tandem with the SIP identity specification [RFC4474] to provide signatures over the connected party identity. It can also be used if a party identity changes mid call due to third party call control actions or PSTN behavior.

RFC 4916，会话启动协议（SIP）中的连接标识：[RFC4916]正式更新RFC 3261。它定义了SIP的扩展，允许主叫用户确定最终被叫用户（连接方）的身份。由于转发和重定目标服务，这可能与调用者最初试图联系的用户不同。该机制与SIP身份规范[RFC4474]协同工作，通过连接方身份提供签名。如果由于第三方呼叫控制操作或PSTN行为，当一方身份在通话中发生更改时，也可以使用此选项。

DOMAIN-CERTS, Domain Certificates in the Session Initiation Protocol (SIP) (B): [DOMAIN-CERTS] clarifies the usage of SIP over TLS with regards to certificate handling, and defines additional procedures needed for interoperability.

域证书，会话启动协议（SIP）（B）中的域证书：[域证书]澄清了SIP over TLS在证书处理方面的用法，并定义了互操作性所需的其他过程。

RFC 4567, Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP) (S): [RFC4567] defines extensions to SDP that allow tunneling of a key management protocol, namely MIKEY [RFC3830], through offer/answer exchanges. This mechanism is one of three Secure Realtime Transport Protocol (SRTP) keying techniques specified for SIP, with Datagram Transport Layer Security (DTLS)-SRTP [SRTP-FRAME] having been selected as the final solution.

RFC 4567，会话描述协议（SDP）和实时流协议（RTSP）的密钥管理扩展：[RFC4567]定义了SDP的扩展，允许通过提供/应答交换对密钥管理协议（即MIKEY[RFC3830]）进行隧道传输。该机制是为SIP指定的三种安全实时传输协议（SRTP）密钥技术之一，数据报传输层安全性（DTLS）-SRTP[SRTP-FRAME]已被选为最终解决方案。

RFC 4568, Session Description Protocol (SDP) Security Descriptions for Media Streams (S): [RFC4568] defines extensions to SDP that allow for the negotiation of keying material directly through offer/answer, without a separate key management protocol. This mechanism, sometimes called sdescriptions, has the drawback that the media keys are available to any entity that has visibility to the SDP. It is one of three SRTP keying techniques specified for SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the final solution.

RFC 4568，媒体流的会话描述协议（SDP）安全描述：[RFC4568]定义了SDP的扩展，允许直接通过提供/应答协商密钥材料，而无需单独的密钥管理协议。这种机制（有时称为sdescriptions）的缺点是，任何对SDP可见的实体都可以使用媒体密钥。它是为SIP指定的三种SRTP键控技术之一，DTLS-SRTP[SRTP-FRAME]已被选为最终解决方案。

SRTP-FRAME, Framework for Establishing an SRTP Security Context using DTLS (S): [SRTP-FRAME] defines the overall framework and SDP and SIP processing required to perform key management for RTP using Datagram TLS (DTLS) [RFC4347] directly between endpoints, over the media path. It is one of three SRTP keying techniques specified for SIP, with DTLS-SRTP [SRTP-FRAME] having been selected as the final solution.

SRTP-FRAME，使用DTL建立SRTP安全上下文的框架：[SRTP-FRAME]定义了在端点之间通过媒体路径直接使用数据报TLS（DTL）[RFC4347]执行RTP密钥管理所需的总体框架以及SDP和SIP处理。它是为SIP指定的三种SRTP键控技术之一，DTLS-SRTP[SRTP-FRAME]已被选为最终解决方案。

RFC 3853, S/MIME Advanced Encryption Standard (AES) Requirement for SIP (S): [RFC3853] formally updates RFC 3261. It is a brief specification that updates the cryptography mechanisms used in SIP S/MIME. However, SIP S/MIME has seen very little deployment.

RFC 3853，SIP的S/MIME高级加密标准（AES）要求：[RFC3853]正式更新RFC 3261。这是一个简短的规范，用于更新SIP S/MIME中使用的加密机制。然而，SIP S/MIME很少部署。

CERTS, Certificate Management Service for the Session Initiation Protocol (SIP) (S): [CERTS] defines a certificate service for SIP whose purpose is to facilitate the deployment of S/MIME. The certificate service allows clients to store and retrieve their own certificates, in addition to obtaining the certificates for other users.

CERTS，会话启动协议（SIP）的证书管理服务：[CERTS]为SIP定义了一个证书服务，其目的是促进S/MIME的部署。除了为其他用户获取证书外，证书服务还允许客户端存储和检索自己的证书。

RFC 3893, Session Initiation Protocol (SIP) Authenticated Identity Body (AIB) Format (S): [RFC3893] defines a SIP message fragment that can be signed in order to provide an authenticated identity over a request. It was an early predecessor to [RFC4474], and consequently AIB has seen no deployment.

RFC 3893，会话启动协议（SIP）身份验证主体（AIB）格式：[RFC3893]定义了可以签名的SIP消息片段，以便通过请求提供身份验证。它是[RFC4474]的早期前身，因此AIB没有部署。

SAML, SIP SAML Profile and Binding (S): [SAML] defines the usage of the Security Assertion Markup Language (SAML) within SIP, and describes how to use it in conjunction with SIP identity [RFC4474] to provide authenticated assertions about a user's role or attributes.

SAML、SIP SAML配置文件和绑定：[SAML]定义了SIP中安全断言标记语言（SAML）的用法，并描述了如何将其与SIP标识[RFC4474]结合使用，以提供有关用户角色或属性的经过身份验证的断言。

RFC 5360, A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP) (S): [RFC5360] defines several extensions to SIP, including the Trigger-Consent and Permission-Missing header fields. These header fields, in addition to the other procedures defined in the document, define a way to manage membership on "SIP mailing lists" used for instant messaging or conferencing. In particular, it helps avoid the problem of using such amplification services for the purposes of an attack on the network by making sure a user authorizes the addition of their address onto such a service.

RFC 5360是会话启动协议（SIP）中基于同意的通信框架：[RFC5360]定义了SIP的几个扩展，包括触发同意和许可缺失头字段。除了文档中定义的其他过程外，这些标题字段还定义了管理用于即时消息或会议的“SIP邮件列表”成员资格的方法。特别是，通过确保用户授权将其地址添加到此类服务上，它有助于避免将此类放大服务用于网络攻击的问题。

RFC 5361, A Document Format for Requesting Consent (S): [RFC5361] defines an XML object used by the consent framework. Consent documents are sent from SIP "mailing list servers" to users to allow them to manage their membership on lists.

RFC 5361，一种用于请求同意的文档格式：[RFC5361]定义了同意框架使用的XML对象。同意文件从SIP“邮件列表服务器”发送给用户，以允许用户管理其在列表中的成员资格。

RFC 3329, Security Mechanism Agreement for SIP (S): [RFC3329] defines a mechanism to prevent bid-down attacks in conjunction with SIP authentication. The mechanism has seen very limited deployment. It was defined as part of the 3GPP IP Multimedia Subsystem (IMS) specification suite [3GPP.24.229], and is needed only when there is a multiplicity of security mechanisms deployed at a particular server. In practice, this has not been the case.

