New subject: Kamailio behind NAT

30 Jan 2014

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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