Hi.
I have a case when a peer answering a call (INVITE) is behind a NAT.
So, in its SIP OK message I would like to see the SDP containing a valid IP and
media Port valid to receive audio from the caller. That is, the caller need to
know a valid IP and Port where he/she can send his/her audio packets.
1 - Is it possible to "fix" SDP content for such objective?
2 - Can OpenSIPS do something for this idea works or must I to use something
more like a stun server?
3 - What is the OpenSIPS module that can help me with this task?
I guess I will have to fix 2 fiels in SDP:
Media Description, name and address (m): audio 55142 RTP/AVP 8 101 ( to
fix the port)
Connection Information (c): IN IP4 192.168.100.156
( to fix the IP)
P.S.: I have already a solution (opensips.cfg) that let SIP messages cross NATs
without problems. Only SDP has to be fixed.
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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