[Users] Problems with one sided audio..

[nick]

nick wrote:
> I have a situation where I make a call to a pstn provider, everything 
> works correctly on the SIP level, INVITEs, OKs, ACKs and BYEs are all 
> passing correctly through my server, but for some reason, I'm unable to 
> get audio to work correctly on both sides.
> 
> At the moment I'm using Openser 1.1.0 with mediaproxy (and a slightly 
> modified openser.cfg based on the mediaproxy one on the openser site, I 
> have a some options for accounting and forwarding to the pstn gateway)..
> 
> 89.x.x.16 is my openser server.
> 
> 89.x.x.8 is my NAT firewall, which is portforwarding all UDP from 5000 
> to 30000 to 192.168.1.67 (my internal machine, with X-Lite).
> 
> x.x.x.53 is my PSTN provider's SIP server
> 
> x.x.x.3 is my PSTN provider's media server.
> 
> this is the SIP dialog:

> U 89.x.x.16:5060 -> x.x.x.53:5060
> INVITE sip:00390721111111 at x.x.x.53:5060 SIP/2.0.
> Record-Route: <sip:89.x.x.16;lr=on;ftag=173a892a>.
> Via: SIP/2.0/UDP 89.x.x.16;branch=z9hG4bK860f.0d8bc646.0.
> Via: SIP/2.0/UDP 
> 192.168.1.67:26380;received=89.x.x.8;branch=z9hG4bK-d87543-cb65401207770316-1--d87543-;rport=26380. 
> 
> Max-Forwards: 69.
> Contact: <sip:nick at 89.x.x.8:26380>.
> To: "mobilia"<sip:00390721111111 at pstnprovider.com>.
> From: "Nick Warr - Mobilia"<sip:nick at logycs.it>;tag=173a892a.
> Call-ID: N2YyNGJmMjBhZjYwMjY3OWExMmVmYzYyNDhjMTgzNzY..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> User-Agent: X-Lite release 1006e stamp 34025.
> Content-Length: 261.
> P-hint: outbound.
> .
> v=0.



> o=- 1 2 IN IP4 192.168.1.67.
> s=CounterPath X-Lite 3.0.
> c=IN IP4 192.168.1.67.

^^^^^^^^^^^^^^^^^^^^^^^^^^

This is my problem, right here.


I've found the correct way to correct the problem, fix_nated_spd();
but I'm not sure where in my routing logic I need to put it..

If needed I can send my openser.cfg, I just need to be able to fix the 
SDP NATing.


> t=0 0.
> m=audio 13234 RTP/AVP 0 98 3 101.
> a=alt:1 1 : AtyyaMHs +WwsY5o+ 192.168.1.67 13234.
> a=fmtp:101 0-15.
> a=rtpmap:98 iLBC/8000.
> a=rtpmap:101 telephone-event/8000.
> a=sendrecv.