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BenHoover
Joined: Nov 18, 2013
Messages: 60
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In my call recording application that I am writing, I am registering with a DMCC station and conferencing myself into an active call. However, no matter which RTP/RTCP ports I use to register with, the RTP stream gets sent to port 2048 and the RTCP stream gets set to port 2049. Is there something I'm missing or is there some configuration setting that needs to be changed somewhere?

Thanks,
Ben
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
Online
I have often registered phones in Client Media mode and the RTP has always been sent to the ports that I gave in the RegisterTerminal request.

If you follow the steps in the FAQ "How can I monitor the XML being sent and received by the AE Services Server (debug, log, trace)?", you can check that the RegisterTerminal request received by the AE Services contains the information that you think it should.

Martin
BenHoover
Joined: Nov 18, 2013
Messages: 60
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I confirmed via the logs and Wireshark that the registration request contains the correct ports. I am using TSAPI rather than DMCC to conference the registered DMCC station into a call, would that make a difference?

-Ben
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
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Check the MediaStart events to make sure they are using the correct ports.

Could you have a VPN, Firewall or NAT that might be messing with the messages?

Martin
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
Online
Also do a "status station" at the SAT prompt of the Communication Manager during the call. It produces a fairly obscure output but it should be able to show to which port it is sending the RTP.

Using TSAPI will have no effect.

Martin
BenHoover
Joined: Nov 18, 2013
Messages: 60
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How do I check the MediaStart events? I'm not sure about NATs or firewalls, this is at a remote customer location and I'm not sure about their network topology. The customer also hasn't given me access to the SAT terminal so I can't check that either.
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
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You can start a Media Monitor on the phone. Then, when the Communication Manager is about to start sending media to the phone, you will receive a MediaStart event. You will probably need this event as it contains the parameters that are used when sending the RTP eg. Codec, encryption parameters, RTP address & port, etc.

Martin
BenHoover
Joined: Nov 18, 2013
Messages: 60
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When I get a MediaStart event, the IP and port information for RTP/RTCP are for the source, which appears to be some sort of Avaya gateway. I'm looking to verify the destination port (the address is being set correctly since I can see the traffic at all).

-Ben
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
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My mistake.

In that case, you will need to get the output of status station to see where the media is being sent.

Martin
BenHoover
Joined: Nov 18, 2013
Messages: 60
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Ok, just to clarify, I want to get the station status for the DMCC station that I am registered with? Also, where is the RTP stream being sent from? Is it the AES server?

-Ben
MartinFlynn
Joined: Nov 30, 2009
Messages: 1922
Online
1. Yes
2. The RTP stream comes from the Communication Manager's Media Gateway. This is the IP Address in the MediaStart event.
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