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Description: SCTP协议栈，SCTP是一种IP层的应用协议，实际上是与TCP,UDP并列的协议，SCTP传输比TCP更可靠，目前主要用于SIP类通讯协议。-SCTP, SCTP is a layer of IP Application Protocol is actually TCP, UDP tied to the agreement, SCTP transmission more reliable than TCP, the key category for SIP communication protocol.
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Size: 203625 |
Author: 张三 |
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Description: jsr-180 (SIP) 实现源码。可以在真实手机上使用，但需要获得公网地址，只实现了UDP传输。只可研究学习，不可用于商业用途。-JSR-180 (SIP) to achieve source. In real phone use, but need public address, the only achieved UDP transmission. Learning can only study and may not be used for commercial purposes.
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Size: 55744 |
Author: zs |
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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.
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Size: 11133 |
Author: adonmiao |
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Description: First of all, the Applet-phone is a SIP User-Agent
with audio and text messaging capabilities. But i s also embedded in
an applet where you can use it as usual if your firewall allow you to
use the UDP transport. If not, you can use TCP to carry your voice in a
voice messaging fashion. Your voice is recorded and encoded locally
then sent as a message, decoded and played back by your peer. Voice
messaging allow you to go through any firewall, so that you can
continue to chat and talk with your friends from an applet !
Platform: |
Size: 2630474 |
Author: 康佳 |
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Description: SCTP协议栈，SCTP是一种IP层的应用协议，实际上是与TCP,UDP并列的协议，SCTP传输比TCP更可靠，目前主要用于SIP类通讯协议。-SCTP, SCTP is a layer of IP Application Protocol is actually TCP, UDP tied to the agreement, SCTP transmission more reliable than TCP, the key category for SIP communication protocol.
Platform: |
Size: 203776 |
Author: 张三 |
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Description: jsr-180 (SIP) 实现源码。可以在真实手机上使用，但需要获得公网地址，只实现了UDP传输。只可研究学习，不可用于商业用途。-JSR-180 (SIP) to achieve source. In real phone use, but need public address, the only achieved UDP transmission. Learning can only study and may not be used for commercial purposes.
Platform: |
Size: 55296 |
Author: zs |
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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 |
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Description: First of all, the Applet-phone is a SIP User-Agent
with audio and text messaging capabilities. But i s also embedded in
an applet where you can use it as usual if your firewall allow you to
use the UDP transport. If not, you can use TCP to carry your voice in a
voice messaging fashion. Your voice is recorded and encoded locally
then sent as a message, decoded and played back by your peer. Voice
messaging allow you to go through any firewall, so that you can
continue to chat and talk with your friends from an applet !
Platform: |
Size: 2630656 |
Author: 康佳 |
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Description: 软电话
从功能分为：服务器端和客户端
服务器：记录所有客户的信息，对客户的登陆、注销等请求进行响应
客户端：利用UDP协议与服务器和被叫或主叫客户端进行通讯
点对点通讯中的请求和响应消息是通过SIP协议来完成传输的，音频信息（压缩后）是同构RTP协议来完成传输的-Soft phone functionality is divided into: server-side and client-server: a record of all customer information, customer landing, write-offs, such as in response to client requests: the use of UDP protocol with the server and the called or caller client communication point of point communication in the request and response messages through the SIP protocol to complete the transmission, audio information (compressed) is isomorphic RTP protocol to complete the transmission
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Size: 2237440 |
Author: 苍穹 |
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Description: sip服务器原代码,要求在linux或者unix下安装,可以用于建立自己的voip系统,基于sip协议-OpenSIPS brings:
robust and performant SIP (RFC3261) Registrar server, Location server, Proxy server and Redirect server
small footprint- the binary file is small size, functionality can be stripped/added via modules
plug&play module interface- ability to add new extensions, without touching the core, therefore assuring a great stability of core components
stateless and transactional statefull SIP Proxy processing
support for UDP/TCP/TLS/SCTP transport layers
IPv4 and IPv6
support for SRV and NAPTR DNS
SRV DNS failover
IP Blacklists
multi-homed (mhomed) and multi-domain support
flexible and powerful scripting language for routing logic
variables support in script- script variables, pseudo-variables (access to the SIP messages), AVPs (values persistent per SIP transactions)
management interface via FIFO file and unix sockets
authentication, authorization and accounting (AAA) via database (MySQL, Postgress, text files), RADIUS and DIAMETER
digest and IP authentication
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Size: 3659776 |
Author: Banlyst Yeh |
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Description: 结合OpenSIPs作为高效可扩展的传输媒体流服务器。-MediaProxy is a media relay for RTP/RTCP and UDP streams that works in tandem with OpenSIPs to provide NAT traversal capability for media streams from SIP user agents located behind NAT.
