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共有 9 个源码项目.
1. Cell Phone SMS AIM
- Cell Phone SMS AIM uses the Net::AIM Perl module, signs on a screen name, and routes messages to your cell phone via SMS. It requires lynx. It also supports pagers.
Cell Phone SMS AIM使用Net::AIM Perl模块，在屏幕上签名，并通过SMS把消息发送到你的手机。它需要lynx。它也支持页面。 |
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2. Robust Audio Tool
- RAT is an RTP audio conferencing and streaming application that allows users to particpate in point-to-point and multi-point audio conferences over the Internet. Developing features include multiple sampling rates, mono/stereo, loss concealment, and IPv4/IPv6 support, and both versions feature encryption and run on a range of platforms.
来源：http://freshmeat.net/projects/rat/?topic_id=27%2C40%2C247%2C113
RAT是一个RTP 音频和流媒体软件，允许用户参与到点对点的或多对一的的论坛中。正在开发的功能包括多采样率，单/立体声，支持IPv4/IPv6，所有版本都包括编码功能，支持多平台。 |
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3. esms
- esms is a simple program which sends SMSes to cellular phones. It splits big messages into many fragments, which are later sent with error control. It works for cellular phones in Spain.
esms是一种用来传送SMSes给细胞电话的简单程序。它把大的信息分裂成许多碎片，稍后用差错控制来传送。它用西班牙语为细胞电话工作。 |
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4. VoIM - Jabber Text and Voice IM Client
- VoIM is a Java based Jabber client that will provide excellent text instant messaging capabilities as well as the ability to send and receive voice messages.
VoIM是一个基于Java的 Jabber客户端，除了能够发送和接收语音消息外，还可以提供优秀的文本即时消息功能。
来源: http://sourceforge.net/projects/voim/ |
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5. Cornfed SIP User Agent
- About:
The Cornfed SIP User Agent is a Session Initiation Protocol (SIP)-based Softphone for your x86 Linux systems. It allows you to make Internet phone calls using an Advanced Linux Sound Architecture (ALSA) or Open Sound System (OSS) sound card with speakers and microphone as your telephone handset.
土气的SIP使用代理是基于会议开放协议（SIP）适用于x86 Linux系统的电话软件，它允许使用者通过先进的Linux声音系统（ALSA)或者开放的声音系统(OSS)拨打国际互联网电话，使用麦克风作为电话听筒。
来源：http://freshmeat.net/projects/cornfedsipua/?branch_id=48563&release_id=172156 |
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6. sfront
- Sfront compiles MPEG 4 Structured Audio (MP4-SA) bitstreams into efficient C programs that generate audio when executed. MP4-SA is a standard for normative algorithmic sound, that combines an audio signal processing language (SAOL) with score languages (SASL, and the legacy MIDI File Format). Under Linux and Mac OS X, sfront supports real-time, low-latency audio input/output, local MIDI input from soundcards, and networked MIDI input using RTP and SIP. A SIP server hosted on the Berkeley campus manages sessions. The documentation includes a book about SAOL programming.
sfront将MPEG 4结构的音频(MP4-SA)位流编译成有效的C程序，当它被执行时就可以生成音频。MP4-SA 是标准化算法的声音处理标准，它将一个音频信号处理语言（SAOL）和源语言（SASL, 和传统的MIDI文件格式)组合起来。在Linux 和Mac OS X环境下，sfront 支持实时、低等待时间的音频输入/输出，从声卡的本地MIDI输入，以及使用RTP和SIP的网络MIDI输入。一个位于Berkeley校园的SIP服务器负责管理会话。文档包括一本关于SAOL编程的书。 |
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7. AsteriskPNX IP PBX
- Open Source PNX is a full-featured fusion TDM and H.323 IP PBX supporting a wide variety of telephony interfaces and providing a simple API for developing customer IVR applications, CRM applications, telephony drivers, etc.
开源PNX 是一个功能完整的TDM 和H.323 IP PBX，支持广泛的技术接口，提供了一个简单的API来开发定制的IVR 应用程序，CRM应用程序，电话驱动程序，等等。
来源: http://sourceforge.net/projects/pnx/ |
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8. sipsak
- sipsak is a command line tool for performing various tests on Session Initiation Protocol (SIP) applications and devices. It can make 4 different tests, send the contents of a file, and interpret and react on the responses.
sipsak是一个命令行工具，用于在Session Initiation Protocol (SIP)应用程序和设备上执行多种测试。它能做四种不同的测试，发送文件内容，并解释和作用于响应。 |
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9. Skype
- About:
Skype uses P2P (peer-to-peer) technology to provide voice-based communication with other Internet users. The technology is extremely advanced, but easy to use. It features excellent sound quality, end-to-end encryption, and automatic negotiation of firewalls or routers.
Skype使用P2P技术来提供基于声音与其他网络用户之间的交流。该技术极端先进，但是容易使用。它具有非常杰出的声音质量，端对端的加密，以及防火墙和路由器的自动流通。
来源：http://freshmeat.net/projects/skype/?branch_id=51248&release_id=173676 |
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