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.
Postgraduate Researcher Department of Electrical Engineering & Electronics, *The University of Liverpool*, Brownlow Hill, Liverpool L69 3GJ, UK
Mobile: _+44-(0) 7428844234_ Email: _A.Kotb@liverpool.ac.uk_ Skype: _A.kotb1_ Web: _Http://uk.linkedin.com/in/AOKotb_
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