RFC 3329，SIP的安全机制协议：[RFC3329]定义了一种与SIP身份验证相结合的机制，以防止出价下降攻击。该机制的部署非常有限。它被定义为3GPP IP多媒体子系统（IMS）规范套件[3GPP.24.229]的一部分，只有在特定服务器上部署了多种安全机制时才需要它。实际上，情况并非如此。

RFC 4572, Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP) (S): [RFC4572] specifies a mechanism for signaling TLS-based media streams between endpoints. It expands the TCP-based media signaling parameters defined in [RFC4145] to include fingerprint information for TLS streams so that TLS can operate between end hosts using self-signed certificates.

RFC 4572，《会话描述协议（SDP）（S）中传输层安全（TLS）协议上的面向连接的媒体传输》：[RFC4572]指定了在端点之间发送基于TLS的媒体流的信令的机制。它扩展了[RFC4145]中定义的基于TCP的媒体信令参数，以包括TLS流的指纹信息，以便TLS可以使用自签名证书在终端主机之间运行。

RFC 5027, Security Preconditions for Session Description Protocol Media Streams (S): [RFC5027] defines a precondition for use with the preconditions framework [RFC3312]. The security precondition prevents a session from being established until a security media stream is set up.

RFC 5027，会话描述协议媒体流的安全先决条件：[RFC5027]定义了与先决条件框架[RFC3312]一起使用的先决条件。安全先决条件防止在建立安全媒体流之前建立会话。

RFC 3310, Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (S): [RFC3310] defines an extension to digest authentication to allow it to work with the credentials stored in cell phones. Though technically it is an extension to HTTP digest, its primary application is SIP. This extension is useful primarily to implementors of IMS.

RFC 3310，使用身份验证和密钥协议的超文本传输协议（HTTP）摘要身份验证：[RFC3310]定义了摘要身份验证的扩展，以允许其使用手机中存储的凭据。虽然从技术上讲，它是HTTP摘要的扩展，但它的主要应用程序是SIP。此扩展主要对IMS的实现人员有用。

RFC 4169, Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA) Version-2 (S): [RFC4169] is an enhancement to [RFC3310] that further improves security of the authentication.

RFC 4169，使用身份验证和密钥协议（AKA）版本2的超文本传输协议（HTTP）摘要身份验证：[RFC4169]是对[RFC3310]的增强，进一步提高了身份验证的安全性。

16. Conferencing

16. 会议

Numerous SIP and SDP extensions are aimed at conferencing as their primary application.

许多SIP和SDP扩展都将会议作为其主要应用程序。

RFC 4574, The SDP (Session Description Protocol) Label Attribute (S): [RFC4574] defines an SDP attribute for providing an opaque label for media streams. These labels can be referred to by external documents, and in particular, by conference policy documents. This allows a UA to tie together documents it may obtain through conferencing mechanisms to media streams to which they refer.

RFC 4574中的SDP（会话描述协议）标签属性：[RFC4574]定义了一个SDP属性，用于为媒体流提供不透明标签。这些标签可由外部文件引用，特别是由会议政策文件引用。这允许UA将通过会议机制获得的文档与它们所引用的媒体流绑定在一起。

RFC 3911，SIP连接头字段：[RFC3911]定义连接头字段。当发送邀请时，它会使收件人将生成的对话框加入到另一个正在进行的对话框的会议中。

RFC 4575, A SIP Event Package for Conference State (S): [RFC4575] defines a mechanism for learning about changes in conference state, including conference membership.

RFC 4575，会议状态的SIP事件包：[RFC4575]定义了一种了解会议状态变化的机制，包括会议成员资格。

RFC 5366, Conference Establishment Using Request-Contained Lists in SIP (S): [RFC5366] is similar to [RFC5367]. However, instead of subscribing to the resource, an INVITE request is sent to the resource, and it will act as a conference focus and generate an invitation to each recipient in the list.

RFC 5366，使用SIP中包含的请求列表建立会议：[RFC5366]与[RFC5367]类似。但是，不是订阅资源，而是向资源发送邀请请求，它将充当会议焦点，并向列表中的每个收件人生成邀请。

RFC4579, Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents (B): [RFC4579] defines best practice procedures and call flows for conferencing. This includes conference creation, joining, and dial out, amongst other capabilities.

RFC4579，会话启动协议（SIP）呼叫控制-用户代理会议（B）：[RFC4579]定义了会议的最佳实践程序和呼叫流。这包括会议创建、加入和拨出等功能。

17. Instant Messaging, Presence, and Multimedia

17. 即时消息、状态和多媒体

SIP provides extensions for instant messaging, presence, and multimedia.

SIP提供即时消息、状态和多媒体的扩展。

RFC 3428, SIP Extension for Instant Messaging (S): [RFC3428] defines the MESSAGE method, used for sending an instant message without setting up a session (sometimes called "page mode").

RFC 3428，即时消息的SIP扩展：[RFC3428]定义了用于发送即时消息而无需设置会话（有时称为“页面模式”）的消息方法。

RFC 3856, A Presence Event Package for SIP (S): [RFC3856] defines an event package for indicating user presence through SIP.

RFC 3856，SIP的存在事件包：[RFC3856]定义了一个事件包，用于通过SIP指示用户存在。

TRANSFER-MECH, A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer (S): [TRANSFER-MECH] defines a mechanism for signaling a file transfer session with SIP.

TRANSFER-MECH，一种用于启用文件传输的会话描述协议（SDP）提供/应答机制：[TRANSFER-MECH]定义了一种用于向SIP发送文件传输会话信号的机制。

18. Emergency Services

18. 紧急服务

Emergency services include preemption features, which allow authorized individuals to gain access to network resources in time of emergency, along with traditional emergency calling.

紧急服务包括抢占功能，允许授权人员在紧急情况下访问网络资源，以及传统的紧急呼叫。

RFC 4411, Extending the SIP Reason Header for Preemption Events (S): [RFC4411] defines an extension to the Reason header, allowing a UA to know that its dialog was torn down because a higher priority session came through.

RFC 4411，扩展抢占事件的SIP原因头：[RFC4411]定义了原因头的扩展，允许UA知道其对话已被中断，因为有更高优先级的会话通过。

RFC 4412, Communications Resource Priority for SIP (S): [RFC4412] defines a new header field, Resource-Priority, that allows a session to get priority treatment from the network.

RFC 4412，SIP的通信资源优先级：[RFC4412]定义了一个新的头字段Resource Priority，该字段允许会话从网络获得优先级处理。

LOCATION, Location Conveyance for the Session Initiation Protocol (S): [LOCATION] defines a mechanism for carrying location objects in SIP messages. This is used to convey location from a UA to an emergency call taker.

