I am using a combined ATA and router as the end point on a SIP based service. Prioritising the SIP element is easy because it uses specific ports, however, the actual voice traffic uses RTP on a random port. The ATA/router uses only one IP address on it's WAN interface so I can't identify the traffic by IP.

The ATA sets ToS/DiffServ values but are these honoured by MT? And even if they are, is this enough to garantee bandwidth to VoIP traffic even when the the 'pipe' is completely full?

Thanks in advance for any suggestions.

I guess you need a router that can identify RTP streams. There's a lot of articles on the Cisco website on this subject. Not something I've had to look at yet though.

Regards

Andrew