rtp

I try to make a H264 RTP stream from a Raspberry Pi 3 with a camera module to a video tag. Using the following code to start the stream raspivid -t 0 -h 720 -w 1080 -fps 25 -hf -b 2000000 -o - | \ gst-launch-1.0 -v fdsrc \ ! h264parse \ ! rtph264pay \ ! gdppay \ ! udpsink host="192.168.0.11" port=5000 Then I provide a simple webpage with a video tag: <video id="videoTag" src="h264.sdp" autoplay> <p class="warning">Your browser does not support the video tag.</p> </video> The src references the following SDP file: v=0 m=video 5000 RTP/AVP 96 c=IN IP4 192.168.0.51 a=rtpmap:96 H264/90000 When I

在LiveVideoStackCon2019深圳音视频技术大会前夕，我们邀请到了华为云核心网产品线，高级架构师左俊老师接受采访，采访中左俊老师从自身十五年的架构师经验出发浅谈了自己对于近几年音视频通话技术的发展，并从覆盖场景、数据安全和视频质量审核等方面介绍了华为云通信云服务的优势。 文 / 左俊 整理 / LiveVideoStack LiveVideoStack：左俊你好，能否先向LiveVideoStack的读者介绍下自己，以及你目前主要的工作以及关注的技术方向？ 左俊： 我目前主要的负责是华为云.通信云服务整个解决方案的架构和技术。华为云.通信云服务目前包括如下四个服务：视频通话服务、隐私保护通话服务、语音通话服务、短信服务。当下，我主要关注的有以下两点： 音视频方案中的效果及质量问题，这个是整个音视频解决方案最为基础的部分，华为在运营商市场实时音视频的成功实践，为解决互联网视频通话的效果和质量问题，提供有效的借鉴和支撑，这也是华为云.视频通话服务快速孵化的基础。 解决方案中的安全韧性及合法合规问题，虽然这不是基础业务的一部分，但是这个是客户能够获取持久保障的前提，华为很看重这一块，也能够保证向客户持续地提供可靠可信的服务。 此外还有系统的开放性灵活性，成本管理，客户的个性化服务等，也都是整个解决方案中不可或缺的一环。 LiveVideoStack

I'm trying to implement RTSP protocol in Java according to http://www.csee.umbc.edu/~pmundur/courses/CMSC691C/lab5-kurose-ross.html example; I have succeed up to successful communication with VLC via RTSP requests and streaming RTP packets; RTP packets with JPEG payload are not recognized by VLC well; that's why I supposed I send malformed RTP. I sniffed them with wireshark and compared them with packets, sniffed from successfull RTSP communication of gstreamer RTSP streamer and VLc. I was surprised, that both my app&VLC's RTSP and RTP requests were labeled in wireshark UI as simply TCP and

I want to read from an RTP stream, but when I specify "test.sdp" to avformat_open_input() I get this message: [rtp @ 03928900] Protocol not on whitelist 'file'! Failed: cannot open input. avformat_open_input() fail: Invalid data found when processing input Normally if I were using ffplay on the console, I would add the option -protocol_whitelist file,udp,rtp and it would work fine. So I tried this: AVDictionary *d = NULL; av_dict_set(&d, "protocol_whitelist", "file, udp, rtp", 0); ret = avformat_open_input(&inFormatCtx, filename, NULL, &d); But the same message still pops up. Any ideas? This

I am streaming an mp4(mpeg-4) file from one device to another using gstreamer over RTP stream. Basically I am splitting up the mp4 file into its audio and video file and then sending it all to the other device where it gets streamed. Now, I want to save the mp4 file to disk in the other device, but my problem is that I am able to save the audio and video files seperately and it cannot be played individually. I am confused on how to combine both the audio and video rtp streams to form my mp4 file back and save it to a file in the other device. Here are the command line codes : Sender(Server)

参考官方文档： http://www.rosoo.net/Files/UpFiles/RsProduct/avtools/2009-4/2009491562537854.txt 聚合包的RTP荷载格式的结构见图3 即开头1个字节表示NAL单元类型，接着1个或多个聚合单元，后面是可选的RTP填充。 单时间聚合包STAP 单时刻聚合包(STAP)应该用于当聚合在一起的NAL单元共享相同的NALU时刻。STAP-A荷载不包括DON，至少包含一个单时刻聚合单元见图4 STAP-B荷载包含一个16位的无符号解码顺序号(DON) (网络字节序)紧跟至少一个单时刻聚合单元。见图5. 图 7表示一个例子--一个RTP包包含一个STAP-A. STAP包含两个单时刻聚合单元, 在图中用1，2标记。 开始是4个字节的RTP头，接着1字节NAL单元类型，应该是24，然后2字节表示第1个NALU长度，接着1字节表示第1个NALU类型，然后若干个字节的NALU负载数据，以此类推。 图 8 表示一个RTP包包含一个STAP-B. STAP包含两个单时刻聚合单元, 用 1，2标记。 STAP-B比STAP-A多了一个公共的DON，详见规范文档 分片单元相对来说比聚合单元更简单一点 来源： https://www.cnblogs.com/jixiaohua/p/11899440.html