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@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@89.x.x.8:26380>.
To: "mobilia"<sip:00390721111111@pstnprovider.com>.
From: "Nick Warr - Mobilia"<sip:nick@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.