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- Megaco (H.248)
Internet draft: draft-ietf-megaco-merged-00.txt
The Media Gateway Control Protocol, (Megaco) is a result of joint efforts of the IETF and the ITU-T Study Group 16. The protocol definition of this protocol is common text with ITU-T Recommendation H.248.
The Megaco protocol is used between elements of a physically decomposed multimedia gateway. There are no functional differences from a system view between a decomposed gateway, with distributed sub-components potentially on more than one physical device, and a monolithic gateway such as described in H.246. This protocol creates a general framework suitable for gateways, multipoint control units and interactive voice response units (IVRs).
Packet network interfaces may include IP, ATM or possibly others. The interfaces support a variety of SCN signalling systems, including tone signalling, ISDN, ISUP, QSIG and GSM. National variants of these signalling systems are supported where applicable.
All messages are in the format of ASN.1 text messages.
协议分析图：
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- The G.729 codec is an industry standard which allows for placing more calls in limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways. Asterisk currently supports G.729 Annex A only.
Concerning performance, the G.729 codec translations are performed in software, and the overhead should be considered when sizing the server for an Asterisk system. Internal testing with dual Intel® Xeon 1.8GHz processors allowed 60 concurrent G.729 calls. Dual Xeon 2.8GHz processors allowed 80 concurrent G.729 calls.
G.729 Documentation
Installation Guides
03.25.2008 - README (TXT)
Downloads
03.25.2008 - Linux Registration Utillity (Binary)
03.25.2008 - Codecs for Unsupported Platforms (TAR)
03.25.2008 - Complete G729 Directory (HTML)
03.25.2008 - Registration Utility for Unsupported Platforms (Binary)
03.25.2008 - Linux Codec for Current Asterisk Release (TAR)
Miscellaneous
06.09.2006 - Licensing (5 KB - HTML)
Agreements & Policies
10.20.2007 - G.729 Policy (3 KB - HTML)
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- CODEC
Coder/Decoder
A software library that contains the algorithms necessary to convert an analog signal to and from a digital one. See also: Encode, G.711, G.729, GSM
Context
The dialplan is composed of one or more extension contexts. Each extension context is itself simply a collection of extensions. Each extension context in a dialplan has a unique name associated with it. The use of contexts can be used to implement a number of important features, such as security, routing, autoattendant, multilevel menus, authentication, callback, privacy, macros, etc... See also: Dialplan
Dialplan
Detailed definition See also: Context
E&M
Ear & Mouth
A type of signaling commonly used over T1 and E1 interfaces.
Encode
The process of converting an analog signal into a digital signal that can be manipulated easily by a computer. See also: CODEC
FXO
Foreign Exchange Office
A device usually found on the customer end that is powered by the channel and can interface into the telephone company's network. Digium makes FXO modules that interface with PSTN lines using FXS signalling in either Loopstart(fxs_ls) or the more common Kewlstart(fxs_ks) modes. See also: PSTN, REN
FXS
Foreign Exchange Station
A device usually located on the telephony company's property, a FXS device send power through a channel to a phone on the other end. Digium makes FXS modules that interface with PSTN phones using FXO signalling in either Loopstart(fxo_ls) or the more common Kewlstart(fxo_ks) modes. See also: PSTN, REN
G.711
An uncompressed codec that samples a 64kbps channel at 8 bits per sample using pulse code modulation. The Two varients of G.711 are known formally as uLaw and aLaw. See also: CODEC, G.729, GSM
G.729
The G.729 codec is an industry standard which allows for stuffing more calls in limited bandwidth to utilize IP voice in more cost effective ways. A typical call consumes 64Kbps of voice bandwidth. G.729 reduces the call to 8Kbps (normal IP overhead adds to this number). Many people are using Asterisk with G.729 to replace expensive gateways. See also: CODEC, G.711, GSM
GSM
A compressed speech codec that uses a rate of 13 kbps. See also: CODEC, G.711, G.729
H.323
A VOIP protocol that was deployed early and is widely adopted. See also: IAX, MGCP, SIP, VoIP
IAX
Inter-Asterisk eXchange
A VOIP protocol designed to be much more NAT friendly. IAX currently only transports audio. See also: H.323, MGCP, SIP, VoIP
IVR
Interactive Voice Response
An automated voice system that allows callers to navigate a phone system and be directed to the correct extension by pressing a series of numbers on a tuch-tone phone. (I.E. Push 1 for sales, push 2 for support, etc..)
