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NQOS
- VoIP(voice over IP) 就是通过IP 网络承载语音业务,也称IP 网络电话。当网络出现拥塞或传输差错时,语音包就会产生时延、抖动甚至丢失,导致语音不连续或中断,-VoIP (voice over IP) through the IP network to carry voice traffic. also known as IP telephone network. When the network is expected to congestion or transmissio
19854806iLBCfloatpoint
- iLBC codec 语音编码解码器，互联网标准， IETF3951.特别适用于voip的语音编码解码器-codec iLBC voice codec, Internet standards, IETF3951. 1/1/2006 particularly applicable to the voice codec
demo_client_VE_Lite
- voip 语音通话原码，可编译。语音质量很高-1/1/2006 voice calls the original code can be compiled. High quality voice
Echo_Receiver
- Full Duplex voip application source code - 29.8 Kb Download Echo Receiver source code - 14.1 Kb Use these code-programs when debugging and testing Full Duplex application on the same PC, without internet connection. This software receives your packet
full_duplex
- Full Duplex voip application source code - 29.8 Kb Download Echo Receiver source code - 14.1 Kb Use these code-programs when debugging and testing Full Duplex application on the same PC, without internet connection. This software receives your packet
voip
- 包含有多种语音编码的解码器-this is a decoder concluding many audio format
基于exosip的UA结合ORTP实现的VOIP通信
- 基于eXosip实现UA之间通信； 基于ORTP实现RTP语音通信； 基于LINUX声卡采集语音； 基于播放器播放WAV声音文件
pcm2wav.rar
- 将PCM-A转换为WAV的小工具，MingW编译通过，最初是用来处理VoIP RTP流用的,PCM-A will be converted to WAV gadgets, MingW compile is passed, the first is used to handle VoIP RTP streams used
The_Speex_Codec_Manual_Version_1.2_Beta_3
- Speex是一套开源的专门压缩声音的库，压缩的性能非常高，常用在VoIP或者其它网络程序中。Speex声称自己是不受任何专利限制，并授权根据修订后的BSD许可证发布。它可以用来与Ogg容器格式或直接在UDP / RTP协议下传输。 这份是Speex的编码手册英文版，下面的地址是维基百科中关于Speex的介绍: http://en.wikipedia.org/wiki/Speex-Speex is a free software speech codec that may be used on
softphone_client
- 网络语音电话的客户端，同时上传的还有服务端，采用g729进行数据压缩-VoIP clients, while also upload server, using data compression g729
Frq-Diagram-for-standard-Audio-Modules
- This document include frequency response for a standard Audio module. this Frq responce is the best that suite for Music, Communication, VoIP.
lakeofsoft-vcpub-31f17eb
- 语音沟通(VC)组件在设计简化了开发过程的VoIP音频流应用程序,比如对等语音聊天或音频会议。-Voice Communicator (VC) components were designed to simplify the development process of VoIP audio streaming applications, such as peer-to-peer voice chat or audio conference.
p2p
- 语音通话已经是IM的基本功能了，qq，MSN甚至连刚出来的百度HI都自带语音聊天的功能，大家可能觉得很炫，其实大家都是用的windows平台上的API，懂了原理之后自己也可以做，再说了微软也提供了DirectSound的托管互操作程序集，使.net开发人员也很容易的介入到这个领域，甚至你还可以写一个能跑在window mobile上的语音电话，现在好多手机都支持wifi，这样一个简单的wifi电话就由你的手里诞生了。本帖来和大家一起看看如何来做网络电话。-After the voice call
speexdsp-1.2rc3.tar
- 用于voip的开源语音编解码器 Speex ,最新版本源代码-open free voice codec, speex ， newly source verison.
libg7221
- ITU 的G722.1编码器，包含Annex C，是当前复杂度最低的编码器，并且压缩质量非常好，适合高并发voip使用，实测单i5cpu可以支持3000路以上的实时并行处理，采样率16000和32000 码率16000-48000，以800为梯度。-The ITU G722.1 encoder, including Annex C, is currently the least complex of the encoder, and the compression quality is very
chatroom-new
- It is free source for voice chatting room. It is only available on LAN, and can be extended to VoIP.