In order to measure delay in a VOIP architecture, I use Cisco-Voice-Dial-Control-Mib. Does anyone knows how the VoIPCallActiveRoundTripDelay Mib is calculated?
When I use a small client-server program (udpSpeed) to simulate an audio communication on the IP network, the RTT is in accordance with previsions, even it the network is loaded (I use crtp and ppp multilink). But when I make the same test with an audio communication between two gateways (using dial-peer), the RTT is really too high. The audio packet are well marked with dscp ef flags. I guess there is something wrong in the dial-peer, but I do not know what (one interface is FXO, the other is FXS).
Seems to me this RTT is not measured from timestamps of EF packets, but is measured from control & signalization packets. Thoses are AF31 and are not placed in priority queues. One way to obtain right RTT is to mark thoses packets as EF.
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