MGCP
Media Gateway Control Protocol
A VOIP Protocol that has both signaling and control and was designed to reduce complexity between media gateways. See also: H.323, IAX, SIP, VoIP
Open source
Detailed definition
PBX
Detailed definition
PRI
Primary Rate Interface
A PRI is a truly digital circuit, containing 24 ISDN channels. One of these channels is a dchannel and used for signaling. The rest are bchannels and used to transport audio.
PSTN
Public Switched Telephone Network
This refers to the analog telephone network that's very common in residential networks. PSTN lines are usually much cheaper then T1 lines, but only provide one channel. Digium's FXO and FXS modules interface with this network. See also: FXO, FXS
REN
Ringer Equivalency Number
A number which represents the ringer loading effect on a line. A ringer equivalency number of 1 represents the loading effect of a single traditional telephone set ringing circuit. Most modern telephones probably will have a REN lower than 1. The total REN expresses the total loading effect of the equipment on the ringing current generator (FXS). The REN of a Digium FXS board is 5 (representing "extension," i.e., parallel-connected telephones). The actual number of devices on the line may be greater than the REN limit, if their respective individual RENs are less than 1. See also: FXO, FXS
SIP
Session Initiation Protocol
An IP protocol used for transporting many forms of media, including video and voice. SIP is currently the most common VOIP protocol. See also: H.323, IAX, MGCP, VoIP
TDM
Time Division Multiplexing
A processes of splitting one medium into two or more channels by using timed segments to transmit information.
Transcode
The process of converting a channel with one type of encoding to a different type of encoding in real time.
VoIP
Voice Over Internet Protocol
A general method for transporting voice through the internet. VOIP commonly refers to the general method and not a specific protocol. SIP is currently the most widely used VOIP protocol. See also: H.323, IAX, MGCP, SIP
Zaptel
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- RFC 2705
IETF MGCP
Media Gateway Control Protocol (MGCP) is used for controlling telephony gateways from external call control elements called media gateway controllers or call agents. A telephony gateway is a network element that provides conversion between the audio signals carried on telephone circuits and data packets carried over the Internet or over other packet networks.
MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by external call control elements. The MGCP assumes that these call control elements, or Call Agents, will synchronize with each other to send coherent commands to the gateways under their control. MGCP is, in essence, a master/slave protocol, where the gateways are expected to execute commands sent by the Call Agents.
The MGCP implements the media gateway control interface as a set of transactions. The transactions are composed of a command and a mandatory response. There are eight types of commands:
MGCP Commands
MGC --> MG CreateConnection: Creates a connection between two endpoints; uses SDP to define the receive capabilities of the paricipating endpoints.
MGC --> MG ModifyConnection: Modifies the properties of a connection; has nearly the same parameters as the CreateConnection command.
MGC <--> MG DeleteConnection: Terminates a connection and collects statistics on the execution of the connection.
MGC --> MG NotificationRequest: Requests the media gateway to send notifications on the occurrence of specified events in an endpoint.
MGC <-- MG Notify: Informs the media gateway controller when observed events occur.
MGC --> MG AuditEndpoint: Determines the status of an endpoint.
MGC --> MG AuditConnection: Retrieves the parameters related to a connection.
MGC <-- MG RestartInProgress: Signals that an endpoint or group of endpoints is take in or out of service.
MGC=Media Gateway Controller
MG=Media Gateway
* CreateConnection.
* ModifyConnection.
* DeleteConnection.
* NotificationRequest.
* Notify.
* AuditEndpoint.
* AuditConnection.
* RestartInProgress.
The first four commands are sent by the Call Agent to a gateway. The Notify command is sent by the gateway to the Call Agent. The gateway may also send a DeleteConnection. The Call Agent may send either of the Audit commands to the gateway. The Gateway may send a RestartInProgress command to the Call Agent.
All commands are composed of a command header, optionally followed by a session description. All responses are composed of a response header, optionally followed by a session description. Headers and session descriptions are encoded as a set of text lines, separated by a carriage return and line feed character (or, optionally, a single line-feed character). The headers are separated from the session description by an empty line.