会话启动协议的位置、位置传输：[LOCATION]定义了在SIP消息中承载位置对象的机制。这用于将位置从UA传达给紧急呼叫接受者。

19. Security Considerations

19. 安全考虑

This specification is an overview of existing specifications and does not introduce any security considerations on its own. Of course, the world would be far more secure if everyone would follow one simple rule: "Don't Panic!" [HGTTG].

本规范是对现有规范的概述，不单独介绍任何安全注意事项。当然，如果每个人都遵循一条简单的规则：“不要惊慌！”[HGTTG]，世界将更加安全。

20. Acknowledgements

20. 致谢

The author would like to thank Spencer Dawkins, Brian Stucker, Keith Drage, John Elwell, and Avshalom Houri for their comments on this

作者要感谢斯宾塞·道金斯、布赖恩·斯图克、基思·德拉格、约翰·埃尔威尔和阿夫沙洛姆·霍里对此发表的评论

document.

文件

21. Informative References

21. 资料性引用

[3GPP.24.229] 3GPP, "Internet Protocol (IP) multimedia call control protocol based on Session Initiation Protocol (SIP) and Session Description Protocol (SDP); Stage 3", 3GPP TS 24.229 5.22.0, September 2008.

[3GPP.24.229]3GPP，“基于会话发起协议（SIP）和会话描述协议（SDP）的互联网协议（IP）多媒体呼叫控制协议；第3阶段”，3GPP TS 24.229 5.22.012008年9月。

[ABNF-FIX] Gurbani, V. and B. Carpenter, "Essential correction for IPv6 ABNF in RFC3261", Work in Progress, November 2007.

[ABNF-FIX]Gurbani，V.和B.Carpenter，“RFC3261中IPv6 ABNF的基本纠正”，正在进行的工作，2007年11月。

[BODY-HANDLING] Camarillo, G., "Message Body Handling in the Session Initiation Protocol (SIP)", Work in Progress, November 2008.

[BODY-HANDLING]Camarillo，G.，“会话启动协议（SIP）中的消息正文处理”，正在进行的工作，2008年11月。

[CERTS] Jennings, C. and J. Fischl, "Certificate Management Service for The Session Initiation Protocol (SIP)", Work in Progress, November 2008.

[CERTS]Jennings，C.和J.Fischl，“会话启动协议（SIP）的证书管理服务”，正在进行的工作，2008年11月。

[CONFIG-FRAME] Channabasappa, S., "A Framework for Session Initiation Protocol User Agent Profile Delivery", Work in Progress, February 2008.

[CONFIG-FRAME]Channabasappa，S.，“会话启动协议用户代理配置文件交付框架”，正在进行的工作，2008年2月。

[CONNECT-PRECON] Andreasen, F., Camarillo, G., Oran, D., and D. Wing, "Connectivity Preconditions for Session Description Protocol Media Streams", Work in Progress, October 2008.

[CONNECT-Preco]Andreasen，F.，Camarillo，G.，Oran，D.，和D.Wing，“会话描述协议媒体流的连接先决条件”，正在进行的工作，2008年10月。

[CONNECT-REUSE] Gurbani, V., Mahy, R., and B. Tate, "Connection Reuse in the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[连接重用]Gurbani，V.，Mahy，R.，和B.Tate，“会话启动协议（SIP）中的连接重用”，正在进行的工作，2008年10月。

[DOMAIN-CERTS] Gurbani, V., Lawrence, S., and B. Laboratories, "Domain Certificates in the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[DOMAIN-CERTS]Gurbani，V.，Lawrence，S.，and B.Laboratories，“会话启动协议（SIP）中的域证书”，正在进行的工作，2008年10月。

[ECRIT-FRAME] Rosen, B., Schulzrinne, H., Polk, J., and A. Newton, "Framework for Emergency Calling using Internet Multimedia", Work in Progress, July 2008.

[ECRIT-FRAME]Rosen，B.，Schulzrinne，H.，Polk，J.，和A.Newton，“使用互联网多媒体进行紧急呼叫的框架”，正在进行的工作，2008年7月。

[GRUU] Rosenberg, J., "Obtaining and Using Globally Routable User Agent (UA) URIs (GRUU) in the Session Initiation Protocol (SIP)", Work in Progress, October 2007.

[GRUU]Rosenberg，J.，“在会话启动协议（SIP）中获取和使用全局可路由用户代理（UA）URI（GRUU）”，正在进行的工作，2007年10月。

[GRUU-REG] Kyzivat, P., "Registration Event Package Extension for Session Initiation Protocol (SIP) Globally Routable User Agent URIs (GRUUs)", Work in Progress, July 2007.

[GRUU-REG]Kyzivat，P.，“会话启动协议（SIP）全局可路由用户代理URI（GRUUs）的注册事件包扩展”，正在进行的工作，2007年7月。

[HGTTG] Adams, D., "The Hitchhiker's Guide to the Galaxy", September 1979.

[HGTTG]亚当斯，D.，《银河系搭便车指南》，1979年9月。

[ICE] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", Work in Progress, October 2007.

[ICE]Rosenberg，J.，“交互式连接建立（ICE）：提供/应答协议的网络地址转换器（NAT）遍历协议”，正在进行的工作，2007年10月。

[ICE-TCP] Rosenberg, J., "TCP Candidates with Interactive Connectivity Establishment (ICE)", Work in Progress, July 2008.

[ICE-TCP]Rosenberg，J.，“具有交互式连接建立（ICE）的TCP候选者”，正在进行的工作，2008年7月。

[INTERACT-FRAME] Rosenberg, J., "A Framework for Application Interaction in the Session Initiation Protocol (SIP)", Work in Progress, July 2005.

[INTERACT-FRAME]Rosenberg，J.，“会话启动协议（SIP）中的应用程序交互框架”，正在进行的工作，2005年7月。

[IPv6-TRANS] Camarillo, G., "IPv6 Transition in the Session Initiation Protocol (SIP)", Work in Progress, August 2007.

[IPv6 TRANS]Camarillo，G.，“会话启动协议（SIP）中的IPv6转换”，正在进行的工作，2007年8月。

[LOCATION] Polk, J. and B. Rosen, "Location Conveyance for the Session Initiation Protocol", Work in Progress, November 2008.

[位置]Polk，J.和B.Rosen，“会话启动协议的位置传输”，正在进行的工作，2008年11月。

[LOOP-FIX] Sparks, R., Lawrence, S., Hawrylyshen, A., and B. Campen, "Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies", Work in Progress, October 2008.

[LOOP-FIX]Sparks，R.，Lawrence，S.，Hawrylyshen，A.，和B.Campen，“解决会话启动协议（SIP）分叉代理中的放大漏洞”，正在进行的工作，2008年10月。

[OPTION-TAG] Rosenberg, J., "Indicating Support for Interactive Connectivity Establishment (ICE) in the Session Initiation Protocol (SIP)", Work in Progress, June 2007.

