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Description: RFC 3372协议 中文版
应用于电话的会话初始化协议：内容和体系-RFC 3372 agreement calls for the Chinese version of the Session Initiation Protocol : content and system
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Size: 121816 |
Author: javatea_ppl |
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Description: SIP 协议是下一代网络中的接口协议之一，属于应用控制协议。本标准是以
IETF 和ITU-T 的相关标准为基础，结合中国电信网络的实际情况，并综合中国
电信集团公司对下一代网络的实验成果制定的。-SIP is the next generation network interface protocol, application control agreements belong. The standard is based on the IETF and the ITU-T standards for the relevant infrastructure and, in light of China's actual telecommunications network, Comprehensive and China Telecom network for the next generation of experimental results enacted.
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Size: 1147364 |
Author: 马箫摩 |
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Description: Session Initiation Protocol for Telephones (SIP-T)
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Size: 14195 |
Author: andyboy |
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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: 203776 |
Author: 张三 |
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Description: 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，同时还可通过标准MGCP（Media Gateway Control Protocol ）协议，与华为技术有限公司SoftSwitch 软交换设备配合组网，在SoftSwitch 控制下完成主被叫间的话路接续。-IAD (Integrated Access Device) 132E (T) Integrated Access equipment (hereinafter referred IAD132E (T)) is Huawei Technologies Co., Ltd. NGN (Ne Anhui Generation Network) solutions to the important parts, for the company to provide users such as small-capacity VoIP (Voice over IP)/FoIP (Fax o ver IP) solutions. IAD132E (T) as the VoIP/FoIP Media Access Gateway, NGN users for access layer, complete with simulated voice signals between IP packet switching and through packet-switched data transmission network functions, It can also through standard MGCP (Media Gateway Control Protocol ) agreement with Huawei Technologies Co., Ltd. SoftSwitch equipment with soft switch network, SoftSwitch in complete control of the main Called the ground successor.
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Size: 11264 |
Author: 孙昊 |
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Description: RFC 3372协议 中文版
应用于电话的会话初始化协议：内容和体系-RFC 3372 agreement calls for the Chinese version of the Session Initiation Protocol : content and system
Platform: |
Size: 121856 |
Author: javatea_ppl |
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Description: SIP 协议是下一代网络中的接口协议之一，属于应用控制协议。本标准是以
IETF 和ITU-T 的相关标准为基础，结合中国电信网络的实际情况，并综合中国
电信集团公司对下一代网络的实验成果制定的。-SIP is the next generation network interface protocol, application control agreements belong. The standard is based on the IETF and the ITU-T standards for the relevant infrastructure and, in light of China's actual telecommunications network, Comprehensive and China Telecom network for the next generation of experimental results enacted.
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Size: 1146880 |
Author: 马箫摩 |
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Description: Session Initiation Protocol for Telephones (SIP-T)
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Size: 14336 |
Author: andyboy |
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Description: SIP协议说明，希望对学习SIP协议的同志有所帮助！
SIP 的协议分析与实现是万方的数据，最近在做SIP协议的移植，
感觉挺好的。-SIP agreement, hoping to learn from Comrade SIP protocol help! SIP-protocol analysis with data万方realize is, most recently in the SIP protocol to do the transplant, I feel quite good.
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Size: 2487296 |
Author: qiumh |
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Description: FxRobot width ACE and sip, anthony.wei 2009-11-11, @a t a i n In ShenZhen-FxRobot width ACE and sip, anthony.wei 2009-11-11, @ atain In ShenZhen
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Size: 108544 |
Author: David Lee |
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Description: SIP programming in Java
- applications
- protocols
- socket ...
Tested and developed by MonteCristo(H.U.T)
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Size: 50176 |
Author: montecristo |
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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: 在传统电话系统中，一次通话从建立系统连接到拆除连接都需要一定的信令来配合完成。同样，在IP电话中，如何寻找被叫方、如何建立应答、如何按照彼此的数据处理能力发送数据，也需要相应的信令系统，一般称为协议。目前在国际上，比较有景响的IP电话方面的协议包括ITU-T提出的H.323协议和IETE提出的SIP协议，本节主要介绍目前用得最广泛H.323协议。-In the traditional telephone system, a call from the establishment of systems to connect to dismantle connection with the completion of signaling. Likewise, IP telephony, how to find the called party, how to create a response, how to send data to each other s data processing capabilities, signaling systems, commonly known as the Agreement. Internationally, more King loud IP telephony protocols, including ITU-T H.323 and SIP protocol IETE, this section introduces the most widely used H.323 protocol.
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Size: 12288 |
Author: lxx |
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Description: SIP doesn t need CommandsInterface. The class does nothing but made to work with PhoneBase s constructor.
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Size: 2048 |
Author: riubenmin |
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Description: 广域网质量监控软件，VoIP Monitor 模块帮助您测定与跟踪最需要关注的广域网语音质量性能。-VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters- delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP/SKINNY protocol or SIP/RTP/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio.
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Size: 908288 |
Author: raymond |
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Description: 使用PIC16F877A控制nRF24L01无线模块和DHT11传感器实现无线温湿度采集，24L01接PIC的SIP接口RC3\4\5，单片机晶振4M，接收端串口波特率9600传送H=xx T=-Use PIC16F877A control nRF24L01 wireless module and wireless temperature and humidity sensor DHT11 collection, 24L01 pick PIC' s SIP interface RC3 \ 4 \ 5, microcontroller crystal 4M, 9600 baud serial receiving end transmission H = xx T = xx
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Size: 60416 |
Author: Xingzhixiang |
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Description: jVoIP is a simple SIP Phone based on NIST Sip Communicator.
The phone won t be a fully functional SIP Client:
it won t support REGISTER request
it will be design for LAN voice communication
This phone won t function behind NAT or FireWall
This project isn t actively developed, the interface language is only in italian
Features
nice and simple to use GUI
simple configuration
high-level assumptions
We will not, consider certain functional areas like internationalization, high security, istant messaging and video phone
Screenshots
jVoip main window appears like
jVoIP settings window appears like
Notes
This project is built using Eclipse. To checkout source code refer to the CVS instructions. Then to build it you need install and configure JMF, the Java Media Framework.
To Do list
Localize in english and other languages
Remove uneeded class
Fix and add comments
Related resources
Sip Communicator: http://sip-communicator.dev.java.net/
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Size: 6132736 |
Author: lifawen |
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Description: VoIPmonitor 是一个开源的实时网络抓包并分析SIP和RTP protocol的程序. -VoIPmonitor is open source live network packet sniffer which analyze SIP
and RTP protocol. It can run as daemon or analyzes already captured pcap
files. For each detected VoIP call voipmonitor calculates statistics about
loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according
to ITU-T G.107 E-model. These statistics are saved to MySQL and each
call is saved as pcap dump. Web PHP application (it is not part of open
source sniffer) filters data and graphs latency and loss
distribution. Voipmonitor also detects improperly terminated calls when
BYE or OK was not seen. To accuratly transform latency to loss packets,
voipmonitor simulates fixed and adaptive jitterbuffer.
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Size: 1200128 |
Author: monk_lee |
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Description: java编写的sip客户端聊天。单机多机测试均可通过，希望大家喜欢-java PCclient of sip。
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Size: 2660352 |
Author: 王军伟 |
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Description: 添加使用PHP对接sip协议的封装接口搜索(i don't speak English. you now? sometime i can say i don't used the port)
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Size: 7168 |
Author: charles22
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