MGCP uses a transaction identifier to correlate commands and responses. Transaction identifiers have values between 1 and 999999999. An MGCP entity cannot reuse a transaction identifier sooner than 3 minutes after completion of the previous command in which the identifier was used.
The command header is composed of:
* A command line, identifying the requested action or verb, the transaction identifier, the endpoint towards which the action is requested, and the MGCP protocol version,
* A set of parameter lines, composed of a parameter name followed by a parameter value.
The command line is composed of:
* Name of the requested verb.
* Transaction identifier correlates commands and responses. Values may be between 1 and 999999999. An MGCP entity cannot reuse a transaction identifier sooner than 3 minutes after completion of the previous command in which the identifier was used.
* Name of the endpoint that should execute the command (in notifications, the name of the endpoint that is issuing the notification).
* Protocol version.
These four items are encoded as strings of printable ASCII characters, separated by white spaces, i.e., the ASCII space (0x20) or tabulation (0x09) characters. It is recommended to use exactly one ASCII space separator.
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- Voice-over-IP Overview
Voice-over-IP (VoIP) implementations enables users to carry voice traffic (for example, telephone calls and faxes) over an IP network.
There are 3 main causes for the evolution of the Voice over IP market:
* Low cost phone calls
* Add-on services and unified messaging
* Merging of data/voice infrastructures
A VoIP system consists of a number of different components: Gateway/Media Gateway, Gatekeeper, Call agent, Media Gateway Controller, Signaling Gateway and a Call manager
The Gateway converts media provided in one type of network to the format required for another type of network. For example, a Gateway could terminate bearer channels from a switched circuit network (i.e., DS0s) and media streams from a packet network (e.g., RTP streams in an IP network). This gateway may be capable of processing audio, video and T.120 alone or in any combination, and is capable of full duplex media translations. The Gateway may also play audio/video messages and performs other IVR functions, or may perform media conferencing.
In VoIP, the digital signal processor (DSP) segments the voice signal into frames and stores them in voice packets. These voice packets are transported using IP in compliance with one of the specifications for transmitting multimedia (voice, video, fax and data) across a network: H.323 (ITU), MGCP (level 3,Bellcore, Cisco, Nortel), MEGACO/H.GCP (IETF), SIP (IETF), T.38 (ITU), SIGTRAN (IETF), Skinny (Cisco) etc.
Coders are used for efficient bandwidth utilization. Different coding techniques for telephony and voice packet are standardized by the ITU-T in its G-series recommendations: G.723.1, G.729, G.729A etc.
The coder-decoder compression schemes (CODECs) are enabled for both ends of the connection and the conversation proceeds using Real-Time Transport Protocol/User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the protocol stack.
Quality of Service
A number of advanced methods are used to overcome the hostile environment of the IP net and to provide an acceptable Quality of Service. Example of these methods are: delay, jitter, echo, congestion, packet loss, and missordered packets arrival. As VoIP is a delay-sensitive application, a well-engineered, end-to-end network is necessary to use VoIP successfully. The Mean Opinion Score is one of the most important parameters that determine the QoS.
There are several methods and sophisticated algorithms developed to evaluate the QoS: PSQM (ITU P.861), PAMS (BT) and PESQ.Each CODEC provides a certain quality of service. The quality of transmitted speech is a subjective response of the listener (human or artificial means). A common benchmark used to determine the quality of sound produced by specific CODECs is the mean opinion score (MOS). With MOS, a wide range of listeners judge the quality of a voice sample (corresponding to a particular CODEC) on a scale of 1 (bad) to 5 (excellent).