[OPTION-TAG]Rosenberg，J.，“表示支持会话启动协议（SIP）中的交互式连接建立（ICE）”，正在进行的工作，2007年6月。

[OUTBOUND] Jennings, C. and R. Mahy, "Managing Client Initiated Connections in the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[OUTBOUND]Jennings，C.和R.Mahy，“在会话启动协议（SIP）中管理客户端启动的连接”，正在进行的工作，2008年10月。

[POLICY-PACK] Hilt, V. and G. Camarillo, "A Session Initiation Protocol (SIP) Event Package for Session-Specific Session Policies.", Work in Progress, July 2008.

[POLICY-PACK]Hilt，V.和G.Camarillo，“针对特定会话策略的会话启动协议（SIP）事件包”，《正在进行的工作》，2008年7月。

[QoS-ID] Polk, J., Dhesikan, S., and G. Camarillo, "Quality

[QoS ID]Polk，J.，Dhesikan，S.，和G.Camarillo，“质量

of Service (QoS) Mechanism Selection in the Session Description Protocol (SDP)", Work in Progress, November 2008.

“会话描述协议（SDP）中服务质量（QoS）机制的选择”，正在进行的工作，2008年11月。

[RECORD-ROUTE] Froment, T., Lebel, C., and B. Bonnaerens, "Addressing Record-Route issues in the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[RECORD-ROUTE]Froment，T.，Lebel，C.，和B.Bonnaerens，“解决会话启动协议（SIP）中的记录路由问题”，正在进行的工作，2008年10月。

[RFC2026] Bradner, S., "The Internet Standards Process -- Revision 3", BCP 9, RFC 2026, October 1996.

[RFC2026]Bradner，S.，“互联网标准过程——第3版”，BCP 9，RFC 2026，1996年10月。

[RFC2543] Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg, "SIP: Session Initiation Protocol", RFC 2543, March 1999.

[RFC2543]Handley，M.，Schulzrinne，H.，Schooler，E.，和J.Rosenberg，“SIP：会话启动协议”，RFC 25431999年3月。

[RFC2782] Gulbrandsen, A., Vixie, P., and L. Esibov, "A DNS RR for specifying the location of services (DNS SRV)", RFC 2782, February 2000.

[RFC2782]Gulbrandsen，A.，Vixie，P.和L.Esibov，“用于指定服务位置（DNS SRV）的DNS RR”，RFC 2782，2000年2月。

[RFC2848] Petrack, S. and L. Conroy, "The PINT Service Protocol: Extensions to SIP and SDP for IP Access to Telephone Call Services", RFC 2848, June 2000.

[RFC2848]Petrack，S.和L.Conroy，“PINT服务协议：电话呼叫服务IP访问的SIP和SDP扩展”，RFC 28482000年6月。

[RFC2976] Donovan, S., "The SIP INFO Method", RFC 2976, October 2000.

[RFC2976]Donovan，S.，“SIP信息方法”，RFC 29762000年10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。

[RFC3204] Zimmerer, E., Peterson, J., Vemuri, A., Ong, L., Audet, F., Watson, M., and M. Zonoun, "MIME media types for ISUP and QSIG Objects", RFC 3204, December 2001.

[RFC3204]Zimmerer，E.，Peterson，J.，Vemuri，A.，Ong，L.，Audet，F.，Watson，M.，和M.Zonoun，“ISUP和QSIG对象的MIME媒体类型”，RFC 32042001年12月。

[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002.

[RFC3261]Rosenberg，J.，Schulzrinne，H.，Camarillo，G.，Johnston，A.，Peterson，J.，Sparks，R.，Handley，M.，和E.Schooler，“SIP：会话启动协议”，RFC 3261，2002年6月。

[RFC3262] Rosenberg, J. and H. Schulzrinne, "Reliability of Provisional Responses in Session Initiation Protocol (SIP)", RFC 3262, June 2002.

[RFC3262]Rosenberg，J.和H.Schulzrinne，“会话启动协议（SIP）中临时响应的可靠性”，RFC 32622，2002年6月。

[RFC3263] Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol (SIP): Locating SIP Servers", RFC 3263, June 2002.

[RFC3263]Rosenberg，J.和H.Schulzrinne，“会话启动协议（SIP）：定位SIP服务器”，RFC 3263，2002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.

[RFC3265]Roach，A.，“会话启动协议（SIP）-特定事件通知”，RFC3265，2002年6月。

[RFC3310] Niemi, A., Arkko, J., and V. Torvinen, "Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA)", RFC 3310, September 2002.

[RFC3310]Niemi，A.，Arkko，J.，和V.Torvinen，“使用身份验证和密钥协议（AKA）的超文本传输协议（HTTP）摘要身份验证”，RFC 331102002年9月。

[RFC3311] Rosenberg, J., "The Session Initiation Protocol (SIP) UPDATE Method", RFC 3311, October 2002.

[RFC3311]Rosenberg，J.，“会话启动协议（SIP）更新方法”，RFC3311，2002年10月。

[RFC3312] Camarillo, G., Marshall, W., and J. Rosenberg, "Integration of Resource Management and Session Initiation Protocol (SIP)", RFC 3312, October 2002.

[RFC3312]Camarillo，G.，Marshall，W.，和J.Rosenberg，“资源管理和会话启动协议（SIP）的集成”，RFC 3312，2002年10月。

[RFC3313] Marshall, W., "Private Session Initiation Protocol (SIP) Extensions for Media Authorization", RFC 3313, January 2003.

[RFC3313]Marshall，W.“用于媒体授权的专用会话启动协议（SIP）扩展”，RFC 3313，2003年1月。

[RFC3320] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu, Z., and J. Rosenberg, "Signaling Compression (SigComp)", RFC 3320, January 2003.

[RFC3320]Price，R.，Bormann，C.，Christofferson，J.，Hannu，H.，Liu，Z.，和J.Rosenberg，“信号压缩（SigComp）”，RFC3320，2003年1月。

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

[RFC3323]Peterson，J.，“会话启动协议（SIP）的隐私机制”，RFC3323，2002年11月。

[RFC3325] Jennings, C., Peterson, J., and M. Watson, "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity within Trusted Networks", RFC 3325, November 2002.

[RFC3325]Jennings，C.，Peterson，J.，和M.Watson，“在可信网络中声明身份的会话启动协议（SIP）的私有扩展”，RFC 33252002年11月。

[RFC3326] Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason Header Field for the Session Initiation Protocol (SIP)", RFC 3326, December 2002.

[RFC3326]Schulzrinne，H.，Oran，D.，和G.Camarillo，“会话启动协议（SIP）的原因头字段”，RFC 3326，2002年12月。

[RFC3327] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP) Extension Header Field for Registering Non-Adjacent Contacts", RFC 3327, December 2002.

