Possibility to switch to other then UDP transport protocol in diplicate_uri
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#### Description
Sometimes it's preferable to do the duplication over TCP or another transport, this little fix removes the hard-coded UDP from the diplicate_uri module parameter.
You can view, comment on, or merge this pull request online at:
https://github.com/kamailio/kamailio/pull/3108
-- Commit Summary --
* siptrace: fix hardcoded PROTO_UDP in duplicate_uri module parameter
-- File Changes --
M src/modules/siptrace/siptrace_send.c (2)
-- Patch Links --
https://github.com/kamailio/kamailio/pull/3108.patchhttps://github.com/kamailio/kamailio/pull/3108.diff
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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+rtpengine---->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.
![MrOpee](https://user-images.githubusercontent.com/105465203/168183786-c10a822b-402b-4666-be10-b2bc5ad98e85.png)
[ServersideWebrtc-external2.txt](https://github.com/kamailio/kamailio/files/…
[Clientside(webrtc2external)__III.txt](https://github.com/kamailio/kamailio/…
Thank you.
@rfuchs @doublec @ibc @fredposner @linuxmaniac @miconda
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