Daniel-Constantin Mierla wrote:
I am not using mediaproxy currently, so I do not know if you need to use functions from nathelper. However, you should call fix_nated_sdp() for the INVITE and 183/200 replies. Looking at next configs together might help:
http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy http://voip-info.org/wiki/view/OpenSER+And+RTPProxy
Cheers, Daniel
On 11/15/06 18:39, nick wrote:
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.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Thank you, that's the sort of info I needed!!