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


--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>


From: sr-users <[email protected]> On Behalf Of Kelvin 
Ezenwaka
Sent: Friday, May 13, 2022 12:46 PM
To: users <[email protected]>; sr-users-owner 
<[email protected]>
Cc: [email protected]; Ope Shokunbi <[email protected]>
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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