[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Jeff Pyle]

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" <ibc at aliax.net> 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.