websocket is working, webrtc not

Arsen G.

Member
Dear Max,
we faced with a problem - our stream is working using websocket, but via webRTC is not working :(
Could you please suggest what is a problem?
Thank you in advance.
 

Max

Administrator
Staff member
Hello,

Could you please clarify the following details?
- The issue is that a stream cannot be played as WebRTC (though can be played as WebSocket)? Does playback fail, or there’s no video, or something else?
- What browser or mobile application is used for playing as WebRTC?
- What type of stream (WebRTC, RTMP, RTMFP, RTSP) – i.e., how the stream is published?
- Can the issue be reproduced with the demo server ( https://wcs5-eu.flashphoner.com/dashboard.xhtml )?
 

Arsen G.

Member
Hi Max!

1. the stream cannot be played as WebRTC, but can be played as WebSocket. The player establishing connection and failed about 1 minute later.
2. I'm using Google Chrome for playing as WebRTC
3. We are using RTSP stream
4. on your demo site we have the same result - WebSocket is OK, but WebRTC it keeps trying to play.
Please also find attached our flashphoner.properties file.
 

Attachments

Max

Administrator
Staff member
Generally we do not check this email.
Please use logs@flashphoner.com for any emails related the forum.
We found sent RTSP URL.
It does not play in VLC for me.
Please check on your end.
This is result of ffprobe
Code:
ffprobe rtsp://host:1935/stream
Code:
ffprobe version 3.2.3 Copyright (c) 2007-2017 the FFmpeg developers
  built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-11)
  configuration: --enable-shared --disable-logging --enable-gpl --enable-pthread                                                                                                                                                             s --enable-libx264 --enable-librtmp --disable-yasm
  libavutil      55. 34.101 / 55. 34.101
  libavcodec     57. 64.101 / 57. 64.101
  libavformat    57. 56.101 / 57. 56.101
  libavdevice    57.  1.100 / 57.  1.100
  libavfilter     6. 65.100 /  6. 65.100
  libswscale      4.  2.100 /  4.  2.100
  libswresample   2.  3.100 /  2.  3.100
  libpostproc    54.  1.100 / 54.  1.100
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 91 packets
[h264 @ 0x1ce2540] error while decoding MB 75 12, bytestream -37
[h264 @ 0x1ce2540] concealing 2614 DC, 2614 AC, 2614 MV errors in I frame
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 5 packets
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 11 packets
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 2 packets
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 43 packets
[rtsp @ 0x1cdbbc0] max delay reached. need to consume packet
[rtsp @ 0x1cdbbc0] RTP: missed 6 packets
Input #0, rtsp, from 'rtsp://43.251.71.12:1935/livedealer/testBBIN-vpc2.stream':
  Metadata:
    title           : testBBIN-vpc2.stream
  Duration: N/A, start: 0.341000, bitrate: N/A
    Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp
    Stream #0:1: Video: h264 (High), yuvj420p(pc, bt709, progressive), 1280x720, 29.97 tbr, 90k tbn, 180k tbc
It claims missed RTP packets.
Looks like your RTSP server does not send RTP packets properly.
 

Arsen G.

Member
Dear Max,
I checked at my side and I can play the stream with VLC. Could you please check at your side once more.
Also I do check with ffprobe. Please find the file attached.
 

Attachments

Max

Administrator
Staff member
The stream has resolution 1280x720, video doesn't appear because there is not enough bandwidth.
It can be played on WCS as WebRTC if specify video constraints for the stream to decrease playback resolution (checked with the latest v. 2429).

var constraints = {
audio: true,
video: {
width: 640,
height: 480
}
};
session.createStream({
name: streamName,
display: remoteVideo,
constraints: constraints
})...
}).play();


As in Media Devices example: https://github.com/flashphoner/flas...reaming/media_devices_manager/manager.js#L333
 

Arsen G.

Member
Dear Max,
I did upgrade Webcallserver to version 5.0.2429
I tried to reduce playback resolution but again no luck (the same stream is working great on another Flashphoner server).
I think the problem is not bandwidth :(
I would like to mention that on the same server we have "Flashphoner" and "Wowza Media Server" installed.
Could it cause such problem with WebRTC playback? Any ideas?
 

Max

Administrator
Staff member
Can the issue be reproduced when Wowza is not running? I.e., stop WCS, stop Wowza and other services which could use the port range, and start WCS.
 

Arsen G.

Member
Can the issue be reproduced when Wowza is not running? I.e., stop WCS, stop Wowza and other services which could use the port range, and start WCS.
Hello Max,
I did as you suggest but no luck. I stopped Wowza and WCS, then started WCS and tried to play the RTSP stream - via Websocket it was OK, but via WebRTC its failed :(
 

Arsen G.

Member
Dear Max,
sorry for delay with answer. It's difficult to give you access to server at this moment.
Maybe I can send you log files via e-mail? Please provide log file names you need.
Thank you in advance.
 

Max

Administrator
Staff member
1. Run tcpdump
tcpdump -s 4096 -w log.pcap
2. Make a test
3. Press CTRL+C to stop tcpdump.
4. Gather following files
  • WCS_HOME/logs/flashphoner_manager.log
  • WCS_HOME/logs/server_logs/flashphoner.log
  • WCS_HOME/conf
  • log.pcap
5. Run report tool.
cd WCS_HOME/tools
./report.sh
and find report file
Example: WCS_HOME/logs/reports/report-2017-09-07-09-41-30.tar.gz

Please send us:
  • WCS_HOME/logs/flashphoner_manager.log
  • WCS_HOME/logs/server_logs/flashphoner.log
  • WCS_HOME/conf
  • log.pcap
  • report...tar.gz
to logs@flashphoner.com
 

Arsen G.

Member
Hi Max. Sorry for delay with answer. You were right, the problem was in UDP port blocking.
Now everything is working fine.
Thank you for your assistance.
 

Max

Administrator
Staff member
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