[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Iñaki Baz Castillo]

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 <ibc at aliax.net>