No RTP on SIP trunk call. 2

[Saiyan656]

Happy Friday!!

So I set up a SIP trunk to an 3rd party application. I was able to successfully call the SIP trunk from Avaya phone to 3rd part application via a SIP trunk connection with RDP audio successful in both directions. Now if I call from 3rd party application to Avaya via Session manager> SMGR > CM > H323 set, I get the call on the correct trunk and establish connection via TAC, but with no Audio, I verified on the 3rd party application that RTP is good sending traffic via PCAP. Now if I do a trace -w in SM I only see SIP protocol, should I be seeing RTP in PCAP? How do I review RTP traffic is coming to SM and if it is not what should I look for to resolve RTP? The application has been set to allow all ports on firewall for testing purposes, but has no change in outcome.

Thanks in advance for the help.

 

[kyle555]

SM doesn't handle RTP. It can simulate where RTP would flow in a trace based on what the SIP messages say, but no media ever flows to Session Manager. It will move packets between things, and those packets have SDP that are intended to indicate to the other party where it should send its audio.

So, Phone A calls Phone B through SM. Phone A puts his own IP in the SDP for Phone B. Phone B does the same. All signaling messages through Phones A and B will be through SM, but Phone A will send its audio directly to Phone B and vice versa.

In your case with CM, CM will provide the SIP signaling for the H323 phone.

So, your signaling flow would be:
3rd party application-->SM-->CM-->H323 phone where CM handles the SIP/H323 conversion

Your media flow would be:
3rd party thing-->h323 phone

The media flow can be different if you force a gateway DSP in between.
The media flow can be different if you have early h323 media on your SIP signaling group in CM
The media flow can be through a DSP first and shuffle direct between the phone and other thing with SIP reinvites

Depending on what the other thing supports, you may need to tune CM's settings accordingly to give it something it can handle.

 

[Saiyan656]

Ok. So the call I am sending on trunk CM > SM > 3rd party is on one CM-1(TRU 900) and the trunk I am coming back on is another CM-2(TRU 950). With tie lines connecting the two, I will try sending the call back on 900 rather than 950.

Do you know how of any way to view RTP (IP's) in CM while I fine tune CM?

Thanks for the response!

 

[Saiyan656]

I don't see any issues on our CM side. Let me work with 3rd part application vendor and see if they are sending/receiving audio to SM and get back to this tread... Thanks again.

 

[kyle555]

CM won't show it to you unless you MST trace the SIP sig group.

Your best bet is to look through SM traces. They'll include the media setup information. If you have a complete packet capture you can share, I don't mind taking a quick look.

 

[Saiyan656]

I can send you SIP PCAP, but don't know how

 

[Saiyan656]

Sent

 

[Saiyan656]

Resolved!!! Had set type as 96xxsip and change 9611!!