It will be helpful if you can provide a pcap with the network capture of sip packets for such call taken on kamailio server. Then we can see if rtp relaying was engaged or not.

Cheers,
Daniel


On 04/08/16 04:55, Kotb, Amir wrote:

[03:50] <meamo> I have a small query, can anybody assist please?
[03:51] <meamo> I am running kamailio + rtpengine on a google compute cloud instance
[03:51] <meamo> have setup them to work with devices behind nat, and everything works
[03:52] <meamo> no i have added freeswitch. but I don't know how to configure rtpengine to work with both
[03:52] <meamo> when I use #!define with_nat and #!define with_freeswitch, I get no voice in the calls. When I remove, #!define with_freeswitch, everything works normally

Best regards,
Amir

--
Amir KOTB, Msc., Bsc.

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Department of Electrical Engineering & Electronics,
The University of Liverpool,
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Liverpool L69 3GJ,
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