One problem with that theory. At 40ms you have more samples per packet making it more difficult for a PLC algorithm to interpolate . Bigger chunks of audio are now missing. Sent from my iPhone
On Jun 9, 2014, at 9:45 PM, "Mark R Lindsey, ECG" <lindsey at e-c-group.com> wrote:
On Mon, Jun 9, 2014 at 4:14 PM, Alex Balashov <abalashov at evaristesys.com> wrote:
On 06/09/2014 02:50 PM, Mark R Lindsey wrote: 2. Increase the ptime from 20 ms to 30-40 ms to reduce packet-drop exposure
Or does this thesis lean on countervailing tendencies, such as overall reduced PPS in a higher ptime scenario?
You're on the right track with ptime. The theory idea is that:
(A) Most packet loss is due to congestion
(B) When congestion occurs the router selects a packet to drop
(C) The routers pick a packet to discard more-or-less at random
(D) Therefore, A 180 byte packet is just as likely to be dropped as a 1500 byte packet.
(E) A ptime=20 generates twice the packets as ptime=40, and therefore ptime=20 has twice the exposure to the discards
(F) You can reduce your exposure to discards by reducing the number of packets you have in the queue.
(G) Reduced discards mean better audio quality.
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