El Sábado, 11 de Julio de 2009, Jeff Pyle escribió: > Iñaki, > > The PSTN gateway must support in-call reinvites because it sends its RTP to > the Mediaproxy after Asterisk sends its reinvite. Here's a sample of the > RTP from the perspective of the Mediaproxy relay (an obfuscated tshark > output): > > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16448 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16452 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16452 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16452 > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16452 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > PSTN_gateway -> Mediaproxy UDP src port: 14618 dst port: 16452 > SIP_phone -> Mediaproxy UDP src port: 16892 dst port: 16454 > Mediaproxy -> Asterisk UDP src port: 16452 dst port: 17276 > > Looking at the above capture, we can see that both the PSTN gateway and the > SIP phone are sending their RTP to the Mediaproxy. But, the Mediaproxy > relays the SIP phone's packets to Asterisk, which still has the socket open > to relay them to the PSTN gateway. That's why the SIP phone can be heard > on the PSTN, the but the PSTN phone cannot be heard on the SIP phone. > > The only difference I can see between an inbound call and an outbound call > from a media perspective is that in inbound has no pre-connect media (180 > w/o SDP) while an outbound call has media (183 w/ SDP). MIght that be > relevant?
It shouldn't. At this point, a SIP trace (ngrep) would be very useful. -- Iñaki Baz Castillo <i...@aliax.net> _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users