利用JMF开发的语音聊天客户端，采用UDP传输

这是一个呼叫中心的源码哦,具体内容大家可以-This is a call center source Oh, everybody can be the specific content of

this sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks -this is sip voip source code too. i dont know why it does not work. please check and then send back to me. thanks

这是我们教研室的是师兄课题，已经通过了验证，并且有产品实践过了这是控制端的-This is the Teaching and Research Section of older subjects had been passed validation and products over the practice which is controlled side of the

在开发 VoIP 系统时，主要考虑的一个方面就是连接到模拟电话的接口。设计人员必须了解 PSTN 中存在的电话接口要求，因为它们必须在 VoIP 系统中得到支持。本文将重点介绍连接到标准 POTS 电话的两种最常用接口：Foreign eXchange Subscriber (FXS) 与 Foreign eXchange Office (FXO)。重点说明了设计人员在为 VoIP 小区网关的模拟电话接口提供支持时可能面临的某些挑战。-In the development of VoIP system, one aspect of the main considerations is connected to the analog telephone interface. Designers must understand the existing PSTN telephone interface requirements, because they are in the VoIP system must be supported. This article will focus on the standard POTS telephone connected to the two most commonly used interface: Foreign eXchange Subscriber (FXS) and Foreign eXchange Office (FXO). Focusing on the design staff in the Residential Gateway for VoIP analog telephone interface may provide support to some of the challenges faced.

学习SIP(Session Initiation Protocol)协议的最佳文档。-learning SIP (Session Initiation Protocol) the best documentation.

GSM 06.10 13 kbit/s RPE/LTP语音压缩-GSM 06.1013 kbit/S RPE/LTP voice compression

语音聊天的程序，看看或许有帮助-voice chat procedures, it might be helpful to see

vic语音编解码标准，应用于voip系统-speech codec standard applied voip system

jrtp流媒体传输的code，适合大家参考编程用-jrtp streaming media transmission of code, suitable for everyone with reference programming

Delphi写的IP电话源码-Delphi written source IP telephone