Hi Dave,
if you already checked the signalling (the ip ans ports in SDP) and
everything is fine (all properly mangled for nat), you should make a
network trace to see the flow of RTP streams - how ans where is sending RTP.
regards,
bogdan
Dave wrote:
I had posted the following a few days ago and received
no response.
Can someone please help!
Hello,I have the following setup:
UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW
|
IVR
Nat'd user calls a number that is forwarded (t_relay)
to Prepaid and the user is prompted for
PIN#/Destination Phone# and all works fine. Mediaproxy
is invoked and a mediaproxy port# (35774) is assigned
during the whole process.
Once the user enters the destination number, the
Prepaid forwards the call to the PSTNGW and the PSTN
phone rings (Call signalling working fine). When the
PSTN phone is answered, no media can be heard!!!
I did some investigation and here is what is
happening:
Once the user enters the destination# to dial, The
Prepaid/B2BUA does two things:
1) Sends a re-invite to put the UA on hold.
2) Sends an INVITE to the PSTN GW and retrieves the
SDP from the response from PSTN GW
3) Sends a second-reinvite to the UA via SER with the
SDP info of the PSTNGW
4) SER invokes the mediaproxy (since it is reinvite)
and assigns the SAME MEDIAPROXY PORT# as earlier on
when the media was flowing fine (SDP has audio port#
35774)
5) SER forwards the re-INVITE TO UA
6) UA responds with a 200 OK and sends 200 OK to SER.
SER agains invokes mediaproxy and assigns the SAME
MEDIAPROXY PORT# (audio port 35774) and forwards the
response to Prepaid (which sends it to PSTNGW).
I checked on Cisco PSTNGW that it is creating a
session with the mediaproxy (port:35774). BUT NO AUDIO
CAN BE HEARD IN ANY DIRECTION.
Can anyone please help? Is this a mediaproxy issue
that when reinvites are sent and mediaproxy is invoked
multiple times, issues arise?
I am running the latest mediaproxy
version.
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