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Size: 176128 |
Author: 黄先生 |
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Description: 本系统设计并实现了基于 LPC2292, DM9000 的嵌入式网络通讯协议，协议遵循 RFC 标准 TCP/IP 规范。使用标准 C++ 语言在 ADS 下开发，完全源代码，方便移植与维护，代码小且运行效率高，容易进行二次开发。
系统实现了：以太网数据帧的收发、ARP 请求与应答、ICMP 中的 Ping 请求与应答、IP、UDP、简易的路由处理，可跨越网关收发数据。
使用本系统可进行各种具备网络功能产品的开发，如 VoIP 电话系统, 网络打印机，终端系统，各种监控系统、数据采集系统、远程控制系统等。-The system was designed and implemented based on LPC2292, DM9000 embedded network communication protocol, protocol to follow RFC standards for TCP/IP standard. Using standard C++ language, developed in the ADS, complete source code to facilitate the migration and maintenance, the code small and efficient operation, easy for secondary development. System implementation: the sending and receiving Ethernet data frames, ARP requests and responses, ICMP Ping requests and response in, IP, UDP, simple routing, may send and receive data across the gateway. Use of this system with network function can be a variety of product development, such as VoIP phone systems, network printers, end systems, various monitoring systems, data acquisition system, remote control systems.
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Size: 13312 |
Author: limei |
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Description: linphone-3.3.0 版本，这个就不多做介绍了，知道的人都知道！如果还是不知道，那么看看下面英文的介绍吧。-Features
Linphone provides a large amount of functionalities.
List of supported features:
* audio with the following codecs: speex (narrow band and wideband), G711 (ulaw,alaw), GSM, and iLBC (through an optional plugin)
* video with codecs: H263-1998, MPEG4, theora and H264 (thanks to a plugin based on x264), with resolutions from QCIF(176x144) to SVGA(800x600) provided that network bandwidth and cpu power are sufficient.
* Supports any webcam with V4L or V4L2 driver under linux
* Any webcam on windows
* text instant messaging and presence (using the SIMPLE standart)
* Addressbook
* DTMF (telephone tones) support using SIP INFO or RFC2833
* understands SIP ENUMS (sip phone numbers using the naptr DNS service, without proxy)
* echo cancelation using the great speex echo canceler
* SIP proxy support: registrar, proxies, with digest authentication
* STUN support for traversal of UDP NATs (=firewall)
* sound backend using either ALSA (t
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Size: 8246272 |
Author: 独孤一笑 |
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Description: SIP协议是NGN中的重要协议，是一个基于文本的应用层控制协议，独立于底层传输协议TCP/UDP/SCTP，用于建立、修改和终止IP网上的双方或多方多媒体会话。-NGN SIP protocol is an important protocol is a text-based application-layer control protocol, independent of the underlying transport protocols TCP/UDP/SCTP, for creating, modifying and terminating IP network multimedia sessions two or more parties
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Size: 564224 |
Author: chen |
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Description: Full Internet Library with Protocols and RFC Standards SIP, UDP, STUN, DNS, SNTP, POP3 SMTP, TCP, UDP, UPnP for Internet development 2015.
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Size: 6663168 |
Author: pedroapereza |
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Description: webrtc sip协议互转,websocket to udp-Sip webrtc protocol interoperability
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Size: 430080 |
Author: 张世杰 |
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Description: SIP UDP redirect and registrar server
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Size: 199680 |
Author: ph379411 |
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