Services
The following are examples of services provided by a Voice over IP network according to market requirements:
Phone to phone, PC to phone, phone to PC, fax to e-mail, e-mail to fax, fax to fax, voice to e-mail, IP Phone, transparent CCS (TCCS), toll free number (1-800), class services, call center applications, VPN, Unified Messaging, Wireless Connectivity, IN Applications using SS7, IP PBX and soft switch implementations.... 0 篇回复 | 参与讨论 | Ray | | | | | | | | |
- VoIP 相关参考资料（Voice Over IP Reference Page）
VoIP在今天变的越来越重要，下面是我们收集的一些资料系统对您有帮助。相关协议介绍会相继更新完善。
信令 Signaling
H.323 H.323
Megaco H.248 Gateway Control Protocol
MGCP Media Gateway Control Protocol
RVP over IP Remote Voice Protocol Over IP Specification
SAPv2 Session Announcement Protocol
SGCP Simple Gateway Control Protocol
SIP Session Initiation Protocol
Skinny Skinny Client Control Protocol (Cisco)
Media
DVB Digital Video Broadcasting
H.261 Video stream for transport using the real-time transport
H.263 Bitstream in the Real-time Transport Protocol
RTCP RTP Control protocol
RTP Real-Time Transport
H.323 Protocols Suite
H.225 Covers narrow-band visual telephone services
H.225 Annex G
H.225E
H.235 Security and authentication
H.323SET
H.245 Negotiates channel usage and capabilities
H.450.1 Series defines Supplementary Services for H.323
H.450.2 Call Transfer supplementary service for H.323
H.450.3 Call diversion supplementary service for H.323
H.450.4 Call Hold supplementary service
H.450.5 Call Park supplementary service
H.450.6 Call Waiting supplementary service
H.450.7 Message Waiting Indication supplementary service
H.450.8 Calling Party Name Presentation supplementary service
H.450.9 Completion of Calls to Busy Subscribers supplementary service
H.450.10 Call Offer supplementary service
H.450.11 Call Intrusion supplementary service
H.450.12 ANF-CMN supplementary service
RAS Manages registration, admission, status
T.38 IP-based fax service maps
T.125 Multipoint Communication Service Protocol (MCS).
SIP 协议 SIP Protocols
MIME
SDP Session Description Protocol
SIP Session Initiation Protocol
VoIP参考资料
VoIP Standards
H.323 Architecture
Converged Network Architecture
SIP Architecture