[RFC3327]Willis，D.和B.Hoeneisen，“用于注册非相邻联系人的会话启动协议（SIP）扩展头字段”，RFC 3327，2002年12月。

[RFC3329] Arkko, J., Torvinen, V., Camarillo, G., Niemi, A., and T. Haukka, "Security Mechanism Agreement for the Session Initiation Protocol (SIP)", RFC 3329,

[RFC3329]Arkko，J.，Torvinen，V.，Camarillo，G.，Niemi，A.，和T.Haukka，“会话启动协议（SIP）的安全机制协议”，RFC 3329，

January 2003.

2003年1月。

[RFC3372] Vemuri, A. and J. Peterson, "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures", BCP 63, RFC 3372, September 2002.

[RFC3372]Vemuri，A.和J.Peterson，“电话会话启动协议（SIP-T）：上下文和体系结构”，BCP 63，RFC 3372，2002年9月。

[RFC3388] Camarillo, G., Eriksson, G., Holler, J., and H. Schulzrinne, "Grouping of Media Lines in the Session Description Protocol (SDP)", RFC 3388, December 2002.

[RFC3388]Camarillo，G.，Eriksson，G.，Holler，J.，和H.Schulzrinne，“会话描述协议（SDP）中媒体线路的分组”，RFC 3388，2002年12月。

[RFC3398] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Integrated Services Digital Network (ISDN) User Part (ISUP) to Session Initiation Protocol (SIP) Mapping", RFC 3398, December 2002.

[RFC3398]Camarillo，G.，Roach，A.，Peterson，J.，和L.Ong，“综合业务数字网（ISDN）用户部分（ISUP）到会话发起协议（SIP）的映射”，RFC 3398，2002年12月。

[RFC3401] Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part One: The Comprehensive DDDS", RFC 3401, October 2002.

[RFC3401]Mealling，M.“动态委托发现系统（DDDS）第一部分：综合DDDS”，RFC 3401，2002年10月。

[RFC3420] Sparks, R., "Internet Media Type message/sipfrag", RFC 3420, November 2002.

[RFC3420]Sparks，R.，“互联网媒体类型消息/sipfrag”，RFC 3420，2002年11月。

[RFC3427] Mankin, A., Bradner, S., Mahy, R., Willis, D., Ott, J., and B. Rosen, "Change Process for the Session Initiation Protocol (SIP)", BCP 67, RFC 3427, December 2002.

[RFC3427]Mankin，A.，Bradner，S.，Mahy，R.，Willis，D.，Ott，J.，和B.Rosen，“会话启动协议（SIP）的更改过程”，BCP 67，RFC 3427，2002年12月。

[RFC3428] Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C., and D. Gurle, "Session Initiation Protocol (SIP) Extension for Instant Messaging", RFC 3428, December 2002.

[RFC3428]Campbell，B.，Rosenberg，J.，Schulzrinne，H.，Huitema，C.，和D.Gurle，“即时消息的会话启动协议（SIP）扩展”，RFC 34282002年12月。

[RFC3482] Foster, M., McGarry, T., and J. Yu, "Number Portability in the Global Switched Telephone Network (GSTN): An Overview", RFC 3482, February 2003.

[RFC3482]Foster，M.，McGarry，T.，和J.Yu，“全球交换电话网络（GSTN）中的号码可移植性：概述”，RFC 3482，2003年2月。

[RFC3486] Camarillo, G., "Compressing the Session Initiation Protocol (SIP)", RFC 3486, February 2003.

[RFC3486]Camarillo，G.“压缩会话启动协议（SIP）”，RFC 3486，2003年2月。

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

[RFC3515]Sparks，R.，“会话启动协议（SIP）引用方法”，RFC3515，2003年4月。

[RFC3524] Camarillo, G. and A. Monrad, "Mapping of Media Streams to Resource Reservation Flows", RFC 3524, April 2003.

[RFC3524]Camarillo，G.和A.Monrad，“媒体流到资源保留流的映射”，RFC 35242003年4月。

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.

[RFC3550]Schulzrinne，H.，Casner，S.，Frederick，R.，和V.Jacobson，“RTP：实时应用的传输协议”，STD 64，RFC 35502003年7月。

[RFC3578] Camarillo, G., Roach, A., Peterson, J., and L. Ong, "Mapping of Integrated Services Digital Network (ISDN) User Part (ISUP) Overlap Signalling to the Session Initiation Protocol (SIP)", RFC 3578, August 2003.

[RFC3578]Camarillo，G.，Roach，A.，Peterson，J.，和L.Ong，“综合业务数字网（ISDN）用户部分（ISUP）重叠信令到会话发起协议（SIP）的映射”，RFC 3578，2003年8月。

[RFC3581] Rosenberg, J. and H. Schulzrinne, "An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing", RFC 3581, August 2003.

[RFC3581]Rosenberg，J.和H.Schulzrinne，“对称响应路由会话启动协议（SIP）的扩展”，RFC 3581，2003年8月。

[RFC3605] Huitema, C., "Real Time Control Protocol (RTCP) attribute in Session Description Protocol (SDP)", RFC 3605, October 2003.

[RFC3605]Huitema，C.，“会话描述协议（SDP）中的实时控制协议（RTCP）属性”，RFC3605，2003年10月。

[RFC3608] Willis, D. and B. Hoeneisen, "Session Initiation Protocol (SIP) Extension Header Field for Service Route Discovery During Registration", RFC 3608, October 2003.

[RFC3608]Willis，D.和B.Hoeneisen，“注册期间服务路由发现的会话启动协议（SIP）扩展头字段”，RFC 3608，2003年10月。

[RFC3665] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K. Summers, "Session Initiation Protocol (SIP) Basic Call Flow Examples", BCP 75, RFC 3665, December 2003.

[RFC3665]Johnston，A.，Donovan，S.，Sparks，R.，Cunningham，C.，和K.Summers，“会话发起协议（SIP）基本呼叫流示例”，BCP 75，RFC 36652003年12月。

[RFC3666] Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and K. Summers, "Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows", BCP 76, RFC 3666, December 2003.

[RFC3666]Johnston，A.，Donovan，S.，Sparks，R.，Cunningham，C.，和K.Summers，“会话发起协议（SIP）公共交换电话网络（PSTN）呼叫流”，BCP 76，RFC 3666，2003年12月。

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

[RFC3680]Rosenberg，J.，“用于注册的会话启动协议（SIP）事件包”，RFC3680，2004年3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。

[RFC3830] Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K. Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830, August 2004.

[RFC3830]Arkko，J.，Carrara，E.，Lindholm，F.，Naslund，M.，和K.Norrman，“米奇：多媒体互联网键控”，RFC 3830，2004年8月。

[RFC3840] Rosenberg, J., Schulzrinne, H., and P. Kyzivat, "Indicating User Agent Capabilities in the Session

[RFC3840]Rosenberg，J.，Schulzrinne，H.，和P.Kyzivat，“指出会话中的用户代理功能

Initiation Protocol (SIP)", RFC 3840, August 2004.

