problem with playing of RTSP stream

Arsen G.

Member
Dear Support team,
we faced with a problem with playing of RTSP stream in webRTC/websocket player
on our flashphoner dashboard and on your demo website also.
Could you please check and assist what's going wrong?

RTSP stream I'll send via e-mail to support@flashphoner.com

Thank you in advance.
Regards,
Arsen.
 

Max

Administrator
Staff member
Hello

We have reproduced this issue with your stream and server build 2631.
I will inform you once we have any news.

Stream looks good according ffprobe:
Code:
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: h264 (Main), yuvj420p(pc, bt709, progressive), 1280x720, 29.97 tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: aac (LC), 48000 Hz, stereo, fltp
Possible difference is order of audio and video tracks, as you can see, audio track is 0:0 and video track is 0:1 for working stream. We will check that.
Code:
ffprobe rtsp://str81.creacast.com/grandlilletv/low
Duration: N/A, start: 0.128000, bitrate: N/A
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp
Stream #0:1: Video: h264 (Constrained Baseline), yuv420p(progressive), 640x360 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc
 

Max

Administrator
Staff member
Hello,
No updates yet. The issue is in our backlog.
We will inform once we have any updates. It may take a time.
 

Max

Administrator
Staff member
Hello
We plan to take the issue in progress.
Please do not close RTSP stream for further testing.
 

Max

Administrator
Staff member
Hello
The root cause of issue was incorrect RTSP/RTP stream from the RTSP source.
Each RTP interleaved packet was marked (had marker bit).
We have fixed this behavior adding custom setting in WCS_HOME/conf/flashphoner.properties
Code:
rtp_in_reset_marker=true
You have to update to the latest 2647 build to apply changes.
Code:
service webcallserver update 2647
 
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