Fine-Tuning Voice over Packet Services
Voice-over-data network gear ahead of business demand
Voice over ATM
Voice over Frame Relay, IP and ATM
Voice over Frame Relay
Voice over IP Testing: A Practical Guide
H.323 Tutorial
VoIP 文章 VoIP Articles
Voice Over IP: The Battle Heats Up
VoIP in the Enterprise
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- 融合网络体系架构图 converged network architecture
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- 历史回顾
SIP 出现于二十世纪九十年代中期，源于哥伦比亚大学计算机系副教授 Henning Schulzrinne 及其研究小组的研究。Schulzrinne 教授除与人共同提出通过 Internet 传输实时数据的实时传输协议 (RTP) 外，还与人合作编写了实时流传输协议 (RTSP) 标准提案，用于控制音频视频内容在 Web 上的流传输。
Schulzrinne 本来打算编写多方多媒体会话控制 (MMUSIC) 标准。1996 年，他向 IETF 提交了一个草案，其中包含了 SIP 的重要内容。1999 年，Shulzrinne 在提交的新标准中删除了有关媒体内容方面的无关内容。随后，IETF 发布了第一个 SIP 规范，即 RFC 2543。虽然一些供应商表示了担忧，认为 H.323 和 MGCP 协议可能会大大危及他们在 SIP 服务方面的投资，IETF 继续进行这项工作，于 2001 年发布了 SIP 规范 RFC 3261。
RFC 3261 的发布标志着 SIP 的基础已经确立。从那时起，已发布了几个 RFC 增补版本，充实了安全性和身份验证等领域的内容。例如，RFC 3262 对临时响应的可靠性作了规定。RFC 3263 确立了 SIP 代理服务器的定位规则。RFC 3264 提供了提议/应答模型，RFC 3265 确定了具体的事件通知。
早在 2001 年，供应商就已开始推出基于 SIP 的服务。今天，人们对该协议的热情不断高涨。Sun Microsystems 的 Java Community Process 等组织正在使用通用的 Java 编程语言定义应用编程接口 (API)，以便开发商能够为服务提供商和企业构建 SIP 组件和应用程序。最重要的是，越来越多的竞争者正在借助前途光明的新服务进入 SIP 市场。SIP 正在成为自 HTTP 和 SMTP 以来最为重要的协议之一。
通信提供商及其合作伙伴和用户越来越渴求新一代基于 IP 的服务。现在有了 SIP（会话启动协议），一解燃眉之急。SIP 是不到十年前在计算机科学实验室诞生的一个想法。它是第一个适合各种媒体内容而实现多用户会话的协议，现在已成了 Internet 工程任务组 (IETF) 的规范。
今天，越来越多的运营商、CLEC（竞争本地运营商）和 ITSP（IP 电话服务商）都在提供基于 SIP 的服务，如市话和长途电话技术、在线信息和即时消息、IP Centrex/Hosted PBX、语音短信、push-to-talk（按键通话）、多媒体会议等等。独立软件供应商 (ISV) 正在开发新的开发工具，用来为运营商网络构建基于 SIP 的应用程序以及 SIP 软件。网络设备供应商 (NEV) 正在开发支持 SIP 信令和服务的硬件。现在，有众多 IP 电话、用户代理、网络代理服务器、VOIP 网关、媒体服务器和应用服务器都在使用 SIP。
SIP 从类似的权威协议－－如 Web 超文本传输协议 (HTTP) 格式化协议以及简单邮件传输协议 (SMTP) 电子邮件协议－－演变而来并且发展成为一个功能强大的新标准。但是，尽管 SIP 使用自己独特的用户代理和服务器，它并非自成一体地封闭工作。SIP 支持提供融合的多媒体服务，与众多负责身份验证、位置信息、语音质量等的现有协议协同工作。
本白皮书对 SIP 及其作用进行了概括性的介绍。它还介绍了 SIP 从实验室开发到面向市场的过程。本白皮书说明 SIP 提供哪些服务以及正在实施哪些促进发展的方案。它还详细介绍了 SIP 与各种协议不同的重要特点并说明如何建立 SIP 会话。
新一代的服务
SIP 较为灵活，可扩展，而且是开放的。它激发了 Internet 以及固定和移动 IP 网络推出新一代服务的威力。SIP 能够在多台 PC 和电话上完成网络消息，模拟 Internet 建立会话。
与存在已久的国际电信联盟 (ITU) SS7 标准（用于呼叫建立）和 ITU H.323 视频协议组合标准不同，SIP 独立工作于底层网络传输协议和媒体。它规定一个或多个参与方的终端设备如何能够建立、修改和中断连接，而不论是语音、视频、数据或基于 Web 的内容。