启动协议（SIP）”，RFC3840，2004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。

[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。

[RFC3853] Peterson, J., "S/MIME Advanced Encryption Standard (AES) Requirement for the Session Initiation Protocol (SIP)", RFC 3853, July 2004.

[RFC3853]Peterson，J.，“会话启动协议（SIP）的S/MIME高级加密标准（AES）要求”，RFC3853，2004年7月。

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

[RFC3856]Rosenberg，J.，“会话启动协议（SIP）的存在事件包”，RFC3856，2004年8月。

[RFC3857] Rosenberg, J., "A Watcher Information Event Template-Package for the Session Initiation Protocol (SIP)", RFC 3857, August 2004.

[RFC3857]Rosenberg，J.，“会话启动协议（SIP）的观察者信息事件模板包”，RFC3857，2004年8月。

[RFC3890] Westerlund, M., "A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)", RFC 3890, September 2004.

[RFC3890]Westerlund，M.“会话描述协议（SDP）的传输无关带宽修改器”，RFC 38902004年9月。

[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”。

[RFC3892] Sparks, R., "The Session Initiation Protocol (SIP) Referred-By Mechanism", RFC 3892, September 2004.

[RFC3892]Sparks，R.，“机制引用的会话启动协议（SIP）”，RFC 38922004年9月。

[RFC3893] Peterson, J., "Session Initiation Protocol (SIP) Authenticated Identity Body (AIB) Format", RFC 3893, September 2004.

[RFC3893]Peterson，J.，“会话启动协议（SIP）认证身份主体（AIB）格式”，RFC 3893，2004年9月。

[RFC3903] Niemi, A., "Session Initiation Protocol (SIP) Extension for Event State Publication", RFC 3903, October 2004.

[RFC3903]Niemi，A.，“事件状态发布的会话启动协议（SIP）扩展”，RFC 3903，2004年10月。

[RFC3910] Gurbani, V., Brusilovsky, A., Faynberg, I., Gato, J., Lu, H., and M. Unmehopa, "The SPIRITS (Services in PSTN requesting Internet Services) Protocol", RFC 3910, October 2004.

[RFC3910]Gurbani，V.，Brusilovsky，A.，Faynberg，I.，Gato，J.，Lu，H.，和M.埃莫霍帕，“精神（PSTN中请求互联网服务的服务）协议”，RFC 39102004年10月。

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

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

October 2004.

2004年10月。

[RFC3959] Camarillo, G., "The Early Session Disposition Type for the Session Initiation Protocol (SIP)", RFC 3959, December 2004.

[RFC3959]Camarillo，G.“会话启动协议（SIP）的早期会话处置类型”，RFC 3959，2004年12月。

[RFC3960] Camarillo, G. and H. Schulzrinne, "Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)", RFC 3960, December 2004.

[RFC3960]Camarillo，G.和H.Schulzrinne，“会话启动协议（SIP）中的早期媒体和铃声生成”，RFC 39602004年12月。

[RFC4028] Donovan, S. and J. Rosenberg, "Session Timers in the Session Initiation Protocol (SIP)", RFC 4028, April 2005.

[RFC4028]Donovan，S.和J.Rosenberg，“会话启动协议（SIP）中的会话计时器”，RFC4028，2005年4月。

[RFC4032] Camarillo, G. and P. Kyzivat, "Update to the Session Initiation Protocol (SIP) Preconditions Framework", RFC 4032, March 2005.

[RFC4032]Camarillo，G.和P.Kyzivat，“会话启动协议（SIP）先决条件框架的更新”，RFC 4032，2005年3月。

[RFC4091] Camarillo, G. and J. Rosenberg, "The Alternative Network Address Types (ANAT) Semantics for the Session Description Protocol (SDP) Grouping Framework", RFC 4091, June 2005.

[RFC4091]Camarillo，G.和J.Rosenberg，“会话描述协议（SDP）分组框架的替代网络地址类型（ANAT）语义”，RFC 4091，2005年6月。

[RFC4117] Camarillo, G., Burger, E., Schulzrinne, H., and A. van Wijk, "Transcoding Services Invocation in the Session Initiation Protocol (SIP) Using Third Party Call Control (3pcc)", RFC 4117, June 2005.

[RFC4117]Camarillo，G.，Burger，E.，Schulzrinne，H.，和A.van Wijk，“使用第三方呼叫控制（3pcc）的会话启动协议（SIP）中的代码转换服务调用”，RFC 41172005年6月。

[RFC4145] Yon, D. and G. Camarillo, "TCP-Based Media Transport in the Session Description Protocol (SDP)", RFC 4145, September 2005.

[RFC4145]Yon，D.和G.Camarillo，“会话描述协议（SDP）中基于TCP的媒体传输”，RFC 41452005年9月。

[RFC4168] Rosenberg, J., Schulzrinne, H., and G. Camarillo, "The Stream Control Transmission Protocol (SCTP) as a Transport for the Session Initiation Protocol (SIP)", RFC 4168, October 2005.

[RFC4168]Rosenberg，J.，Schulzrinne，H.，和G.Camarillo，“作为会话启动协议（SIP）传输的流控制传输协议（SCTP）”，RFC 4168，2005年10月。

[RFC4169] Torvinen, V., Arkko, J., and M. Naslund, "Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA) Version-2", RFC 4169, November 2005.

[RFC4169]Torvinen，V.，Arkko，J.，和M.Naslund，“使用认证和密钥协议（AKA）版本2的超文本传输协议（HTTP）摘要认证”，RFC 4169，2005年11月。

[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。

[RFC4240] Burger, E., Van Dyke, J., and A. Spitzer, "Basic

[RFC4240]E.伯格、J.范·戴克和A.斯皮策，“基础

Network Media Services with SIP", RFC 4240, December 2005.

带有SIP的网络媒体服务”，RFC 4240，2005年12月。

[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。

[RFC4320] Sparks, R., "Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non-INVITE Transaction", RFC 4320, January 2006.

[RFC4320]Sparks，R.，“解决会话启动协议（SIP）非邀请事务已识别问题的措施”，RFC 4320，2006年1月。

[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer Security", RFC 4347, April 2006.

[RFC4347]Rescorla，E.和N.Modadugu，“数据报传输层安全”，RFC 4347，2006年4月。

[RFC4411] Polk, J., "Extending the Session Initiation Protocol (SIP) Reason Header for Preemption Events", RFC 4411, February 2006.

[RFC4411]Polk，J.“扩展抢占事件的会话启动协议（SIP）原因头”，RFC 4411，2006年2月。

[RFC4412] Schulzrinne, H. and J. Polk, "Communications Resource Priority for the Session Initiation Protocol (SIP)", RFC 4412, February 2006.

