Hi All,
I have this scenario:
An user1, connected trought an UA1 (behind NAT), make a sip call to an
user2.
user2 is logged into the system on 2 UAs (one behind a NAT and the other
on a public IP Address).
user1 sent an INVITE message to the SIPProxy (SP) in wich there is an
SDP with 16442 RTP Audio Stream port
SP sent the INVITE to the 2 UAs in wich there in an SDP with 30506 RTP
Audio Stream port
The second UA sent to the SP a 200 OK with 10808 RTP Audio Stream port
The SP sent to user1 a 200 OK with 30508 RTP Audio Stream port
There is a problem, the media proxy sent the RTP Audio Proxy to the
second UA from the source port 30510 (and not from 30506), the
destination client is behind a NAT and the RTP Audio Stream isn't
established.
I'm in a multi-proxy environment. I can provide the wireshark trace.
Do you have any suggestions? There is already a bug fix for this problem?
Thanks in advance
MD
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