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Description: ip电话范例,采用sip协议簇,支持rtp, g.729系列编解码标准,rtp采用md5加密-ip phone model used sip agreement cluster support rtp, codec G.729 series standards, rtp encryption using md5
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Size: 51873 |
Author: 田野 |
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Description: 基于RTP协议的IP电话，有很强的理论可行性。详细介绍了RTP
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Size: 71035 |
Author: 王子 |
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Description: 基于IP的H.264关键技术：介绍了 IP网络应用环境，H.264的错误恢复工具；介绍如何在IP网络使用RTP分组传输H.264的视频技术和RTP载荷格式草案。
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Size: 11225 |
Author: menmen |
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Description: rtp 编程 媒体指的是在网络中使用流技术传输的连续时基媒体，其特点是在播放前不需要下载整个文件，而是采用边下载边播放的方式，它是视频会议、IP 电话等应用场合的技术基础。RTP 是进行实时流媒体传输的标准协议和关键技术，本文介绍如何在 Linux 下利用 JRTPLIB 进行实时流媒体编程。
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Size: 9676 |
Author: 军军 |
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Description: Delphi写的IP电话源码-IP phone sourcecode writed by Delphi
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Size: 221184 |
Author: 站长 |
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Description: VOIP电话源程序-source VoIP telephone
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Size: 395264 |
Author: 王海业 |
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Description: 此文件包含两个独立程序,Sender和Receiver,用于测试网络对RAW类型的IP包的传输，可以用这两个程序测试路由器是否支持RAW类型的IP包传输，是否支持组播。程序演示了Windows下RAW socket的使用和RTP包协议的利用.程序工程由VC7.0生成。-this document contains two separate procedures, Sender and Receiver for testing network to RAW types of IP packet transmission, could use this test procedure two routers support RAW types of IP packet transmission, whether to support multicast. Procedures demonstrated Windows RAW socket and the use of RTP packets agreement utilization. VC7.0 procedures generated by the project.
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Size: 31744 |
Author: 鲁锐 |
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Description: 此文件包含两个独立程序,Sender和Receiver,用于测试网络对RAW类型的IP包的传输，可以用这两个程序测试路由器是否支持RAW类型的IP包传输，是否支持组播。程序演示了Windows下RAW socket的使用和RTP包协议的利用.程序工程由VC7.0生成。-this document contains two separate procedures, Sender and Receiver for testing network to RAW types of IP packet transmission, could use this test procedure two routers support RAW types of IP packet transmission, whether to support multicast. Procedures demonstrated Windows RAW socket and the use of RTP packets agreement utilization. VC7.0 procedures generated by the project.
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Size: 31744 |
Author: 鲁锐 |
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Description: Vovida出品的包括RTP、RTCP、SIP等各种基础协议栈和SIP UAC、UAS等功能的IP电话服务器端程序。编译环境为linux-Vovida produced in RTP, RTCP, and other SIP based protocol stack and SIP UAC and UAS capabilities of IP telephony server procedures. Linux compiler environment
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Size: 16835584 |
Author: 张远文 |
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Description: RTP/UTP/IP网络编程-RTP/UTP/IP network programming
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Size: 4854784 |
Author: 施拓 |
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Description: H.264 RTP 打包格式中文译稿，针对视频应用/网络电话/IPTV人士-H.264 RTP packetization format Montagne video applications/VoIP/IPTV who
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Size: 45056 |
Author: 闫乐敏 |
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Description: 实现基于ip2022的MPEG2 TS的IP组播接收-based ip2022 MPEG2 TS receive IP Multicast
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Size: 1138688 |
Author: 赵孜恺 |
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Description: ip电话范例,采用sip协议簇,支持rtp, g.729系列编解码标准,rtp采用md5加密-ip phone model used sip agreement cluster support rtp, codec G.729 series standards, rtp encryption using md5
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Size: 51200 |
Author: 田野 |