[RFC4412]Schulzrinne，H.和J.Polk，“会话启动协议（SIP）的通信资源优先级”，RFC 4412，2006年2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。

[RFC4474] Peterson, J. and C. Jennings, "Enhancements for Authenticated Identity Management in the Session Initiation Protocol (SIP)", RFC 4474, August 2006.

[RFC4474]Peterson，J.和C.Jennings，“会话启动协议（SIP）中身份验证管理的增强”，RFC 4474，2006年8月。

[RFC4483] Burger, E., "A Mechanism for Content Indirection in Session Initiation Protocol (SIP) Messages", RFC 4483, May 2006.

[RFC4483]Burger，E.“会话初始化协议（SIP）消息中的内容间接寻址机制”，RFC 4483，2006年5月。

[RFC4488] Levin, O., "Suppression of Session Initiation Protocol (SIP) REFER Method Implicit Subscription", RFC 4488, May 2006.

[RFC4488]Levin，O.“会话启动协议（SIP）的抑制是指方法隐式订阅”，RFC 4488，2006年5月。

[RFC4497] Elwell, J., Derks, F., Mourot, P., and O. Rousseau, "Interworking between the Session Initiation Protocol (SIP) and QSIG", BCP 117, RFC 4497, May 2006.

[RFC4497]Elwell，J.，Derks，F.，Mourot，P.，和O.Rousseau，“会话启动协议（SIP）和QSIG之间的互通”，BCP 117，RFC 4497，2006年5月。

[RFC4508] Levin, O. and A. Johnston, "Conveying Feature Tags with the Session Initiation Protocol (SIP) REFER Method", RFC 4508, May 2006.

[RFC4508]Levin，O.和A.Johnston，“使用会话启动协议（SIP）引用方法传递功能标签”，RFC 4508，2006年5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。

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

[RFC4566]Handley，M.，Jacobson，V.，和C.Perkins，“SDP：会话描述协议”，RFC4566，2006年7月。

[RFC4567] Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E. Carrara, "Key Management Extensions for Session Description Protocol (SDP) and Real Time Streaming Protocol (RTSP)", RFC 4567, July 2006.

[RFC4567]Arkko，J.，Lindholm，F.，Naslund，M.，Norrman，K.，和E.Carrara，“会话描述协议（SDP）和实时流协议（RTSP）的密钥管理扩展”，RFC 4567，2006年7月。

[RFC4568] Andreasen, F., Baugher, M., and D. Wing, "Session Description Protocol (SDP) Security Descriptions for Media Streams", RFC 4568, July 2006.

[RFC4568]Andreasen，F.，Baugher，M.和D.Wing，“媒体流的会话描述协议（SDP）安全描述”，RFC 4568，2006年7月。

[RFC4572] Lennox, J., "Connection-Oriented Media Transport over the Transport Layer Security (TLS) Protocol in the Session Description Protocol (SDP)", RFC 4572, July 2006.

[RFC4572]Lennox，J.，“会话描述协议（SDP）中传输层安全（TLS）协议上的面向连接的媒体传输”，RFC 4572，2006年7月。

[RFC4574] Levin, O. and G. Camarillo, "The Session Description Protocol (SDP) Label Attribute", RFC 4574, August 2006.

[RFC4574]Levin，O.和G.Camarillo，“会话描述协议（SDP）标签属性”，RFC 45742006年8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。

[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。

[RFC4583] Camarillo, G., "Session Description Protocol (SDP) Format for Binary Floor Control Protocol (BFCP) Streams", RFC 4583, November 2006.

[RFC4583]Camarillo，G.“二进制地板控制协议（BFCP）流的会话描述协议（SDP）格式”，RFC4583，2006年11月。

[RFC4662] Roach, A., Campbell, B., and J. Rosenberg, "A Session Initiation Protocol (SIP) Event Notification Extension for Resource Lists", RFC 4662, August 2006.

[RFC4662]Roach，A.，Campbell，B.，和J.Rosenberg，“资源列表的会话启动协议（SIP）事件通知扩展”，RFC 4662，2006年8月。

[RFC4730] Burger, E. and M. Dolly, "A Session Initiation Protocol (SIP) Event Package for Key Press Stimulus (KPML)", RFC 4730, November 2006.

[RFC4730]Burger，E.和M.Dolly，“按键刺激（KPML）的会话启动协议（SIP）事件包”，RFC 4730，2006年11月。

[RFC4733] Schulzrinne, H. and T. Taylor, "RTP Payload for DTMF Digits, Telephony Tones, and Telephony

[RFC4733]Schulzrinne，H.和T.Taylor，“DTMF数字、电话铃声和电话的RTP有效载荷

Signals", RFC 4733, December 2006.

信号”，RFC 47332006年12月。

[RFC4796] Hautakorpi, J. and G. Camarillo, "The Session Description Protocol (SDP) Content Attribute", RFC 4796, February 2007.

[RFC4796]Hautakorpi，J.和G.Camarillo，“会话描述协议（SDP）内容属性”，RFC 47962007年2月。

[RFC4896] Surtees, A., West, M., and A. Roach, "Signaling Compression (SigComp) Corrections and Clarifications", RFC 4896, June 2007.

[RFC4896]Surtees，A.，West，M.和A.Roach，“信号压缩（SigComp）纠正和澄清”，RFC 48962007年6月。

[RFC4916] Elwell, J., "Connected Identity in the Session Initiation Protocol (SIP)", RFC 4916, June 2007.

[RFC4916]Elwell，J.，“会话启动协议（SIP）中的连接身份”，RFC 4916，2007年6月。

[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC 4960, September 2007.

[RFC4960]Stewart，R.，“流控制传输协议”，RFC 49602007年9月。

[RFC5027] Andreasen, F. and D. Wing, "Security Preconditions for Session Description Protocol (SDP) Media Streams", RFC 5027, October 2007.

[RFC5027]Andreasen，F.和D.Wing，“会话描述协议（SDP）媒体流的安全先决条件”，RFC 5027，2007年10月。

[RFC5049] Bormann, C., Liu, Z., Price, R., and G. Camarillo, "Applying Signaling Compression (SigComp) to the Session Initiation Protocol (SIP)", RFC 5049, December 2007.

[RFC5049]Bormann，C.，Liu，Z.，Price，R.，和G.Camarillo，“将信令压缩（SigComp）应用于会话启动协议（SIP）”，RFC 5049，2007年12月。

[RFC5079] Rosenberg, J., "Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)", RFC 5079, December 2007.

[RFC5079]Rosenberg，J.，“拒绝会话启动协议（SIP）中的匿名请求”，RFC 5079，2007年12月。

[RFC5360] Rosenberg, J., Camarillo, G., and D. Willis, "A Framework for Consent-Based Communications in the Session Initiation Protocol (SIP)", RFC 5360, October 2008.

