Actually I was exploring other features of sip communicator and it deserves more kind words , For The log files I attached them to this email so that can be analyzed to know the problem.
Also I tested other Softphones; X-lite and ExpressTalk are successful only when "RealTunnel NAT Traversal" is used as Outbound Proxy ( localhost:5061).
The only software that I found working out of the box is Emansip (from Antisip.com) with TLS as Trasport Protocol (Also tested outgoing calls, incoming calss OK), I also attached a Wireshark dump for it in case it might be handy.
Also I have 2 Windows Servers connected to Public network if there be anything necessary (proxy, tunnel, SIP proxy, RTP proxy) to be tested I can be a help.
sip-communicator0.log.txt (34.2 KB)
wireshark-capture-dump-emansip.pcap (22.7 KB)
wireshark-capture-dump-sip-communicator.pcap (23.8 KB)
From: Emil Ivov <firstname.lastname@example.org>
Cc: michael lopez <email@example.com>
Sent: Tue, 18 May, 2010 15:53:45
Subject: Re: [sip-comm-dev] Does Sip Communicator support Symmetric NAT?
На 18.05.10 12:36, michael lopez написа:
And thanks for the great program just checked it out and it has shiny
Thank you for your kind words
I just checked the SIP feature of the program and noticed that it just
can't register the UA when he is behind Symmetric NAT
Symmetric NATs, or rather NATs with endpoint dependent mapping, are
generally a problem for media and not signalling itself. Therefore, if
you can't register then there's probably another reason. Could you
please send log files and wireshark dumps so that we could try to see
, So question in my
mind has TURN or Tunneling or other feature that can do the job being
implemented in SIP communicator yet?
Currently SIP Communicator relies on the servers to handle NAT
traversal. Most of the popular deployments and server implementations
would overwrite your SDP and then relay media.
We are currently working on an ICE implementation within the ice4j 
project. TURN is one of the features that will be implemented in ice4j
but that's probably not going to affect you since there are currently
very few SIP providers that are maintaining TURN server deployments.
In addition just to note I tried to to do NAT Traverse using RealTunnel
NAT Traverse but it didn't work either ( It works in X-lite , Express Talk)
Never really tried it. Would probably need to look at dumps to see what
may be going wrong.
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