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基于SIP的VoIP相关协议、协议栈及应用
lionwq | 2008-01-08 15:59:40    阅读:12279   发布文章

基于SIP的VoIP相关协议、协议栈及应用

 相关协议

SIP(Session Initiation Protocol)协议是IETF多媒体数据和控制体系结构的一个组成部分,是一个基于文本的协议。

SIP协议的出发点是想借鉴Web业务成功的经验,以现有的Internet为基础来构架IP电话业务网,因此SIP有着与H.323完全不同的设计思想。它是一个分散式的协议,它将网络设备的复杂性向网络边缘推,使核心网络仍是一个“Best Effort″的传送通道,这就是SIP系统中核心网络服务器可以不保留状态(stateless)的原因(SIP消息本身含有一个呼叫的所有信息)。因为核心网络服务器需要处理大量的呼叫,不保留每一呼叫的状态,将大大提高系统的处理能力,为组建大规模的IP电话业务网奠定了基础,而边缘网络服务器可以是有状态的(stateful)。这种stateless和stateful结合的模式既可以充分发挥SIP的特点(如用户定位和查找)又保留了Internet无法连接数据传送的设计思路。相对于H.323而言,SIP协议更简单,易于实现,易于扩展以支持智能用户设备和实现一些高级功能,并支持终端的移动性,3GPP已经要求3G的终端设备需要支持SIP协议,因此尽管现在市场上还是H.323的设备占据相当大的份额,但SIP设备前景看好。

SIP主要相关协议有:

RFC3261(2000年发布版本): SIP: Session Initiation Protocol

RFC2543(99年发布版本): SIP: Session Initiation Protocol

SDP: SDP: Session Description Protocol

用于描述呼叫双方媒体信息的协议。

在SIP应用中的Invite方法及其200ok回应以及Options方法中用来描述

RTP负载类型、地址及端口信息

 

相关协议有:

RTP/RTCP协议:

RFC3550(2003年发布,现为标准草案)

RTP: A Transport Protocol for Real-Time Applications
RFC1889(1996年发布)

RTP: A Transport Protocol for Real-Time Applications

RFC1890:

RTP Profile for Audio and Video Conferences with Minimal Control

RFC2833: 描述电话相关的一些信号在RTP上的传输方法

RTP Payload for DTMF Digits, Telephony Tones

and Telephony Signals

RFC3389: 描述在不支持舒适噪音(CN)生成的编码方式上如何传输舒适噪音

Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
etc.

 

RTSP协议: Real Time Streaming Protocol (RTSP)

SAP协议: Session Announcement Protocol
 

STUN协议: 用于查询NAT类型的轻量级协议

STUN - Simple Traversal of User Datagram Protocol (UDP)

Through Network Address Translators (NATs)

RFC3235:

Network Address Translator (NAT)-Friendly Application Design Guidelines

 

其它协议见ietf官方主页,可使用IETF rfc搜索引擎进行查询

目前China-pubRFC文档中文翻译计划已将部分RFC文档翻译为中文。


 

协议栈

SIP协议栈:
  • dissipate: C++; 许可: GPL

主页:http://www.div8.net/dissipate/

  • dissipate2: C++; 许可: GPL

主页: http://www.wirlab.net/kphone/

前者的更新版本,是kphone的一部分

  • GNU osip; C; 许可: LGPL

主页: http://www.gnu.org/software/osip/

即libosip,现在版本已经升为libosip2

  • GNU eXosip; C; License: GPL

主页: http://savannah.nongnu.org/projects/exosip/

在libosip上提供了一个UA层以控制SIP的呼叫建立及相关的扩展功能

  • SIP from vovida.org: C++; License: Vovida Software License

主页: http://www.vovida.org/protocols/downloads/sip/

 

 

RTP协议栈(开放源代码)
  • Common Multimedia Library:来自UCL London; C; 许可: Free

    主页:http://www-mice.cs.ucl.ac.uk/multimedia/software/common/

  • jrtplib: C++; 许可: Free

主页: http://lumumba.luc.ac.be/jori/jrtplib/jrtplib.html

  • ortp: C; 许可: LGPL

主页: http://www.linphone.org/ortp/; 无RTCP, 是linphone的一部分

  • GNU ccRTP: C++; 许可: GPL (with linking exception)

主页: http://www.gnu.org/software/ccrtp/

  • LIVE.COM Streaming Media: C++; 许可: LGPL

主页: http://live.com/liveMedia/

  • Morgan RTP DirectShow Filters: C++

主页: http://www.morgan-multimedia.com/RTP/; based on liveMedia library

  • RTP from vovida.org: C++; 许可: Vovida Software License

主页: http://www.vovida.org/protocols/downloads/rtp/

  • RTPlib: RTP library from Lucent Technologies/Cloumbia University; C

许可: Non-exklusive source code license;

