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(a)liverpool.ac.uk_
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