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? - Jeff On 7/11/09 9:09 AM, "Iñaki Baz Castillo" <[email protected]> wrote: > Most probably, your PSTN gateway doesn't support/allow media address change > during a call, this is, it doesn't react when Asterisk sends it a re-INVITE > with a new media address in the SDP and the Gw remains using the first SDP. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
