independentsoft SIP的来源。

一个linux下的高速sip router-a sip of high-speed router

在ip电话或者视频通讯开发中，常常需要穿越nat，stun是个不错的选择-In the ip phone, or video communication development, often through the nat, stun is a good choice

网络电话。采用GSM语音格式，2.2KB/s的数据流量，支持HTTP, Socks4/5单一代理或二者同时代理。本版本修复了以前版本的一些小问题，增加了动态调整延时，打开、关闭话筒和声音，保存和自动加载通话地址列表，代理服务器列表，使用全部或部分本机地址开机-telephone network. GSM voice format, 2.2KB/s data traffic, supports HTTP, Socks4/5 of a single agent or combination of both agents. This version fixes previous versions of some small problem, the dynamic adjustment of delay, opening, closing and microphone voice, preservation and automatically load calls address list, the proxy list, the use of all or part of the machine addresses boot

H.323视频会议的相关文档参考-H.323 video frequency conference related documents refer

一个音频编码库，支持多种音频编码方法 可以用于VoIP-an audio coding the support of a variety of audio coding method can be used for VoIP

实现局域网内点对点的音频通讯，数据编解码使用G.729,16:1的压缩比，数据传输使用UDP协议。音频模块可在公网内经行通讯-LAN communications within the peer-to-peer audio, data, use codec G.729, 16:1 compression ratio, data transmission using the UDP protocol. Audio module can be in the public network through the communication line

rtp模块， 包括rtp协议报文组包，解析，功能独立，需要自行添加rtp的session管理-rtp module, including the rtp protocol packet group package, analytical, functional independence, the need to add the session management rtp

一个可以进行远程视频的程序，通过ip连接进行视频传输。-One can program the remote video, video transmission via ip connection.

JSIP chat

IETF RFC3261(SIP协议)的中文翻译版；觉得看英文比较慢的朋友可以先看看这个，有少量翻译错误请对照英文版校对-IETF RFC3261 (SIP) version of the Chinese translation; Think of English slower you can take a look at this, a small amount of translation errors in the English version control Please proofread