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Description: 随着移动通信技术的发展和移动终端性能的提高,移动通信网络提供流媒体服务已经成为可能。分析移动流媒体网络传输协议和H.264视频编码格式特点,参照802.11标准,提出移动流媒体客户端的总体架构,并针对Windows Mobile平台特点予以实现。移动流媒体客户端可以运行在无线局域网中,主要由网络模块、解码模块和音视频回放模块组成。网络模块以动态库的方式实现RTP/RTCP/RTSP网络协议,采用消息队列的设计模式提高网络模块的效率 H.264与AMR解码模块由PC向嵌入式平台移植并作相应的封装 流媒体数据回放模块根据系统提供的资源,综合考虑效率与通用性,自动选择最优回放模式。根据3GPP移动流媒体总体结构要求,结合移动网络资源有限和移动智能终端处理能力较弱的特点,扩展局域网移动流媒体客户端的功能,优化流媒体播放器的架构,提高播放系统的效率。在局域网流媒体客户端设计的基础上,使用内网穿越技术穿透移动网关,实现流媒体数据通过移动通信网络传输。在网络模块中加入传输码率适配技术以根据移动通信网络的带宽自动调整服务器端数据发送速率。修改流媒体缓冲区数据排队策略,充分利用网络数据包之间的相关性-Streaming media services over mobile communication network come to be possible because of the development of the mobile communication technology and the improvement of the mobile terminals performance.Based on 802.11 standards, we have submitted the framework of the mobile streaming media clients and it is implemented on the Windows Mobile platform through the analysis on the mobile streaming media network transport protocol and the features of H.264 video encoding format.
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Size: 1184768 |
Author: zhz |
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Description: 基于java的rtp实现，采用rtp和rtcp来控制实时传输-Java-based implementation of the rtp, rtp and rtcp used to control real-time transmission. . .
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Size: 146432 |
Author: Allen young |
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Description: ORTP，新的rtp/rtcp库，纯c写的，也有c++应用例子介绍-ORTP, the new rtp/rtcp library, written in pure c, c++ applications also introduce an example
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Size: 453632 |
Author: guyong |
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Description: 基于RTP/RTCP的H.264视频实时传输系统设计-Based on RTP/RTCP real-time H.264 video transmission system design
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Size: 178176 |
Author: guyong |
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Description: vocal源码，及sip/rtp/rtcp使用用例及ua工程。-vocal source, and the sip/rtp/rtcp use cases and ua project.
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Size: 1546240 |
Author: 古红 |
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Description: In this paper, we present a modification of the ns2 code for the
RTP/RTCP protocols. The legacy RTP/RTCP code in ns2 has not
yet been validated but it provides a framework of the protocol’s
specification for experimental use. We have modified the code by
adding all the RTP/RTCP protocol’s attributes that are defined in
RFC 3550 and related to QoS metrics. We have also
implemented additional algorithms and functions in order to
enhance our modified code with TCP friendly bandwidth share
behavior. Our protocol, named RTPUP (“UP” stands for the
University of Patras), is offered as a package and is fully
documented so that it can be used for simulations and research
within the ns2 simulation environment. -In this paper, we present a modification of the ns2 code for the
RTP/RTCP protocols. The legacy RTP/RTCP code in ns2 has not
yet been validated but it provides a framework of the protocol’s
specification for experimental use. We have modified the code by
adding all the RTP/RTCP protocol’s attributes that are defined in
RFC 3550 and related to QoS metrics. We have also
implemented additional algorithms and functions in order to
enhance our modified code with TCP friendly bandwidth share
behavior. Our protocol, named RTPUP (“UP” stands for the
University of Patras), is offered as a package and is fully
documented so that it can be used for simulations and research
within the ns2 simulation environment.