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Description: rtp tunnel 一个非常好的rtp tunnel程序，支持tcp和udp tunnel，希望对大家有用
-rtp tunnel a very good rtp tunnel procedures, support tcp and udp tunnel, we hope to useful
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Size: 34816 |
Author: granite |
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Description: internet的tcp/ip协议客户端的实现源码，包括rtp,rtcp,rtsp-the internet tcp/ip client agreements to achieve the source, including rtp. rtcp, rtsp
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Size: 3072 |
Author: apple_spring_hu |
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Description: G726局域网语音通话程序和源代码
这是使用G726语音压缩（16kbps）和RTP进行传输的程序，因为我没有带WIFI的PPC，所以每个程序都是单独测试的，PC端和PPC端分别都工作正常。
G726编解码算法来自OpenH323.传输使用的RTP可以在RTP程序中找到讲解，这个程序主要是G726的函数。将整个 G726封装为g726_Encode和g726_Decode两个函数，参数为压缩和解压数据存储的地址指针，可以将960字节压缩到120字节和将 120字节解压为960字节。这里G726使用的时候，音频设置为8kHz，16位量化，单声道。
使用方法很简单，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。
PPC端的运行比较稳定，已经进行了自收自发近一个小时连续工作的测试，非常稳定，话音清晰。-G726 LAN voice calls procedures and the source code is the use of voice compression G726 (16kbps ) and RTP for the transmission process, because I did not bring WIFI the PPC, there are separate procedures for each test, PC- and PPC-were all normal work. G726 codec algorithm from the OpenH323. RTP transmission can be used in the process to find RTP stresses Xie, this procedure is mainly a function of G726. Packaging whole G726 for g726_Encode and g726_Decode two functions, Parameters for the compression and decompression of data storage address pointer, 960 bytes can be compressed to 120 to 120 bytes and bytes to 960 bytes decompression. G726 used here, Audio set to 8kHz, 16 quantify mono. It is very simple to use, with only two ends of the respective input each other's IP, then press
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Size: 1002496 |
Author: Xia Tao |
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Description: rtp的c++库。rtp是VoIP等IP多媒体传输协议，是处理网络延时、抖动、丢包的关键模块。-rtp the c library. Rtp of VoIP and other IP multimedia transmission protocol, network delay, jitter, packet loss of key modules.
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Size: 560128 |
Author: raosiyong |
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Description: G726局域网语音通话源代码(vc++ 6)
这是使用G726语音压缩（16kbps）和RTP进行传输的程序，使用方法很简单，因为没多少时间，并且RTP不面向连接，所以我也没做连接确认的，只用两端各自输入对方的IP，然后按下“开始对话”，就可以进行语音通信了。-G726 LAN voice calls source code (vc 6) This the voice compression using G726 (- ns 16-kb ps) and RTP for transmission, the use of approach is very simple, because no amount of time, RTP and not connection-oriented, so I do not recognize the link, only two ends of the respective input each other's IP, then press the "start a dialogue", it can implement a voice communication.
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Size: 577536 |
Author: Arnold |
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Description: 基于RTP协议的IP电话，有很强的理论可行性。详细介绍了RTP-Based on the RTP protocol IP phones, have a strong theoretical feasibility. Detailed information on RTP
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Size: 70656 |
Author: 王子 |
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Description: The Real-time Transport Protocol (RTP) provides a framework for delivery of audio and video across IP networks and unprecedented quality and reliability. In RTP: Audio and Video for the Internet, Colin Perkins, a leader of the RTP standardization process in the IETF, offers readers detailed technical guidance for designing, implementing, and managing any RTP-based system.-The Real-time Transport Protocol (RTP) provides a framework for delivery of audio and video across IP networks and unprecedented quality and reliability. In RTP: Audio and Video for the Internet, Colin Perkins, a leader of the RTP standardization process in the IETF , offers readers detailed technical guidance for designing, implementing, and managing any RTP-based system.
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Size: 3009536 |
Author: windl |
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