Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

["Asveren, Tolga" <[e-mail]>]

Ah yes, I meant too high…

i- As a general principle “Absence of proof does not mean proof of absence”. That you/I/x has not seen an application in their lifetime using a value lower than 40ms does not mean a use case does not exist/can’t exist in the future.

ii- As another general principle, introducing a restriction without a good reason is a bad idea IMHO. So far, I yet have to see a good argument in favor of an enforced range (other than that we should assume that intelligence level of app. developer’s is equal to a chimpanzee)

iii- I saw logs using values less than 40ms (and am aware of SBCs supporting lower than 40ms values) but I am not sure right now what was the emitting node (but again, this really shouldn’t be the determining factor in this debate; ii- is the main point IMHO for a clean solution).

Thanks,
Tolga

From: Roman Shpount [mailto:[e-mail]]
Sent: Tuesday, March 01, 2016 1:32 PM
To: Asveren, Tolga <[e-mail]>
Cc: Stefan Håkansson LK <[e-mail]>; Harald Alvestrand <[e-mail]>; Ted Hardie <[e-mail]>; [e-mail]
Subject: Re: [rtcweb] Fwd: Last Call: <draft-ietf-rtcweb-audio-10.txt> (WebRTC Audio Codec and Processing Requirements) to Proposed Standard

On Tue, Mar 1, 2016 at 6:27 AM, Asveren, Tolga <[e-mail]<mailto:[e-mail]>> wrote:
 And 40ms is definitely too low of a value, I am aware of VoIP systems where lower values are used (but I definitely think that no enforced limit is the right thing to do rather than merely decreasing min).

I assume you meant 40 ms is too high of the value. If you have an example of an application (except testing apps) which is using a lower value, please provide it to the list.
_____________
Roman Shpount