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Size: 272384 |
Author: noname |
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Description: rtp/rtcp/rtsp文档,主要是资料查询-rtp/rtcp/rtsp doc, for search iportant info
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Size: 266240 |
Author: zhang |
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Description: This code forms a set of C++ libraries for multimedia streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries - which can be compiled for Unix (including Linux and Mac OS X), Windows, and QNX (and other POSIX-compliant systems) - can be used to build streaming applications. The libraries are already being used to implement applications such as "the LIVE555 Media Server" (a RTSP server application), "liveCaster" and "playRTPMPEG" (for streaming MP3 audio using RTP/RTCP), and "vobStreamer" (for streaming DVD content using RTP/RTCP/RTSP).-This code forms a set of C++ libraries for multimedia streaming, using open standard protocols (RTP/RTCP, RTSP, SIP). These libraries- which can be compiled for Unix (including Linux and Mac OS X), Windows, and QNX (and other POSIX-compliant systems)- can be used to build streaming applications. The libraries are already being used to implement applications such as "the LIVE555 Media Server" (a RTSP server application), "liveCaster" and "playRTPMPEG" (for streaming MP3 audio using RTP/RTCP), and "vobStreamer" (for streaming DVD content using RTP/RTCP/RTSP).
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Size: 3809280 |
Author: gary |
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Description: 这是介绍RTP/RTCP协议的文件，对开发流媒体的然很有用-This is to introduce the RTP/RTCP protocol file, and then on the development of streaming media is useful
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Size: 193536 |
Author: 钟普 |
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Description: RTP/RTCP方面的学习.方便大家学习.-RTP/RTCP learning-RTP/RTCP leanrning,to study,interaction
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Size: 655360 |
Author: xt |
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Description: live555 是一个为流媒体提供解决方案的跨平台的C++开源项目，它实现了对标准流媒体传输协议如RTP/RTCP、RTSP、SIP等的支持-live555 is a streaming media solutions for the cross-platform C++ open source projects, it implements the standard streaming protocol such as RTP/RTCP,
RTSP, SIP and other support
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Size: 477184 |
Author: Tom |
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Description: 本人整理的RFC中关于rtp/rtcp等网络传输相关文档，包括经典的rfc3550、rfc3551、rfc4585，还有一些相关技术介绍文档，如rfc2119，rfc2354，rfc2733,rfc3448,rfc3556,rfc4340,rfc4566,rfc5109。对于研究视频通话和视频会议有很大帮助。文档都是.txt格式，英文的（个人感觉读文档还是英文的好，这样对一些专业术语都有很好的掌握）。-I sort of RTP/RTCP RFC and other network transmission relevant documents, including the classics rfc3550, rfc3551, rfc4585, and some related technical presentations, such as rfc2119, rfc2354, rfc2733, rfc3448, rfc3556, rfc4340, rfc4566, rfc5109. Video call for research and video conference has very great help. Documents are. TXT format, English (personal feeling read English documentation or good, so to some professional terms have very good master).
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Size: 336896 |
Author: 圣少友 |
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Description: 最新的jrtplib版本,多媒体实时流传输协议（RTP/RTCP）C++ 了lib包平台要求(linux,unix,windows)用于网络音频视频传输 --Jrtplib the latest version of real-time streaming multimedia transport protocol (RTP/RTCP) C++ The lib package require platform (linux, unix, windows) for network audio video transmission
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Size: 178176 |
Author: qdn |
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Description: 包含了与视频、RTSP相关的一些文档：H.264 RTP payload 格式、Linux下的实时流媒体编程(RTP,RTCP,RTSP)、RTSP、用实例分析H264 RTP payload 等。-Contains some documents related to the video, RTSP: H.264 RTP the payload format, real-time streaming media programming under Linux (RTP, RTCP, RTSP), RTSP example, analysis of the H264 and the RTP the payload.
