problem with playing of RTSP stream

[Arsen G.]

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]

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:

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.

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

 

[Arsen G.]

Dear Max,
do you have any update on our case?
Thank you for assistance.

 

[Max]

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

 

[Max]

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

 

[Max]

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

rtp_in_reset_marker=true

You have to update to the latest 2647 build to apply changes.

service webcallserver update 2647