Hi, Arne,
1. What I find difficult to understand is if your applications have any
native rate adaptation, except packet drops at sender when the sender buffer
(of 5 packets?) have been saturated.
Is it so that *if* (and when) TFRC
gives you an equivalent rate below G.711 rate of 95.2kbps, or 39.2kbps for
G.729, your sender WILL drop packets when using TFRC (any variant), while
the UDP system will hand the problem over to the network routers?
TFRC can cap the send rate at the sender, which UDP doesn't.
I.e., my
main question is if G.711 and G.729 have any methods for lowering the codec
rate output below 64 and 8kbps, respectively (by quantiser scale change, or
any other means)
2. You have an experimental set-up. Why do you not also set-up real
receivers (players) so that the perceptual quality can be evaluated, instead
of, or in addition to, your "E-model R-score"
3. You use DummyNet to insert packet loss and delay. Are you considering
experimenting without DummyNet, but instead inject more real traffic to
create real router congestion? I think that will give you a more natural
environment in which you can test TFRC performance under more realistic
scenarios.
Lars -- Lars Eggert NEC Network Laboratories
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