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Description: 嵌入式IP电话终端的研究与实现，经典论文，包含原理，具体设计。-Embedded IP telephone terminals research and implementation, classic thesis, including the principle, the specific design.
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Size: 3539968 |
Author: 徐自清 |
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Size: 110592 |
Author: chengan |
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Description: JAVA局域网聊天程序,实现简单的局域网内聊天功能。可以查看源码用于初学者学习。也可以在此基础上完善。-chat programme.
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Size: 463872 |
Author: mengchun |
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Description: Skype是创建Kazaa的组织在2003年开发的一个基于Peer-to-Peer(对等网络)的VoIP客户端。它可以几乎无缝的穿越NAT和防火墙，并且语音质量比其他的VoIP客户端软件要好很多。他加密了端到端的通话，分散式存储用户信息，支持即时消息通信和网络语音会议。
-Skype are creating Kazaa organization developed in 2003 based on Peer-to-Peer (peer-to-peer networks) VoIP client. It can be almost seamless across NAT and firewall, and voice quality than other VoIP client software much better. He had end-to-end call encryption, decentralized storage of user information in support of instant messaging and online voice communication session.
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Size: 221184 |
Author: wj |
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Description: 国际电信的G729语音编码的主要程序，主要应用在VOIP中，在VC下已经调试成功，直接可以用来使用-International Telecommunications G729 speech coding of the main program, mainly used in VOIP, in the VC debugger already successful, can be used to direct the use of
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Size: 2081792 |
Author: 胡斌 |
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Description: 使用waveInXXX 底层WINDOWS 语音 API实现录音与播放. waveInXXX可实现流控制,应用于 VOIP 的实时压缩-using waveInXXX API to obtain voice record and play. Steaming is also available,can be used in VoIP data processing
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Size: 36864 |
Author: Jiacong Fang |
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Description: G711 audio codec that can be use for video chat or voip
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Size: 26624 |
Author: easybig |
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Description: asp编写的网络电话。功能完整，实现简单-asp VoIP prepared. Functional integrity, the realization of a simple
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Size: 3627008 |
Author: iy.i |
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Description: 基于GNU的VoIP SIP协议（RFC3261）实现代码-The GNU oSIP library is an implementation of SIP- rfc3261.
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Size: 751616 |
Author: Peter |
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Description: Soft phone for voip !download now.
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Size: 411648 |
Author: Thanh nam |
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Description: jain_sip_sdp.jar,建立会话-jain_sip_sdp.jar, establish a session
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Size: 673792 |
Author: allengong |
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Description: 网络电话源代码，用osip协议栈编程，代码简单对于初学者是很好的工具。-VoIP source code, programmers use osip protocol stack, a simple code for beginners is a very good tool.
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Size: 5002240 |
Author: wj |
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Description: Ekiga (formely known as GnomeMeeting) is an open source VoIP and video conferencing application for GNOME. Ekiga uses both the H.323 and SIP protocols. It supports many audio and video codecs, and is interoperable with other SIP compliant software and also with Microsoft NetMeeting
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Size: 5749760 |
Author: Kukacka |
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Description: 标准sip的ＲＦＣ３２６１文档，是开发基于sip的voip通信方案的必备手册.对其中的重点部分用红色进行了标记-Sip the RFC3261 standard documents, the development of voip in sip-based communications program manual required. The focus of some of them were marked in red
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Size: 254976 |
Author: zgs |
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Description: The function of an SS7 GW is to allow interconnection between the signalling of the PSTN, based on SS7, and the
signalling systems of the VoIP networks – either SIP or H.323. Additionally, the SS7 GW must provide connection
control for interworking the TDM circuits of the PSTN to RTP streams in the IP network.
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Size: 218112 |
Author: jefferychang |
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Description: ppp协议源代码，全部编译通过，在基于PA1688的VOIP的硬件平台实现过，希望对大家有帮助。-ppp protocol source code, all the compiler is passed, the VOIP in the PA1688-based hardware platform, and would like to help everyone.
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Size: 37888 |
Author: 黄学达 |
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Description: 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 VoIP applications and podcasts. Speex claims to be free of any patent restrictions and is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP.
The Speex designers see their project as complementary to the Vorbis general-purpose audio compression project.
Speex is a lossy format, meaning quality is permanently degraded to reduce file size.
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Size: 410624 |
Author: 瞿志超 |
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Description: sipxproxy在基于CS6220的VOIP硬件平台验证通过，希望对大家有帮助-CS6220-based sipxproxy of VOIP hardware platform authentication is passed, we would like to help
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Size: 196608 |
Author: 黄学达 |
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Description: 说明：
1．本文档探讨基于Asterisk如何实现VoIP的一些基本功能，包括
基本呼叫功能的方案选取、主叫号码透传、如何编写Asterisk AGI程序、 Radius认
证计费模块等。
2．本文档VoIP软终端使用X-Lite，其它终端均可以接入测试。
-Note: 1. Asterisk Based on this document to explore how to achieve some of the basic functions of VoIP, including basic call functions of the program selected, Caller ID through Communication, Asterisk AGI procedures on how to prepare, Radius authentication and accounting module. 2. This document the use of VoIP softphone X-Lite, other terminals can access the test.
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Size: 2257920 |
Author: kenken |
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Description: FIle to make auto calls and broadcast the auto converted gsm file to the given number through voip
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Size: 1024 |
Author: dharmesh |
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