[RFC5360]Rosenberg，J.，Camarillo，G.，和D.Willis，“会话启动协议（SIP）中基于同意的通信框架”，RFC 5360，2008年10月。

[RFC5361] Camarillo, G., "A Document Format for Requesting Consent", RFC 5361, October 2008.

[RFC5361]Camarillo，G.“请求同意的文件格式”，RFC 5361，2008年10月。

[RFC5362] Camarillo, G., "The Session Initiation Protocol (SIP) Pending Additions Event Package", RFC 5362, October 2008.

[RFC5362]Camarillo，G.“会话启动协议（SIP）待添加事件包”，RFC 5362，2008年10月。

[RFC5363] Camarillo, G. and A. Roach, "Framework and Security Considerations for Session Initiation Protocol (SIP) URI-List Services", RFC 5363, October 2008.

[RFC5363]Camarillo，G.和A.Roach，“会话启动协议（SIP）URI列表服务的框架和安全注意事项”，RFC 5363，2008年10月。

[RFC5365] Garcia-Martin, M. and G. Camarillo, "Multiple-Recipient MESSAGE Requests in the Session Initiation Protocol (SIP)", RFC 5365, October 2008.

[RFC5365]Garcia Martin，M.和G.Camarillo，“会话启动协议（SIP）中的多收件人消息请求”，RFC 5365，2008年10月。

[RFC5366] Camarillo, G. and A. Johnston, "Conference Establishment Using Request-Contained Lists in the Session Initiation Protocol (SIP)", RFC 5366, October 2008.

[RFC5366]Camarillo，G.和A.Johnston，“使用会话启动协议（SIP）中包含的请求列表建立会议”，RFC 5366，2008年10月。

[RFC5367] Camarillo, G., Roach, A., and O. Levin, "Subscriptions to Request-Contained Resource Lists in the Session Initiation Protocol (SIP)", RFC 5367, October 2008.

[RFC5367]Camarillo，G.，Roach，A.，和O.Levin，“会话启动协议（SIP）中包含资源列表的请求订阅”，RFC 5367，2008年10月。

[RFC5368] Camarillo, G., Niemi, A., Isomaki, M., Garcia-Martin, M., and H. Khartabil, "Referring to Multiple Resources in the Session Initiation Protocol (SIP)", RFC 5368, October 2008.

[RFC5368]Camarillo，G.，Niemi，A.，Isomaki，M.，Garcia Martin，M.，和H.Khartabil，“会话启动协议（SIP）中的多资源引用”，RFC 5368，2008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。

[RTCP-SUM] Clark, A., Pendleton, A., Johnston, A., and H. Sinnreich, "Session Initiation Protocol Package for Voice Quality Reporting Event", Work in Progress, October 2008.

[RTCP-SUM]Clark，A.，Pendleton，A.，Johnston，A.，和H.Sinnreich，“语音质量报告事件的会话启动协议包”，正在进行的工作，2008年10月。

[SAML] Tschofenig, H., Hodges, J., Peterson, J., Polk, J., and D. Sicker, "SIP SAML Profile and Binding", Work in Progress, November 2008.

[SAML]Tschofenig，H.，Hodges，J.，Peterson，J.，Polk，J.，和D.Sicker，“SIP SAML配置文件和绑定”，正在进行的工作，2008年11月。

[SDP-CAP] Andreasen, F., "SDP Capability Negotiation", Work in Progress, July 2008.

[SDP-CAP]Andreasen，F.，“SDP能力谈判”，正在进行的工作，2008年7月。

[SDP-MEDIA] Gilman, R., Even, R., and F. Andreasen, "SDP media capabilities Negotiation", Work in Progress, July 2008.

[SDP-MEDIA]Gilman，R.，Even，R.，和F.Andreasen，“SDP媒体能力谈判”，进展中的工作，2008年7月。

[SESSION-POLICY] Hilt, V., Camarillo, G., and J. Rosenberg, "A Framework for Session Initiation Protocol (SIP) Session Policies", Work in Progress, November 2008.

[SESSION-POLICY]Hilt，V.，Camarillo，G.，和J.Rosenberg，“会话启动协议（SIP）会话策略框架”，正在进行的工作，2008年11月。

[SIMPLE] Rosenberg, J., "SIMPLE made Simple: An Overview of the IETF Specifications for Instant Messaging and Presence using the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[简单]Rosenberg，J.，“简单变得简单：使用会话启动协议（SIP）的即时消息和状态的IETF规范概述”，正在进行的工作，2008年10月。

[SIPS-URI] Audet, F., "The Use of the SIPS URI Scheme in the Session Initiation Protocol (SIP)", Work in Progress, November 2008.

[SIPS-URI]Audet，F.，“会话启动协议（SIP）中SIPS URI方案的使用”，正在进行的工作，2008年11月。

[SRTP-FRAME] Fischl, J., Tschofenig, H., and E. Rescorla, "Framework for Establishing an SRTP Security Context using DTLS", Work in Progress, October 2008.

[SRTP-FRAME]Fischl，J.，Tschofenig，H.，和E.Rescorla，“使用DTL建立SRTP安全上下文的框架”，正在进行的工作，2008年10月。

[SUBNOT-ETAGS] Niemi, A., "An Extension to Session Initiation Protocol (SIP) Events for Conditional Event Notification", Work in Progress, July 2008.

[SUBNOT-ETAGS]Niemi，A.，“用于条件事件通知的会话启动协议（SIP）事件的扩展”，正在进行的工作，2008年7月。

[TRANSFER-MECH] Garcia, M., Isomaki, M., Camarillo, G., Loreto, S., and P. Kyzivat, "A Session Description Protocol (SDP) Offer/Answer Mechanism to Enable File Transfer", Work in Progress, November 2008.

[TRANSFER-MECH]Garcia，M.，Isomaki，M.，Camarillo，G.，Loreto，S.，和P.Kyzivat，“支持文件传输的会话描述协议（SDP）提供/应答机制”，正在进行的工作，2008年11月。

[UA-PRIVACY] Munakata, M., Schubert, S., and T. Ohba, "UA-Driven Privacy Mechanism for SIP", Work in Progress, October 2008.

[UA-隐私]Munakata，M.，Schubert，S.，和T.Ohba，“SIP的UA驱动隐私机制”，正在进行的工作，2008年10月。

[UPDATE-PAI] Elwell, J., "Updates to Asserted Identity in the Session Initiation Protocol (SIP)", Work in Progress, October 2008.

[UPDATE-PAI]Elwell，J.，“会话启动协议（SIP）中断言身份的更新”，正在进行的工作，2008年10月。

Author's Address

作者地址

Jonathan Rosenberg Cisco Iselin, NJ US

Jonathan Rosenberg Cisco Iselin，美国新泽西州

   EMail: jdrosen@cisco.com
   URI:   http://www.jdrosen.net

   EMail: jdrosen@cisco.com
   URI:   http://www.jdrosen.net

Full Copyright Statement

This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights.

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This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

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