[SR-Users] WebRTC video call terminated after ~30 seconds

[Mathias Brodala - Pagemachine AG]

Hi,

I am currently testing a setup to do video calls via WebRTC and have set up
Kamailio as SIP server.
So far the calls are working fine, but after around 30 seconds the calls
are terminated.

I can reproduce this easily both with the JsSIP demo [0] as well as the
SIPjs demo [1].

The browser console using JsSIP log [2] shows me this:

> JsSIP:WebSocketInterface WebSocket wss://sip.example.org closed +60s
> jssip-3.2.10.js:23490 JsSIP:WebSocketInterface WebSocket abrupt
disconnection +1ms

Please also have a look at my kamailio.cfg [3]. I tried to follow the docs
of the websocket module [4]
as close as possible to get everything up and running.

Kamailio is currently running on a private DigitalOcean droplet and I
already tried to disable all
firewall rules without change. I also asked the DO staff if something could
lead to UDP packets
getting dropped after some time but I was told that there is no such logic
in the DO network.

Please let me know if you can find anything unusual which could cause this
behavior or if you
need more information.

Thank you,
Mathias Brodala

[0] https://tryit.jssip.net/
[1] https://sipjs.com/demo-phone/
[2] https://gist.github.com/mbrodala/0f81aaa174be87096cf20502633f3ac3
[3] https://gist.github.com/mbrodala/066faa455da46362dab3dd5326ba8ad1
[4] https://www.kamailio.org/docs/modules/5.1.x/modules/websocket.html
-- 

Dipl.-Inf. (FH) Mathias Brodala
Tel.: 069.260.99.70.45 Fax: .31
mbrodala at pagemachine.de

<https://www.pagemachine.de/>


Pagemachine AG
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