SIP Trunk and RTP ports range and Conductor vTS port range:

[tknapp]

I am trying to get a explanation of SIP Trunk and the use of RTP start and stop range of 16384-32767.

this is in regards to Telepresence conferences being hosted on vTS.   SIP trunk is built for AD-HOC and Meet Me conferences to a Conductor.   The Conductor is paired with a vTS server which is useing Video ports 49152-65535.    My question is the following:  DX80 A and DX80 B are in point to point call.  Video and Audio.   DX80 A brings in DX80 C and creates a ad-hoc video/audio conference call.

The SIP trunk between CUCM and Conductor is set for RTP ports 16384-32767.  HOW is video passed across the trunk to the Conductor?  I understand Audio getting across but how does Video get to Conductor. 

any help appreciated thank you.

[Jaime Valencia]

Conductor does NOT handle any media, conductor is the liaison between CUCM/devices and video conference bridges to "orchestrate"the conferences.

https://www.cisco.com/c/en/us/products/collateral/conferencing/telepresence-conductor/data_sheet_c78-709046.html

https://www.youtube.com/watch?v=Qk1VPc51wQg

[tknapp]

Thank you for your response.   With the provided information, I still do not understand the issue SIP trunk profile Media port range purpose.

The vTS is set for ports 49152-65535.  The SIP trunks to the Conductor are set for 16,384-32767.

Are the SIP profile port ranges NOT relevant? 

The SIP trunk is only for Signalling?

Thank you

[Jaime Valencia]

In this case, that SIP trunk to conductor, is just used for signaling as he will then find either the rendezvous meeting, or create an ad-hoc meeting and let know the endpoints what to dial, and then the media will flow between endpoints and the video CFB.