主页: http://www-out.bell-labs.com/project/RTPlib/

  • librtp: C; 许可: GPL

主页: http://gphone.sourceforge.net/template.php3?page=librtp;

源自 Gnome-o-phone

 

应用

SIP Phone (User Agent, Softphone, Proxy)
  • Ubiquity User Agent: Java based SIP Client for Windows, very useful, you have to register (free) to get an license

主页: http://www.ubiquity.net/useragent.php

  • Linphone: A SIP Softphone for Linux (GNOME), needs libosip and oRTP

主页: http://www.linphone.org/

  • KPhone: A SIP Softphone for Linux (KDE)

主页: http://www.wirlab.net/kphone/index.html

  • Vovida: Complete SIP Suite for Linux (Uaser Agent, Proxy, ...), very, very big software

主页: Vovida.org

  • Siphon: Linux SIP Softphone

主页: http://siphon.sourceforge.net/index.html

  • AVAZ SIP Phone: Cool looking SIP Phone for Windows, crashes very often at my PC, but works well on my friends PC

主页:http://www.avaz.com/products/software/sip/index.html

  • EZ-Phone (Evaluation Version): SIP Phone for Windows

主页: http://www.hssworld.com/voip/download.htm

  • MySIP: SIP User Agent from Siemens

主页: http://www.mysip.ch/

  • MSN Messenger: Microsofts Messenger, Version 4.6 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of iptel.org.

主页: http://messenger.microsoft.com/; local download of Version 4.6 for Windows NT (2000).

  • MSN Messenger: Microsofts Messenger, Version 4.7 allows also connections to other SIP servers than microsofts one. Nice design, works very well. Can be used with the SIP service of iptel.org.

主页: http://messenger.microsoft.com/; local download of Version 4.7 for Windows XP.

  • SJPhone: SIP and H.323 Softphone for Windows, Linux and PocketPC from: http://www.sjlabs.com/products.html.

The configuration for SIP is a little bit tweaky. And there must not be another SIP client running on port 5060 or the SJPhone won't work.

  • instant xpressa: The software version of the xpressa SIP phone from pingtel: http://www.pingtel.com/appdev.jsp;

A really impressive SIP phone with a lot of features. Can be extended with Java programs, but no free version available.

  • xphone: A SIP client for Windows and Windows CE, http://xphone.xten.net/.

The beta version is free.

  • SIPPS: SIP softphone with answering machine and a lot of features, but, IMHO, a not very intuitive user interface, which should be better in the next version (try it yourself).http://www.sippstar.com/en/index.html.

A Demo for testing is available.

 

SIP 应用 (Proxy, Location Server)
  • Sip Express Router (ser): Highspeed GNU SIP proxy with a lot of features and a lot of ongoing development.

主页: http://www.iptel.org/ser/,开发主页在:development homepage.

  • Asterisk: Linux Software PBX with Gateway, SIP Proxy, Gateway (SIP, H.323, PSTN, ...)

主页: http://www.asteriskpbx.com/

 

 

SIP测试
  • sipsak: SIP Swiss Army Knife, very useful test utility (Linux)

主页: http://sipsak.berlios.de/

  • SIPNess: Ortena Networks SIP Messenger, very useful test utility for windows;

主页: http://www.ortena.com/download.htm

 

 

RTP应用
  • RAT - Robust Audio Tool; Supports a large number of codecs, ... 许可: Free

主页: http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/

  • JMF - Java Media Framework: Can receive and send RTP streams

主页: http://java.sun.com/products/java-media/jmf/

  • MP3/RTP Plugin for Winamp:

主页: http://www.live.com/multikit/winamp-plugin.html

  • Vomit - Voice over Missconfigured Internet Telephones: Plays back captured voice conversation

主页: http://vomit.xtdnet.nl/

  • RTP Tools: Several RTP utilities from the Columbia University

主页: http://www.cs.columbia.edu/IRT/software/rtptools/

  • UDP Packet Reflector/Forwarder: A tiny tool which forwards or reflects UDP packets. You can also add delay and packet loss. Very useful if you want to test RTP applications.

主页: http://www.cs.ucl.ac.uk/staff/s.bhatti/teaching/z02/reflector.html. As I was not able to compile this tool I searched and found a binary somewhere in the web. 也可在镜像下载

 

NAT相关

Vocal1.5 from vovida.org增加了对NAT的部分支持,但还需做些修改,并提供了STUN测试工具,可在vovida.org上下载

Ridgeway公司提供VoIP穿越NAT的解决方案,可以去Ridgewaysystems主页看看,他们在国内也有代理

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