SIP 大大优于现有的一些协议，如将 PSTN 音频信号转换为 IP 数据包的媒体网关控制协议 (MGCP)。因为 MGCP 是封闭的纯语音标准，所以通过信令功能对其进行增强比较复杂，有时会导致消息被破坏或丢弃，从而妨碍提供商增加新的服务。而使用 SIP，编程人员可以在不影响连接的情况下在消息中增加少量新信息。
例如，SIP 服务提供商可以建立包含语音、视频和聊天内容的全新媒体。如果使用 MGCP、H.323 或 SS7 标准，则提供商必须等待可以支持这种新媒体的协议新版本。而如果使用 SIP，尽管网关和设备可能无法识别该媒体，但在两个大陆上设有分支机构的公司可以实现媒体传输。
而且，因为 SIP 的消息构建方式类似于 HTTP，开发人员能够更加方便便捷地使用通用的编程语言（如 Java）来创建应用程序。对于等待了数年希望使用 SS7 和高级智能网络 (AIN) 部署呼叫等待、主叫号码识别以及其他服务的运营商，现在如果使用 SIP，只需数月时间即可实现高级通信服务的部署。
这种可扩展性已经在越来越多基于 SIP 的服务中取得重大成功。Vonage 是针对用户和小企业用户的服务提供商。它使用 SIP 向用户提供 20,000 多条数字市话、长话及语音邮件线路。Deltathree 为服务提供商提供 Internet 电话技术产品、服务和基础设施。它提供了基于 SIP 的 PC 至电话解决方案，使 PC 用户能够呼叫全球任何一部电话。Denwa Communications 在全球范围内批发语音服务。它使用 SIP 提供 PC 至 PC 及电话至 PC 的主叫号码识别、语音邮件，以及电话会议、统一通信、客户管理、自配置和基于 Web 的个性化服务。
某些权威人士预计，SIP 与 IP 的关系将发展成为类似 SMTP 和 HTTP 与 Internet 的关系，但也有人说它可能标志着 AIN 的终结。迄今为止，3G 界已经选择 SIP 作为下一代移动网络的会话控制机制。Microsoft 已经选择 SIP 作为其实时通信策略并在 Microsoft XP、Pocket PC 和 MSN Messenger 中进行了部署。Microsoft 同时宣布 CE.net 的下一个版本将使用基于 SIP 的 VoIP 应用接口层，并承诺向用户 PC 提供基于 SIP 的语音和视频呼叫。
另外，MCI 正在使用 SIP 向 IP 通信用户部署高级电话技术服务。用户将能够通知主叫方自己是否有空以及首选的通信方式，如电子邮件、电话或即时消息。利用在线信息，用户还能够即时建立聊天会话和召开音频会议。使用 SIP 将不断地实现各种功能。
SIP 的优点：类似 Web 的可扩展开放通信
使用 SIP，服务提供商可以随意选择标准组件，快速驾驭新技术。不论媒体内容和参与方数量，用户都可以查找和联系对方。SIP 对会话进行协商，以便所有参与方都能够就会话功能达成一致以及进行修改。它甚至可以添加、删除或转移用户。
不过，SIP不是万能的。它既不是会话描述协议，也不提供会议控制功能。为了描述消息内容的负载情况和特点，SIP 使用 Internet 的会话描述协议 (SDP) 来描述终端设备的特点。SIP 自身也不提供服务质量 (QoS)，它与负责语音质量的资源保留设置协议 (RSVP) 互操作。它还与若干个其他协议进行协作，包括负责定位的轻型目录访问协议 (LDAP)、负责身份验证的远程身份验证拨入用户服务 (RADIUS) 以及负责实时传输的 RTP 等多个协议。
SIP 要求
SIP 规定了以下基本的通信要求：
1. 用户定位服务
2. 会话建立
3. 会话参与方管理
4. 特点的有限确定
SIP 的一个重要特点是它不定义要建立的会话的类型，而只定义应该如何管理会话。有了这种灵活性，也就意味着 SIP 可以用于众多应用和服务中，包括交互式游戏、音乐和视频点播以及语音、视频和 Web 会议。
下面是 SIP 在新的信令协议中出类拔萃的一些其他特点
SIP 消息是基于文本的，因而易于读取和调试。新服务的编程更加简单，对于设计人员而言更加直观。
SIP 如同电子邮件客户机一样重用 MIME 类型描述，因此与会话相关的应用程序可以自动启动。
SIP 重用几个现有的比较成熟的 Internet 服务和协议，如 DNS、RTP、RSVP 等。不必再引入新服务对 SIP 基础设施提供支持，因为该基础设施很多部分已经到位或现成可用。
对 SIP 的扩充易于定义，可由服务提供商在新的应用中添加，不会损坏网络。网络中基于 SIP 的旧设备不会妨碍基于 SIP 的新服务。例如，如果旧 SIP 实施不支持新的 SIP 应用所用的方法/标头，则会将其忽略。
SIP 独立于传输层。因此，底层传输可以是采用 ATM 的 IP。SIP 使用用户数据报协议 (UDP) 以及传输控制协议 (TCP)，将独立于底层基础设施的用户灵活地连接起来。
