I would disable TLS for debugging, and use TCP. If TCP works, TLS should
work too (in the context of NAT traversal) and it is much more easier to
debug.
good luck
klaus
On 30.01.2014 10:24, John Smith wrote:
Hello Klaus,
I have been making some tests just to be sure the network is not the problem.
While using a simpler config file for Kamailio with just the TLS and NAT which
does not involve Asterisk in the scenario, calls work properly between
softphones. OTOH, using the template provided in the KB referenced in my first
post disabling the Asterisk define, calls are connected properly but no audio
flows between phones.
Now I am in the process of trying to locate where is the problem by comparing
how both files handle the NAT support.
Thank you
----- Original Message -----
From: Klaus Darilion
Sent: 01/23/14 08:12 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio behind NAT
On 23.01.2014 10:29, John Smith wrote:
Hello Klaus,
I had already two sockets bound each to two independent physical interfaces. I
have added the force_send_socket at each rtpproxy
Just for clarity:
force_send_socket is for near_end NAT traversal of the SIP signaling,
whereas manage_rtpproxy() is for the NAT traversal (near end and far
end) of the RTP stream.
It is necessary to use the cwie / cwei flags in the rtpproxy_manage call?
If rtpproxy uses only a single listen-IP, then these flags are not
needed. Only if you operate rtpproxy in bridge mode, then you need these
flags. Bridge mode is necessary if you do not have IP routing between
the internal network and the "virtual external" network, or if you want
to bridge between IPv4 and IPv6.
Currently audio does not flow back to the softphones, it gets lost at Kamailio.
Actually it should get lost at rtpproxy.
Please post a SIP trace: ngrep -Wbyline -q -t -P "" port 5060
and post the setup (external + internal IP addresses) (you can send it
privately to me or mangle the IP addresses if they are sensitive)
regards
Klaus
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