RTP question

[C_Noble]

I understand that after SIP call setup, RTP is between two end devices unless there is an MTP in the path.

I want to confirm my understanding for the following call scenario for MGCP and T1 PRI:

SIP Phone (1.1.1.1) > HQ CUCM > MGCP GW - T1PRI > Telco > MGCP GW - T1 PRI > Branch CUCM > SIP Phone (2.2.2.2)

After call setup for the above, is the source and destination of the media:
1.1.1.1 to 2.2.2.2

Or does RTP terminate at the first MGCP gateway before traversing the T1?

[TechLvr · Accepted solution]

Since there is a T1/PRI link between the two gateways, the is no IP address info exchanged during the call setup at all. T1/PRI is TDM based, not IP based. The two SIP phones in your example won't know about each other's IP addresses/RTP ports, and RTP is NOT end to end between the two endpoints.

So yes, "RTP terminates at the first MGCP gateway before traversing the T1" trunk.

[Deepak Kumar R]

You have PSTN(Telco) and MGCP GW in between so RTP cannot directly traverse from 1.1.1.1 to 2.2.2.2. Moreover, the voice gateway (MGCP in your case) connects your enterprise voice network to PSTN. So if a call is initiated from 1.1.1.1 to 2.2.2.2 then the RTP will traverse as below

1.1.1.1 < ----> MGCP GW <----> PSTN <-----> MGCP GW <-----> 2.2.2.2

[Roger Kallberg]

The second option. There is no way that the RTP traffic can flow between the disparate networks that are interconnected via PSTN.