搜索资源列表
astguiclient_1.0.3
- This program was designed as a GUI client for the Asterisk PBX with Digium Zaptel cards and SIP VOIP hard or softphones as extensions, it could be adapted to other functions, but It was designed for Zap/SIP users. The program will run on X and Win32.
Analysis of Skype
- 权威的skype协议分析，详细分析skype的网络结构、防火墙穿越方法、编解码等实现细节。-authority of the agreement, skype detailed analysis of the network structure, through the firewall, such as codecs achieve details.
网络视频电话
- 高性能的可视电话软件 NetTalk效率很高，比我所见的一些可视通话软件要好（如AVPHONE，注：在此并无破坏AVPHONE所在公司形象之意图，仅将软件作客观上的比较） 全图形界面 全图形的界面使得操作方便快捷-high-performance video telephony software NetTalk high efficiency, I have seen some better visualization software calls (such
数字电视SI及其编码
- 关于数字电视mpeg-2业务信息方面的书，非常详细（差不多是规范翻译过来的），通俗易懂。除了印刷效果一般般外，其他的都还满意。值得工程技术人员参考-on digital TV MPEG-2 business information, the book is very detailed (almost a literal translation norms), accessibly. In addition to the general effect as printing, other are
sipXcallLib
- sipXtapi is a comprehensive client library and software development kit (SDK) for SIP-based user agents. It includes SIP signalling support as well as a media framework. A complete and very feature rich softphone can be built easily by adding a graph
zlib123
- The deflation algorithm used by gzip (also zip and zlib) is a variation of LZ77 (Lempel-Ziv 1977, see reference below). It finds duplicated strings in the input data. The second occurrence of a string is replaced by a pointer to the previous
ChatRoom+G
- 这是关于语音聊天软件的一个程序代码 当运行该软的时候，至少需要有一方按下“建立服务器”按钮，表示该台机子用于做服务器，成功时文本显示区会显示“Server Has Been Set OK!”，才有可能接受来自其他机子的连接，当然客户端要输入服务器的IP地址和指定端口后，然后按“连接服务器”按钮，若成功则会在本地文本显示区显示“Client Connection Succeed”，双方建立了连接后，就可以开始通信了。你可以在文本输入区输入你想说的话，按回车，就会在本地文本显示区显示你刚才说的话
AACPLUS
- 语音编码，极好的视听编解码标准，可以用在iptv等领域。-voice coding, excellent audio-visual codec standards, can be used in areas such as IPTV.
opensrc-sipstack-phone
- 开源sip stack以及软终端, 开源sip stack以及软终端,-open source stack and sip soft terminals, as well as open-source stack sip soft terminal, revenue sip stack and soft terminal,
Voip_voice
- voip中用到的一些编码，如711、723等等。-1/1/2006 which used some codes, such as 711,723, and so on.
iad132e
- IAD （Integrated Access Device ）132E(T) 综合接入设备（下文简称IAD132E(T) ）是华为技术有限公司下一代网络NGN（Next Generation Network ）解决方案中的重要部件，用以向公司等用户提供小容量VoIP （Voice over IP）/FoIP（Fax over IP ）解决方案。IAD132E(T) 作为VoIP/FoIP 媒体接入网关，应用于NGN 用户接入层，完成模拟话音信号与IP 包之间的转换，并通过包交换网络传送数据的功能
exosip-v2.0
- extended osip的作为osip的扩展库来使用，实现了SIP的一些扩展功能。此为2.0版本，供大家参考。-extended osip the osip as the extensions for use, Implementation of the SIP, some of the expanded function. For this 2.0 version for reference.
FullDuplexUDPVoiceChat
- 这个为一段语音聊天代码,分为两个客户端和所用控件的代码。-this as a voice chat code, is divided into two client and controls used in the code.
rtrytvbi
- iLBC 产生背景 在VoIP的应用中，大部分厂商采用CELP (Code Excited Linear Prediction) 算法的低速率语音编解码，如ITU G.729和G.723.1等。而VoIP应用主要在包交换的IP网络上进行传输，无法避免IP网络的丢包、延时、抖动等实时传输问题，而传统的这几个CELP算法对高丢包的处理不是很好，因而很大程度上会影响语音通话效果。 -iLBC background in the application of VoIP, Most manu
ooh323c-0.8.3.tar
- Objective Systems Open H.323 for C (ooh323c) is a simple H.323 protocol stack developed in C. The ASN.1 PER messaging code was developed using the ASN1C compiler using a modified version of our core run-time libraries. Additional open source componen
vc_speakfs
- 实现网络在线语音对话.支持GSM编码,ADPCM编码,LPC编码,LPC-10编码 支持RTP,vat协议,并有广播发送,按组进行多点传送,文字交谈等功能-voice network online dialogue. GSM code, ADPCM encoding, LPC coding, LPC-10 encoding support RTP, vat agreement, and this is the broadcasting, according to multicast group,
g.726
- g726算法c源码,应用于语音压缩，如voip应用-algorithm source code for voice compression, as voip application
gsmnew
- gsm算法最新源码，适用于语音压缩，如gsm和voip系统-gsm latest source algorithm applicable to voice compression, such as GSM and voip system
JRTP
- 使用JRTPLIB库实现RTP的传输RTP是目前解决流媒体实时传输问题的最好办法，如果需要在Linux平台上进行实时流媒体编程，可以考虑使用一些开放源代码的RTP库，如LIBRTP、JRTPLIB等。JRTPLIB是一个面向对象的RTP库，它完全遵循RFC 1889设计，在很多场合下是一个非常不错的选择，下面就以JRTPLIB为例，讲述如何在Linux平台上运用RTP协议进行实时流媒体编程。-JRTPLIB use of RTP for achieving the transmission RT
kphone-4.1.1.tar
- 版本V4.1.1,Linux上的开放源代码的IP电话软件SIP协议，主要用于语音通话，但它也支持发送文本的即时消息，即可以像ICQ那样地用文字聊天。 如果网络带宽足够的话，它还支持可视电话 Kphone是GPL的开放源代码的软件。-version 4.1.1, It is IP phone software of Linux open source code based on SIP protocol, it can be used to voice communication, and sur