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Size: 49152 |
Author: 张元 |
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Description: 目前的流媒体服务器代码有Darwin(苹果公司），是用c++开发的，过于复杂不适合用在嵌入式设备上；
live555是c开发的但是只支持mpeg格式的音频；Feng 流媒体服务器支持h.264格式的视频，mp3音频，支持RTSP,RTP/RTCP，可以用来在嵌入式设备上开发流媒体服务器。-The streaming server code Darwin (Apple), C++ development, too complex and not suitable for use in embedded devices
live555 c but only supports mpeg format audio Feng, streaming media server supports h.264 format video, mp3 audio, support RTSP, RTP/RTCP, the development of streaming media server that can be used in embedded devices.
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Size: 465920 |
Author: 徐留建 |
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Description: 新版本（2.1）。feng是一个多媒体的流媒体服务器与互联网多媒体内容实时流IETF的标准兼容。feng实现RTSP协议 - 实时流协议（rfc2326）和RTP / RTCP - 支持实时传输协议/ RTP控制协议（RFC3550）的RTP资料的最小控制（rfc3551）与音频和视频会议。
client 相容列表
libnemesi Live555.com FFPlay GStreamer RealPlayer HelixPlayer QuickTime
General RTSP OK OK OK OK OK OK OK
Seek support Yes Partial1 No No Yes Yes Yes
Pause support Yes Partial1 Yes Yes Yes Yes Yes
General RTP/RTCP OK OK OK OK OK OK Partial2
MPEG Video 1/2 Yes Yes Yes Yes3 Partial4 No Yes
MPEG Audio Yes Yes Yes Yes Yes Yes Partial5
Vorbis Yes No No Yes No No No
Theora No No No Yes No No No
H.264 Yes Yes Yes Yes No No Yes
H.263/H.263+ Yes Yes Yes Yes Partial6 No Partial7
MPEG 4 Visual Yes Yes Yes Yes Partial4 No Yes
AAC Yes Yes Yes Yes Yes No Yes
相依套件
glib version 2
ffmpeg version 0.5
netembryo
libev-new version(2.1). Feng is a multimedia streaming server compliant with the IETF s standards for real-time streaming of multimedia contents over Internet. Feng implements RTSP – Real-Time Streaming Protocol (rfc2326) and RTP/RTCP – Real-Time Transport Protocol/RTP Control Protocol (rfc3550) supporting the RTP Profile for Audio and Video Conferences with Minimal Control (rfc3551).
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Size: 575488 |
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Description: 基于Android的视频通讯协议 RTP / RTCP开发实例源码-Android based video communication protocol RTP/RTCP development source code examples
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Size: 193536 |
Author: hehe |
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Description: RTP/RTCP协议的传输与接收端，RtpDLL_Windows,用VC语言编写的！对初学者很有用！-The RTP/RTCP protocol for transmitting and receiving end, using VC language! Very useful for beginners!
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Size: 417792 |
Author: hyj |
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Description: live555流媒体经典分析,是学习Live555流媒体服务的重要文档-live555 RTSP RTP RTCP.
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Size: 147456 |
Author: 李鹏 |
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Description: GIPS Video&Voice Engine。
GIPSVideoEngine动态库说明：
GIPSVideoEngine是一个基于WebRTC的Video&Voice Engine，当前主要支持Windows和Android，Linux/MAC/iOS也在开发中，
支持如下功能：
1. 音频采集（支持Windows Wave和Windows core），编码（当前主要支持ISAC， iLBC， PCM， g722等，可以扩展支持729，723等），
音频处理（包含发送和接收端的Echo Control， Noise Suppression，Automatic Gain Control， Voice Activity Detectiong），
NetEQ， RTP/RTCP发送接收，解码，回放。
2. 视频的采集，编码（当前支持VP8，264还在开发中），RTP/RTCP发送，解码，回放。-GIPS Video & Voice Engine，support Video&Audio Record，Playback RTP/RTCP transport, Audio Processing(EC, VAD, AGC, NS)
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Size: 4332544 |
Author: 朱五 |
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