On Thu, 7 May 2015, Jonathan Morton wrote:
However the more common characteristic is that delay is sometimes low (link idle) and sometimes high (buffer full) and rarely in between. In other words, delay samples are not statistically independent; loss due to jitter is bursty, and real-time applications like VoIP can't cope with that. For that reason, and due to your low temporal sampling rate, you should take the peak delay observed under load and compare it to the average during idle.
Well, some applications will stop playing if the playout-buffer is empty, and if the packet arrives late, just start playing again and then increase the PDV buffer to whatever gap was observed, and if the PDV buffer has sustained fill, start playing it faster or skipping packets to play down the PDV buffer fill again.
So you'll observe silence or cutouts, but you'll still hear all sound but after this event, your mouth-ear-mouth-ear delay has now increased.
As far as I can tell, for instance Skype has a lot of different ways to cope with changing characteristics of the path, which work a lot better than a 10 year old classic PSTN-style G.711-over-IP style system with static 40ms PDV buffers, which behave exactly as you describe.
-- Mikael Abrahamsson email: [email protected] _______________________________________________ Bloat mailing list [email protected] https://lists.bufferbloat.net/listinfo/bloat
