关于视频传输系统开源框架的调研结果

一位师姐做过类似的调研后说，当初她的公司提出的要求是，在终端必须使用flash播放，这样的话，只支持rtmp协议，就得采用red5。另一点，开发的下一阶段要支持移动终端，这样RTSP协议就比较好，支持RTSP的开源代码主要就是live555了。

live555

使用live555 Streaming media建立RTSP串流服务器

http://www.wl-chuang.com/blog/2011/06/12/building-up-a-rtsp-slash-rtp-server-using-live555/#disqus_thread

Live555 Streaming Media是一套优秀的开放程序代码函示库，拥有完整的RTSP/RTP实作，除了支持RTP over UDP外，尚支持RTP over RTSP以及RTP/RTSP over HTTP，可用于传输多种主流的影音压缩格式如：H.264、MPEG4、MP3….等。知名的播放软件VLC的串流功能就是基于此函式库开发。 利用它我们只要短短几行程序代码即可建立自己的串流服务器。

范例程序代码如下：

#include <liveMedia.hh>

#include <BasicUsageEnvironment.hh>

static void announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
                           char const* streamName, char const* inputFileName);

int main(int argc, char** argv)
{
    UsageEnvironment* env;
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    env = BasicUsageEnvironment::createNew(*scheduler);

    // 1. Create the RTSP server:
    RTSPServer* rtspServer = RTSPServer::createNew(*env, 8554, NULL);
    if (rtspServer == NULL) {
        *env << "Failed to create RTSP server: " << env->getResultMsg() << "\n";
        exit(1);
    }

    // 2. Add a H.264 video elementary stream to RTSP server
    {
        char const* streamName = "h264";
        char const* inputFileName = "test.h264";

        ServerMediaSession* sms = ServerMediaSession::createNew(*env,
                                                                streamName,
                                                                streamName,
                                                                "H.264 video");
        sms->addSubsession(H264VideoFileServerMediaSubsession::createNew(*env,
                                                                         inputFileName,
                                                                         False));
        rtspServer->addServerMediaSession(sms);
        announceStream(rtspServer, sms, streamName, inputFileName);
    }


    // 3. Kick-off the server
    env->taskScheduler().doEventLoop(); /* Never return */

    /* Unreachable */
    return 0;
}

static void  announceStream(RTSPServer* rtspServer, ServerMediaSession* sms,
                            char const* streamName, char const* inputFileName)
{
    char* url = rtspServer->rtspURL(sms);
    UsageEnvironment& env = rtspServer->envir();

    env << "\n\"" << streamName << "\" stream, from the file \""
        << inputFileName << "\"\n";
    env << "Play this stream using the URL \"" << url << "\"\n";

    delete[] url;

}

由范例可知，live555的使用方式十分简单易懂，函式库编译连结后的执行档也非常小巧，比较遗憾的是因为它目前没有提供与外部编码器(encoder)链接的功能以致无法撷取实时影音做实时串流，仅能播放预录的影音文件，这部份的功能有赖函式库使用者者自行撰写。这是与GStreamer提供的RTSP server相比较为美中不足的地方，但是对一般使用者来说应该绰绰有余了。

流讯博客

http://www.liuxun.org/archives/category/live555
视频编解码介绍、流媒体协议简介、RTSP录像的几种方式、live555－OpenRTSP分析、

LIVE555官方网站

http://www.live555.com/

基于标准RTP/RTCP/RTSP/SIP 协议的流媒体源码库。适合做嵌入式和低开销的流媒体应用。参考如下内容：
• "openRTSPTM" – 命令行形式的 RTSP 客户端
• "playSIPTM" – 命令行形式的 SIP 会话 recorder
• "wis-streamer" – Linux WIS GO7007 Encoder Driver 上的开源流媒体服务器
• "MPlayer" media player 上的 RTSP/RTP 流媒体服务  
• "vobStreamerTM" – 网络 DVD player (在LAN上将 DVD 内容化为流媒体)

