Desperately looking for a RTSP server that can stream from a live source (not from a file)

Question

I need a RTSP-server that can listen on a configured port (8554 for example), and then, for example, if I run FFmpeg with:

ffmpeg -f v4l2 -i /dev/video0 -c:v libx264 -intra -an -f rtsp -rtsp_transport tcp rtsp://192.168.1.10:8554/test

Then the RTSP-server will RECORD the video, and to play it, I just need to run it with:

ffplay -i rtsp://192.168.1.10:8554/test

I need the RTSP-server to support TCP transport and H264 video encoder and OPUS audio encoder and stream from a live-video (not from a file) + the program should be unlicensed.

• This server works great, but don't support OPUS.

• Live555 support H264 and OPUS, but only streams from files (VOD).

• I've have found some other servers that can stream directly from /dev/video0, but it's also not a good solution for me.

• Wowza and Red5Pro does answer all the above requirements, except that they are licenced programs.

Any suggestions for a RTSP-server that support all the above requirements?

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EDIT:

I've tried Gstreamer and it looks promising, but I still didn't success. However, I'm quite sure I'm on the right way (perhaps I don't know how to use yet the pipelines).

1. I've built gst-rtsp-server, version 1.13.91.
2. Then, I ran ./test-record "( decodebin name=depay0 ! videoconvert ! rtspsink )"
3. I ran netstat -anp and I can see clearly, the server is listening on tcp port 8554.
4. Now it's time to stream to server. I've tried it once with Gstreamer and once with FFmpeg.

Gstreamer

gst-launch-1.0 videotestsrc ! x264enc ! rtspclientsink location=rtsp://127.0.0.1:8554/test

FFmpeg

ffmpeg -f v4l2 -video_size 640x480 -i /dev/video0 -c:v libx264 -qp 10 -an -f rtsp -rtsp_transport tcp rtsp://127.0.0.1:8554/test

In both cases, I can see the RTP packets in wireshark, and by calling again to netstat -anp, I see:

tcp        0      0 0.0.0.0:8554            0.0.0.0:*               LISTEN      14386/test-record  
tcp        0      0 127.0.0.1:8554          127.0.0.1:46754         ESTABLISHED 14386/test-record  
tcp        0      0 127.0.0.1:46754         127.0.0.1:8554          ESTABLISHED 19479/ffmpeg  

So I can surly understand that I'm streaming (or streaming something...). However, when I'm trying to play the video, I'm getting failure (I've tried to play with Gstreamer, FFplay and VLC - all fails...):

Gstreamer

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test latency=300 ! decodebin ! autovideoconvert ! autovideosink

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not get/set settings from/on resource.
Additional debug info:
gstrtspsrc.c(7507): gst_rtspsrc_retrieve_sdp (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Server can not provide an SDP.
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...

FFplay

ffplay -i rtsp://127.0.0.1:8554/test

[rtsp @ 0x7fb140000b80] method DESCRIBE failed: 405 Method Not Allowed
rtsp://127.0.0.1:8554/test: Server returned 4XX Client Error, but not one of 40{0,1,3,4}

VLC

vlc rtsp://127.0.0.1:8554/test

VLC media player 3.0.8 Vetinari (revision 3.0.8-0-gf350b6b)
[0000000000857f10] main libvlc: Running vlc with the default interface. Use 'cvlc' to use vlc without interface.
Qt: Session management error: None of the authentication protocols specified are supported
[00007f9fdc000ea0] live555 demux error: Failed to connect with rtsp://127.0.0.1:8554/test
[00007f9fdc001d10] satip stream error: Failed to setup RTSP session

Any ideas what I'm doing wrong ?

Answer 1

Wowza SE works with H264, Opus, VP8 as it supports WebRTC.

This plugin provides a turnkey setup for broadcasting channels live with WebRTC, RTMP, RTSP trough Wowza SE. Also can handle all stream types including RTSP with FFMPEG for on demand adaptive transcoding (in example between WebRTC & HLS). https://wordpress.org/plugins/videowhisper-live-streaming-integration/