具有gstreamer-1.0的H264 RTP流 [英] H264 RTP stream with gstreamer-1.0

问题描述

我尝试使用带有相机模块的Raspberry Pi 3制作H264 RTP流到视频标签。

I try to make a H264 RTP stream from a Raspberry Pi 3 with a camera module to a video tag.

使用以下代码启动流

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

然后我提供一个带有视频标签的简单网页：

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>

src引用以下SDP文件：

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

当我加载网页时没有任何反应，并且js控制台完全是空的。

When I load the webpage nothing happens, and the js console is completely empty.

所以我尝试用VLC查看流，并收到以下错误：

So I tried to view the stream with VLC, and got the following error:

[00007efd80c03ea8] es demux error: cannot peek
[00007efd80c03ea8] es demux error: cannot peek
[00007efd80c03ea8] live555 demux error: no data received in 10s, aborting

我认为根本没有UDP通信，所以我从远程机器上测试了它：

I thought that there had been no UDP communication at all, So I tested it from a remote machine:

gst-launch-1.0 udpsrc port=5000 \
               caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" \
               ! fakesink dump=true

ans收到UDP数据包。所以我研究了前方，发现了这个：

ans received UDP packets. So I researched forward and found this:

https://cardinalpeak.com/blog/the-many-ways-to-stream-video-using-rtp-and-rtsp/

现在很明显，我需要2个端口来传输数据并建立RTP控制协议。但是我不知道如何用gstreamer做到这一点。

Now it is clear that I need 2 ports one to stream data and to establish RTP Control Protocol. However I have no idea how can I do it with gstreamer.

我跑的时候最糟糕的是：

Worst of all when I run:

gst-inspect-1.0 | grep -i rtcp

我一无所获。

如何使用RTP协议将gstreamer-1.0的视频流启动到网页内的视频标签？

How can start video stream with gstreamer-1.0 to a video tag inside a webpage using the RTP protocol?

更新

使用 videotestsrc 作为gstreamer videosoruce并删除 gdppay （它导致无效的RTP有效载荷错误）我能够查看来自的视频流具有VLC和此gstreamer代码的远程客户端：

Using videotestsrc as gstreamer videosoruce and removing gdppay (it caused invalid RTP payload error) I was able to view the video stream from a remote client with VLC and with this gstreamer code:

gst-launch-1.0 udpsrc port=5000 \
               caps="application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" \
               ! rtph264depay \
               ! avdec_h264 \
               ! autovideosink

推荐答案

首先你需要提供更多信息：哪个浏览器（试试chrome，也称safari有更好的流媒体支持）..

First you need to provide more info: which browser (try chrome, also safari is said to have better streaming support)..

对于SDP我觉得你错过了它的h264的信息？
和是gdppay仅适用于内部GStreamer only流媒体（gdp意为GStreamer数据协议，其他人无法理解:)）。

To the SDP I think you are missing the info that its h264? And yes gdppay is just for internal "GStreamer only" streaming (gdp means "GStreamer Data Protocol" which noone else understands:) ).

如果你真的希望GStreamer流式RTSP您可以使用gstreamer rtsp服务器实施 - 这在单独的仓库中，例如在Ubuntu的一些软件包中包含。

If you really want GStreamer to stream RTSP you may use gstreamer rtsp server implementation - this is in separate repo and is incuded in some packages in Ubuntu for example.

如果你只想要rtp你正确地做到了 - 正如你看到这种方法适用于例如vlc ..但是是什么让你认为sdp将在HTML5中运行（我只是问我没有最新的信息）？

If you want just rtp you are doing it correctly - as you see this approach works with for example vlc.. but what makes you think the sdp will work in HTML5 (I am just asking I do not have up to date infos on this)?

您也可以使用netcat对此进行测试 - 这对于这类调试来说很好。
你可以这样假冒rtp客户：

You can test this also with netcat - its fine for these kind of debugging. you can fake a rtp client this way:

nc -u -l 5000 

哪会将流量转出。

我读

I read here that there are problems with rtp/rtsp in HTML5, but who knows maybe now it is already working..

您可以尝试使用hls - 这通常用于流式传输，并且在最近的1.6和更多版本中对GStreamer有更好的支持..（提示：使用hlssink）。

You can try hls - which is usually used for streaming and has much better support int GStreamer these days 1.6 and further versions.. (hint: use hlssink).

这里有一些处理hls的js：
https://github.com/dailymotion/hls.js

Here you have some js for processing hls: https://github.com/dailymotion/hls.js

您还可以尝试ogg / vorbis / theora等等（声音）疯了，但是你可以试一试，我读到它适合流媒体的地方）..

You can also try ogg/vorbis/theora and such stuff (sounds crazy, but you can give it a shot, I read somewhere that its suitable for streaming)..

这篇关于具有gstreamer-1.0的H264 RTP流的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持IT屋！