[VoiceOps] High Quality, Reliable Voice via the Internet / SIPNOC
avorlando at yahoo.com
Mon Jun 9 21:56:00 EDT 2014
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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