Hello,
one cause for this 30s call ending could be that the ACK for the 200OK on the INVITE is not routed correctly. Have a look to the signalisation of the clients and/or server.
One way audio can be caused from wrong SDP, check that the media IPs in the SDB are correct.
Cheers,
Henning
From: sr-users <sr-users-bounces@lists.kamailio.org>
On Behalf Of Kelvin Ezenwaka
Sent: Friday, May 13, 2022 12:46 PM
To: users <users@lists.kamailio.org>; sr-users-owner <sr-users-owner@lists.kamailio.org>
Cc: ope@tosoplimited.com; Ope Shokunbi <ope@pressone.co>
Subject: [SR-Users] Failing Audio stream on Kamailio+webrtc with integration to asterisk server
Summary:
We have a Kamailio server as our sip proxy server, sip firewall with WebSocket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang
up after 30s, when we try making calls between internal extensions (eg extension 100 to call extension 105) and external calls (eg extension 100 to call mobile number 09056925668) as well. The call flow is as:
Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco Provider
The asterisk communication between all three boxes is via local IP (All ports open between them). The Kamailio 1 box where we have our sip registration cache, has rtp ports and wss
port open on the internet.
Asterisk 18.9.0
Kamailio 5.5.3 + RTPengine 10.4.0.0
Debian 11 bullseye
Kindly see attached below a diagram depicting the VoIP network flow and also attached are logs files for the webrtc client and server side.
Ezenwaka Kelvin Arinze
VOIP Engineer
Tel: (+234) 9056925668
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