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[Linux-UnixX-Lite_Install.tar

Description: SIP软交换电话终端用于sip服务通信,VOIP电话,免费开源-SIP Soft Switch Telephone Terminal sip services for communications, VoIP phones, free and open source
Platform: | Size: 2094582 | Author: sa | Hits:

[Otheross1_2_4_1

Description: sip服务器,能支持一般的sip客户端.ondosip是一款非常好的服务器,可以免费使用-sip server, will support the general sip client. ondosip section is a very good server can be free
Platform: | Size: 10301809 | Author: zippo | Hits:

[JSP/JavaMyMessenger

Description: * This a simple tool to send/receive UDP packet based on a * free software developed by Neil Deason. * The purpose of program is to simulate a remote MGC So that the * SIP Message debugging/testing is possible.-* This a simple tool to send / receive UDP p acket * based on a free software developed by Nei l Deason. * The purpose of program is to simulate So a remote MGC that the SIP Message debugging * / testing is possible.
Platform: | Size: 11133 | Author: adonmiao | Hits:

[VOIP programlibeXosip2-1.9.1-pre16.tar

Description: 本软件是关于exosip开发库源码的 ,用 C 编写而成 ,在 LINUX WINDOWS 平台下可以编译运行 自由软件。-on exosip source development libraries, C prepared in LINUX Windows platform can run free software compiler.
Platform: | Size: 475136 | Author: 凌庆华 | Hits:

[JSP/JavaMyMessenger

Description: * This a simple tool to send/receive UDP packet based on a * free software developed by Neil Deason. * The purpose of program is to simulate a remote MGC So that the * SIP Message debugging/testing is possible.-* This a simple tool to send/receive UDP p acket* based on a free software developed by Nei l Deason.* The purpose of program is to simulate So a remote MGC that the SIP Message debugging*/testing is possible.
Platform: | Size: 11264 | Author: adonmiao | Hits:

[VOIP programvvPhone

Description: 一个采用SIP标准协议的VOIP软电话,解压后,直接运行BAT文件注册控件就可以完全运行!-A standard protocol for SIP using the VOIP soft phone, after extracting, run the BAT file directly control register can be fully operational!
Platform: | Size: 2751488 | Author: 幻想 | Hits:

[VOIP programSipGen20051013

Description: 免费开放的SIP消息产生器,用于SIP并发测试,能以多种方式循环发送和接收SIP消息,并能根据事先配置的脚本自动统计、显示消息发送或是接收的情况。-Free and open source SIP generator for SIP concurrent testing cycle to a variety of ways to send and receive SIP messages and can be configured in accordance with the prior script automatically statistics, displays the message sent or received by the situation.
Platform: | Size: 44032 | Author: brian | Hits:

[TCP/IP stackSIPDLLDemo

Description: sip 商业开发包,你可以免费使用,谢谢! 你可以免费使用,谢谢!-sip a commercial development kit, you can use free of charge, thank you! You can use free of charge, thank you!
Platform: | Size: 2829312 | Author: | Hits:

[VOIP programkamailio-3.0.2_linux_i386.tar

Description: Kamailio是一个开源的SIP服务器,原名OpenSER 该版本主要修复了代码中的一些小问题,完善了文档,建议使用 3.0 和 3.0.1 版本的用户升级。 -SIP Router (sip-router) is an industrial-strength, free VoIP server based on the Session Initiation Protocol (SIP RFC3261). It is engineered to power IP telephony and presence infrastructures up to large scale. The server keeps track of users, sets up VoIP sessions, relays instant messages and creates space for new plug-in applications. Its proven interoperability guarantees seamless integration with components from other vendors, eliminating the risk of a single-vendor trap. It has successfully participated in various interoperability tests in which it worked with the products of other leading SIP vendors.
Platform: | Size: 6372352 | Author: 侯旭光 | Hits:

[VOIP programPython

Description: 用pjsip的库编写web网络电话,说明: 1.先安装pjsip的库。 2.你需要两个sip账号,分别填入第114行和116行。 第一个参数是SIP服务器,第二是用户名,第三是密码。 3.被叫号在第49行,你可以修改成从文件读或者其他。 4.第131行,从E盘根目录获取“电话号码.call”文件。获取之后就把该文件删掉。并呼叫该号码。 5.第44行,等待音文件,必须是标准的wav文件,可以用windows自带的录音机自己录制,转码出来的有可能不被支持,会报异常,此处未作处理 6.本程序是拿来做网页免费电话(web800)用的,当用户在web上填入一个号码,就可以自动回拨他,播放等待音,并接通第49行设定的客服号码。 7.web部分考虑到各人服务器不一样,未给出,只需按格式写入一个空文件到131行设定的路径就可以-Written in the pjsip the library web network telephone, stating: 1 to install the pjsip the library. You need two sip accounts, respectively, to fill 114 lines and 116 lines. The first parameter is the SIP server, the second is the user name and password. 3 called in line 49, you can modify the file read or other. Four. Line 131, obtained from the root directory of E telephone number call "file. Fetch and then put the file is deleted. And call the number. 5 line 44, waiting for the sound file must be a standard wav file, you can use windows built-in recorder to record your own out of transcoding may not be supported, exception will be reported here is not dealt with This procedure is used as the web pages Toll-free (web800) used when the user fill in a number on the web, you can automatically call back him, the player waiting tone, and connected to the line 49 to set the customer service number. 7.web part of each one taking into account the server is not the same, is not given
Platform: | Size: 541696 | Author: wenda | Hits:

[TCP/IP stacksIP

Description: 随机IPv4地址生成程序,不是很复杂,如果需要批量生成,可自行修改。-Random IPv4 address generator is not very complicated, if you need bulk generation, free to modify.
Platform: | Size: 9099264 | Author: 刘建 | Hits:

[Communicationfreeiris2-3.1.524-stable.tar

Description: Free表示自由,开放,共享, iris为希腊神话中的彩虹女神,宙斯的通信官· 从2005年起Freeiris项目由开源通信爱好者和开源软件推动者所创建的一个平台, 将理念, 技术, 信仰融合在了一起。 Freeiris的代码最早由hoowa.sun编写并以开放源代码软件的形式公布到全世界, 现在已经有了大量的用户. Freeiris实现了更容易方便的传统电话线, ISDN-BRI, 数字中继 ISDN-PRI, 中国7号线路的支持. Freeiris同样也广泛的支持SIP和IAX2这种网络通信协议. 同时支持包括中国,美国,欧洲标准的电话系统以便在商用领域实现融合. Freeiris(前身为Astercon2)是一款开源的电话通信平台,含盖了计费、注册管理、PBX、数字中继、呼叫中心等业务需要。系统基于Asterisk、Perl、Linux、PHP等技术实现,在不修改asterisk本身的情况下采用外挂形式开发。目前系统可以控制管理SIP、IAX、H323、等软协议的通信。在硬件层上,支持大量的硬件产品,包括数字中继(ISDN PRI / ISDN BRI),模拟中继(FXO / FXS),以及各种SIP与IAX的终端设备等。由于freeiris项目将完全停止,包括下载服务器也随即关闭,这里上传收藏一下啊,为freeiris项目做一下备份。-Free said that freedom, openness, sharing, iris for rainbow goddess in Greek mythology, Zeus platform communication officer from 2005 Freeiris open source communications project by enthusiasts and promoters of open source software is created, the concepts, technologies, beliefs fused together. Freeiris code was first written by hoowa.sun and in the form of open source software released to the world now have a large number of users. Freeiris achieve a more easy and convenient traditional telephone lines, ISDN-BRI, digital trunk ISDN- PRI, to support China on the 7th line. Freeiris also broad support SIP and IAX2 this network communication protocols at the same time support, including China, the United States, the European standard telephone system in order to achieve integration in the commercial field. Freeiris (formerly known as Astercon2) is an open source telephony platform, covering the business needs billing, registration management, PBX, digital trunking, call center. System base
Platform: | Size: 8815616 | Author: park | Hits:

[VOIP programpjproject-2.4

Description: PJSIP是用C语言编写实施基于标准协议,如SIP , SDP , RTP , STUN , TURN , ICE和一个自由和开放源码的多媒体通信库。它结合了信令协议( SIP)具有丰富的多媒体框架和NAT功能集成到高级别API,几乎适用于任何类型的系统,从台式机,嵌入式系统,移动手机。-PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging desktops, embedded systems, to mobile handsets.
Platform: | Size: 7506944 | Author: Song Ni | Hits:

[CSharpsipek

Description: sipek sip client . free and open source project .
Platform: | Size: 2407424 | Author: behnam | Hits:

[VOIP programpjproject-2.0-alpha2

Description: PJSIP是一个开放源代码的SIP协议栈,它支持多种SIP的扩展功能 。它的实现是为了能在嵌入式设备上高效实现SIP/VOIP。-PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging desktops, embedded systems, to mobile handsets.
Platform: | Size: 6575104 | Author: 亚新123 | Hits:

[Delphi VCLsipxecs-release-15.06

Description: Component SIP, free and very simple
Platform: | Size: 2048 | Author: bublitz | Hits:

[VOIP programpjproject-2.7.1

Description: PJSIP是一个开源的SIP协议库,它实现了SIP、SDP、RTP、STUN、TURN和ICE。PJSIP作为基于SIP的一个多媒体通信框架提供了非常清晰的API,以及NAT穿越的功能。PJSIP具有非常好的移植性,几乎支持现今所有系统:从桌面系统、嵌入式系统到智能手机。(PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile handsets.)
Platform: | Size: 9094144 | Author: JackyDev | Hits:

[VOIP programHiPhones

Description: PJSIP 由英国Teluu团队主导开发,由Benny Prijono 创建,他的名字缩写pj,所以命名PJSIP 优点: 可移植性强:可运行在windows、windowsmobile、linux、unix、MacOS、RTEMS、Symbian 内存需求小:编译后只需要150k内存空间 支持多种SIP功能以及扩展功能:支持多人会话、事件驱动框架、会话控制(presence)、即时信息、电话传输 文档介绍:官网有教程可以学习 缺点: Demo代码之间关联比较紧密,在编译的时候,需要花费时间寻找依赖关系 文档特别多,也容易理不清(PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with rich multimedia framework and NAT traversal functionality into high level API that is portable and suitable for almost any type of systems ranging from desktops, embedded systems, to mobile handsets.)
Platform: | Size: 5766144 | Author: 肖阳阳 | Hits:

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