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@pagemachine.de

  

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