Re: [rtcweb] RTCWEB needs an Internet Codec

[Randell Jesup <[e-mail]>]

On 8/31/2012 9:38 AM, John Leslie wrote:
> Harald Alvestrand <[e-mail]> wrote:
>> If necessary, we should make the document more explicit to say that we
>> don't solve for use cases where less than 10s of kilobits of Internet
>> bandwidth is available...
>     +1

Agreed.

>     There are today some use-cases with less than 56Kbps; but they're
> dying, and the infrastructure for them isn't being built anywhere
> (including third-world countries, if you must ask).
>
>     I'm very glad there _are_ codecs that can operate in such an
> environment; but that's not where our effort belongs. We want something
> optimized for 5-10 years down John Leslie

If you're talking audio-only calls, I agree - but for audio+video, you 
want a lot of bandwidth available for video.  If the user has 128Kbps, I 
don't want 56K of that devoted to audio; I probably want 15-25K for 
audio (which can give good wideband or better performance with Opus), 
and the rest for video.  I probably wouldn't open audio up to 56 or 64K 
(using Opus) below 300-500K of total bandwidth, because of diminishing 
rate of improvement over ~25K.

Having a *good* low-bitrate (and adaptable) audio codec is *critical* to 
having a good solution for video.  Without that all the cases that 
involve video are blown below 100's of Kbps; with a good low-bandwidth 
codec you can do video calls with 100Kbps or even less.  This is why I 
support Opus (+G.711 for legacy interop and testing) for MTI.

-- 
Randell Jesup
[e-mail]