Lars Eggert, I have read most of this report, which I find very good and interesting. I have three questions for you: 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? 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. Best regards Arne Lie NTNU - Norway > -----Opprinnelig melding----- > Fra: Lars Eggert [mailto:lars.eggert@xxxxxxxxxxxxx] > Sendt: 2. august 2006 09:12 > Til: dccp@xxxxxxxx > Emne: DCCP voice quality experiments > > Hi, > > we finally have the results of our voice quality experiments > over DCCP written up: > http://larseggert.de/tmp/2006-dccp-voip-quality.pdf > > We'd appreciate any feedback you may have on this! > > Thanks, > Lars > -- > Lars Eggert NEC Network > Laboratories > > >
