RTMP error while doing streaming via webrtc

[jasonkc]

Hi,

At times, I'm seeing below errors in server log indicating RTMP errors (log attached) when I realized the stream failed; would appreciate if you could suggest way to rectify it.

 

[Max]

Good day.
Seems like this a normal behaviour when you publish WebRTC stream, then republish it locally as RTMP. In this case, WCS works as RTMP publishing client and server. When the source WebRTC stream is stopped, RTMP client is also stopped. At RTMP servers point, it looks like channel is closed accidentally, so it prints error to the server log.
You can ignore those errors.

 

[jasonkc]

I'm seeing this even i did not stop the stream from the web client, which worries me.

 

[Max]

I'm seeing this even i did not stop the stream from the web client, which worries me.

According to your logs, all the errors are displayed after publiher stops the stream. If there is another errors, please collect a report as described here and send using this form.

 

[jasonkc]

Now i can't start any stream any all, is there a limitation on number of sessions I can run with trial license?

 

[jasonkc]

also thats the range of UDP port RTMP would use?

 

[jasonkc]

also whats the range of UDP port RTMP would use?

 

[Max]

>Now i can't start any stream any all, is there a limitation on number of sessions I can run with trial license?

There are no such limitations.
All limitations listed here

Perhaps your server has not enough CPU or RAM. Or Java 11 issue happened.
Please provide SSH access. We will check.

Try

service webcallserver restart

to restart

>also thats the range of UDP port RTMP would use?

RTMP server port 1935 TCP
RTMP client ports 33000-34000 TCP
WebRTC ports 31000-32000 UDP

Architecture - Web Call Server 5.2 - EN - Flashphoner Documentation

docs.flashphoner.com

RTMP is TCP based protocol. No UDP ports used for RTMP.

 

[jasonkc]

I have tried restarting webcallserver and even reboot the server, but to no avail.

You may ssh to my server, 54.255.9.27, as user centos using attached private key (you need to rename the file to .ppk).

 

[jasonkc]

I'm seeing this error in browser developer tool

 

[jasonkc]

You may hold on from accessing our server, I think I found the root cause....need more time to resolve it; will ping again if I need help.

 

[jasonkc]

when i tried streaming using demo codes ( webrtc-as-rtmp-republishing.js and webrtc-as-rtmp-republishing.html), I could see the stream created successfully (refer to below screen shot); however when i did it from our own streaming dashboard (adopted most of the codes from demo code), the stream stopped abruptly (refer to dashboard screen shot), wonder what could be the issue?

 

[jasonkc]

ignore above, i realized it does not matter of the output of console log in developer tool; in Chrome, though I dont see indication of Webrtc session started, the stream was still successful

 

[Max]

ignore above, i realized it does not matter of the output of console log in developer tool; in Chrome, though I dont see indication of Webrtc session started, the stream was still successful

Please clarify: did you solve the problem? Is WebRTC as RTMP republished successfully?
Plaese also note that you should send any private information (logs, SSH credentials, urls etc) using the special form by clicking "Report" button at the top of topic page. It is not recommended to attach SSH credentials to forum post due to possible security issues.

 

[jasonkc]

noted...the issue is still not totally resolved; we realized if we triggered the RTMP stream with short stream ID from our platform, it will fail. Is there a limitation of the length of RTMP stream ID we should adhere to?

 

[Max]

Is there a limitation of the length of RTMP stream ID we should adhere to?

No. there is no limits.
But, if you're republishing stream to the same WCS server (localhost), you should provide an unique stream key or use the setting

rtmp_transponder_stream_name_prefix=rtmp_

If you're republishing stream to third party server, all the stream key limits are applied according to third party RTMP server requirements (Facebook, Youtube etc)

 

[jasonkc]

issue resolved....thanks!