LIVE555 Streaming Media

http://www.live555.com/liveMedia/

译文博客

http://www.jishuyuanma.com/article/xue_fu/8292841.html

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" and "LIVE555 Proxy Server" (RTSP server applications), "liveCaster" and "playRTPMPEG" (for streaming MP3 audio using RTP/RTCP), and "vobStreamer" (for streaming DVD content using RTP/RTCP/RTSP). The libraries can also be used to stream, receive, and process MPEG, H.264, H.263+, DV or JPEG video, and several audio codecs. They can easily be extended to support additional (audio and/or video) codecs, and can also be used to build basic RTSP or SIP clients and servers, and have been used to add streaming support to existing media player applications, such as "VLC" and "MPlayer". (For some specific examples of how these libraries can be used, see the test programs below.)

Source code

The project source code is available - as a ".tar.gz" file - here. See below for instructions on how to build it.

Mailing list

There is a developers' mailing list: "live-devel@lists.live555.com". Users (or prospective users) of the libraries are encouraged to join this (low-volume) mailing list, and/or to review the mailing list's archives. (You can also search these archives using Google, by adding "site:lists.live555.com" to your search query.) Before posting to the mailing list for the first time, please read the FAQ, to check if your question has already been answered.

Support

The primary means of support for these libraries is the "live-devel@lists.live555.com" mailing list described above. (Note that you must first subscribe to the mailing list before you can post to it.)

Are you planning to implement RTP (and/or RTSP or SIP)? Instead of writing your own implementation from scratch, consider using these libraries. They have already been used in many real-world RTP-based applications, and are well-suited for use within embedded systems. The code includes an implementation of RTCP, and can easily be extended (via subclassing) to support new RTP payload types.

1. Description (including test programs)
2. How to configure and build the code on Unix (including Linux, Mac OS X, QNX, and other Posix-compliant systems)
3. How to configure and build the code on Windows
4. Frequently Asked Questions (FAQ). (Please read this before posting questions to the mailing list.)
5. Source code license
6. To do...
7. Some third-party applications

live555 在VS2010 下live555编译、使用及测试

http://www.cnblogs.com/skyseraph/archive/2012/04/11/2442840.html 

文档内容很全，排版更好看。完全可以参考这个文档做测试。

在windows使用vs2008编译live555

http://www.360doc.com/content/10/1206/17/4914074_75551056.shtml

我做了一下，按这个文档做也可以顺利完成编译。

GStreamer官方网站

http://gstreamer.freedesktop.org/
GStreamer 是一个 基于流水线的多媒体框架，基于GObject，以C语言写成。
凭借GStreamer，程序员可以很容易地创建各种多媒体功能组件，包括简单的音频回放，音频和视频播放,录音，流媒体和音频编辑。基于流水线设计，可以创建诸如视频编辑器、流媒体广播和媒体播放器等等的很多多媒体应用。

RTSP with python

Playing RTSP with python-gstreamer

http://stackoverflow.com/questions/4192871/playing-rtsp-with-python-gstreamer
I use gstreamer for playing RTSP stream from IP cameras (like Axis.)。。。
I want to control it with a gui in pygtk so I use the gstreamer python bindings.。。。

Python-RTSP

Implementation of the RTSP protocol on top of the Twisted Python library. —Read more
http://odie5533.com/
https://github.com/odie5533/Python-RTSP


http://www.flumotion.net/doc/flumotion/reference/trunk/flumotion.twisted.rtsp-module.html

实现RTP协议的H.264视频传输系统

http://blog.csdn.net/gavinr/article/details/7035966

有源码，初看了一下，比较全。

由于是一个公司做的，所以有些地方要注册。权限是否有要求尚且不明。

RTSP 播放器 demo

http://blog.csdn.net/nkmnkm/article/details/7753672

很明显看出来这个公司做的demo非常好用，可以做开发的重要参照。

唯一不足是提供方式是dll不是源码。