websocket is working, webrtc not

Arsen G. · Aug 31, 2017

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 · Aug 31, 2017

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. · Aug 31, 2017

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.

Max · Aug 31, 2017

Please provide RTSP URL of the stream.

Arsen G. · Sep 1, 2017

Max said:
Please provide RTSP URL of the stream.

Dear Max,
the message with URL has been send to forum@flashphoner.com yesterday.
thank you.

Max · Sep 1, 2017

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. · Sep 1, 2017

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.

Max · Sep 1, 2017

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. · Sep 1, 2017

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 · Sep 1, 2017

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. · Sep 4, 2017

Max said:
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

Max · Sep 4, 2017

Could you provide SSH access to the server and access to the WCS dashboard? If yes, please send to logs@flashphoner.com.

Arsen G. · Sep 7, 2017

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 · Sep 7, 2017

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. · Sep 7, 2017

Dear Max,
the log files has been sent to logs@flashphoner.com already

Max · Sep 7, 2017

We don't see any UDP packets in the log.pcap dump.
WebRTC works over UDP.
So it looks like UDP ports are blocked on your server.
Make sure UDP port range 31000-32000 (by default) is open.
Check availability of UDP ports using the following docs:
https://flashphoner.com/docs/wcs5/w...l?additional_tools-ports_routing_checking.htm

Arsen G. · Sep 29, 2017

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 · Sep 29, 2017

To avoid UDP blocking you can install TURN server on another host.
Then users will able to get streams behind firewall over port 443
https://flashphoner.com/docs/wcs5/w...nd_testing-firewall_traversal-turn_server.htm
Example:
https://wcs5-eu.flashphoner.com/cli...l-streaming/firewall-traversal-streaming.html
So if UDP is available, stream will go over UDP.
If not, it will fallback to TCP 443.
You have to launch your own TURN server for production needs.

websocket is working, webrtc not

Arsen G.

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Arsen G.

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