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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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:
- Sends a re-invite to put the UA on hold.
- 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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I did. The RTP stream is flowing from UA To the MP/Port:35774 and from the PSTN GW to the MP/Port:35774. So theoretically MP shold be "proxying" the stream but it is not? What can I do? Thanks a lot. Dave
Bogdan-Andrei Iancu bogdan@voice-system.ro wrote: 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:
- Sends a re-invite to put the UA on hold.
- 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.
Have a question? Yahoo! Canada Answers. Go to *Yahoo! Canada Answers*
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if so, I cannot help you further ...I have no knowledge about MP :(
regards, bogdan
Dave wrote:
I did. The RTP stream is flowing from UA To the MP/Port:35774 and from the PSTN GW to the MP/Port:35774. So theoretically MP shold be "proxying" the stream but it is not? What can I do? Thanks a lot. Dave
*/Bogdan-Andrei Iancu bogdan@voice-system.ro/* wrote:
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. > > ------------------------------------------------------------------------ > Have a question? Yahoo! Canada Answers. Go to *Yahoo! Canada Answers* > > >------------------------------------------------------------------------ > >_______________________________________________ >Users mailing list >Users@openser.org >http://openser.org/cgi-bin/mailman/listinfo/users > >
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Bogdan-Andrei Iancu -
Dave