Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01

Roman Shpount <roman@telurix.com> Thu, 14 March 2013 20:42 UTC

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Date: Thu, 14 Mar 2013 16:41:55 -0400
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From: Roman Shpount <roman@telurix.com>
To: Koen Vos <koen.vos@skype.net>
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Cc: MARJOU Xavier OLNC/OLN <xavier.marjou@orange.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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On Thu, Mar 14, 2013 at 4:32 PM, Koen Vos <koen.vos@skype.net> wrote:

> WebRTC connecting to a Telco network would have to go through a Gateway.
>  The PSTN termination providers who run these Gateways support G.711, G.729
> and perhaps some other  codecs like iLBC.  They do not, however, support
> the codecs you are advocating for.
>
> The lack of support for Transcoding-Free Operation by Telcos to the rest
> of the world has been hurting interop voice quality for a long time, but
> unfortunately we can't fix that here at the IETF.
>
> We can fit our cars with your costly railroad wheels, but you still
> wouldn't let us on your tracks.
>
>
I would got to agree with you regarding cell phone operators, since we've
been trying to get access to native AMR-WB for years.

On the other hand, this is kind of interesting to hear coming from someone
from Skype, which does not allow connections using SILK from outside of
their own network, which has been hurting interop voice quality for a long
time as well.

Finally, it is possible to connect to almost any hosted PBX provider, as
well as quite a few premise based PBXs, using G.722, as long as their end
points support it.
_____________
Roman Shpount