[Asterisk-Dev] Asterisk - sip communicator problem behind NAT

[Partha Sarathi]

Hello All,


I am using Asterik server with Jain sip communicator
as my user agents.It works fine without NAT ie when
NAT = no. But when behind the NATit registerd and
connected properly, but the media is not received on
either side.I had locked results in ethreal and
tcpdump.The result shows that
my user agents passes the RTP and H.261 protocols to
the server properly but the Asterisk is modifying the
sdp data sent by the sip-communicator. 
In it's  invite message the sip-communicator announces
that it would like to receive media on ports 22222 and
22224 and client two receives a different 
sdp announcement telling it to route all media through
Asterisk server.However, after that the proxy server
is not routing media to the ports originally 
announced by Sip Communicator but to the port that the
data was sent 
from 16682.


                        |  Stun server      |  
                        |-------------------|  
                               ^
 |         |       |           |
 | client1 |------->gateway1---|                   
 |---------| LAN 1 |           v
                             |                   |
                             |Asterisk sip server|
                             |-------------------|
 |         |        |          ^ 
 | client2 |------->gateway2---|                     
 |---------| LAN 2  |          | 
                               v
                         |  Stun server      |  
                         |-------------------|  


Stun server's
Primary   -stun01.sipphone.com
secondary -larry.gloo.net 

How can I fix this? Can anybody help me to solve this
problem?


thanks in advance
Partha   





	
		
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