SIP 支持多设备功能调整和协商。如果服务或会话启动了视频和语音，则仍然可以将语音传输到不支持视频的设备，也可以使用其他设备功能，如单向视频流传输功能。
SIP 会话构成
SIP 会话使用多达四个主要组件：SIP 用户代理、SIP 注册服务器、SIP 代理服务器和 SIP 重定向服务器。这些系统通过传输包括了 SDP 协议（用于定义消息的内容和特点）的消息来完成 SIP 会话。下面概括性地介绍各个 SIP 组件及其在此过程中的作用。
SIP 用户代理 (UA) 是终端用户设备，如用于创建和管理 SIP 会话的移动电话、多媒体手持设备、PC、PDA 等。用户代理客户机发出消息。用户代理服务器对消息进行响应。
SIP 注册服务器是包含域中所有用户代理的位置的数据库。在 SIP 通信中，这些服务器会检索参与方的 IP 地址和其他相关信息，并将其发送到 SIP 代理服务器。
SIP 代理服务器接受 SIP UA 的会话请求并查询 SIP 注册服务器，获取收件方 UA 的地址信息。然后，它将会话邀请信息直接转发给收件方 UA（如果它位于同一域中）或代理服务器（如果 UA 位于另一域中）。
SIP 重定向服务器允许 SIP 代理服务器将 SIP 会话邀请信息定向到外部域。SIP 重定向服务器可以与 SIP 注册服务器和 SIP 代理服务器同在一个硬件上。
以下几个情景说明 SIP 组件之间如何进行协调以在同一域和不同域中的 UA 之间建立 SIP 会话：
在同一域中建立 SIP 会话
下图说明了在预订同一个 ISP 从而使用同一域的两个用户之间建立 SIP 会话的过程。用户 A 使用 SIP 电话。用户 B 有一台 PC，运行支持语音和视频的软客户程序。加电后，两个用户都在 ISP 网络中的 SIP 代理服务器上注册了他们的空闲情况和 IP 地址。用户 A 发起此呼叫，告诉 SIP 代理服务器要联系用户 B。然后，SIP 代理服务器向 SIP 注册服务器发出请求，要求提供用户 B 的 IP 地址，并收到用户 B 的 IP 地址。SIP 代理服务器转发用户 A 与用户 B 进行通信的邀请信息（使用 SDP），包括用户 A 要使用的媒体。用户 B 通知 SIP 代理服务器可以接受用户 A 的邀请，且已做好接收消息的准备。SIP 代理服务器将此消息传达给用户 A，从而建立 SIP 会话。然后，用户创建一个点到点 RTP 连接，实现用户间的交互通信。
1.呼叫用户 B
2.查询捻没?B 在哪里？?br> 3.响应捻没?B 的 SIP 地址?br> 4.挚顶呼叫
5. 响应
6. 响应
7. 多媒体通道已建立
在不同的域中建立 SIP 会话
本情景与第一种情景的不同之处如下。用户 A 邀请正在使用多媒体手持设备的用户 B 进行 SIP 会话时，域 A 中的 SIP 代理服务器辨别出用户 B 不在同一域中。然后，SIP 代理服务器在 SIP 重定向服务器上查询用户 B 的 IP 地址。SIP 重定向服务器既可在域 A 中，也可在域 B 中，也可既在域 A 中又在域 B 中。SIP 重定向服务器将用户 B 的联系信息反馈给 SIP 代理服务器，该服务器再将 SIP 会话邀请信息转发给域 B 中的 SIP 代理服务器。域 B 中的 SIP 代理服务器将用户 A 的邀请信息发送给用户 B。用户 B 再沿邀请信息经由的同一路径转发接受邀请的信息。
1. 呼叫用户 B 2. 询问撑胰绾谓油ㄓ?B 中的用户 B？? 3. 响应挚砜刂破鞯挠虻刂窋 4. 挚顶呼叫域 B 的 SIP 代理 5. 查询捻没?B 在哪里？? 6. 用户 B 的地址 7. 代理呼叫 8. 响应 9. 响应 10.响应 11.多媒体通道已建立
无缝、灵活、可扩展：展望 SIP 未来
SIP 能够连接使用任何 IP 网络（有线 LAN 和 WAN、公共 Internet 骨干网、移动 2.5G、3G 和 Wi-Fi）和任何 IP 设备（电话、PC、PDA、移动手持设备）的用户，从而出现了众多利润丰厚的新商机，改进了企业和用户的通信方式。基于 SIP 的应用（如 VOIP、多媒体会议、push-to-talk（按键通话）、定位服务、在线信息和 IM）即使单独使用，也会为服务提供商、ISV、网络设备供应商和开发商提供许多新的商机。不过，SIP 的根本价值在于它能够将这些功能组合起来，形成各种更大规模的无缝通信服务。
使用 SIP，服务提供商及其合作伙伴可以定制和提供基于 SIP 的组合服务，使用户可以在单个通信会话中使用会议、Web 控制、在线信息、IM 等服务。实际上，服务提供商可以创建一个满足多个最终用户需求的灵活应用程序组合，而不是安装和支持依赖于终端设备有限特定功能或类型的单一分散的应用程序。
通过在单一、开放的标准 SIP 应用架构下合并基于 IP 的通信服务，服务提供商可以大大降低为用户设计和部署基于 IP 的新的创新性托管服务的成本。它是 SIP 可扩展性促进本行业和市场发展的强大动力，是我们所有人的希望所在。... 0 篇回复 | 参与讨论 | Ray | | | | | | | | |
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