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

Koen Vos <koen.vos@skype.net> Thu, 14 March 2013 20:57 UTC

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From: Koen Vos <koen.vos@skype.net>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01
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Date: Thu, 14 Mar 2013 20:57:06 +0000
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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 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.

Skype has ever used "interop with the 100s of millions of our endpoints" as an argument :)

koen.

________________________________
From: Roman Shpount
Sent: Thursday, March 14, 2013 1:41 PM
To: Koen Vos
Cc: Espen Berger (espeberg); stephane.proust@orange.com; rtcweb@ietf.org; MARJOU Xavier OLNC/OLN
Subject: Re: [rtcweb] Agenda time request for draft-marjou-rtcweb-audio-codecs-for-interop-01


On Thu, Mar 14, 2013 at 4:32 PM, Koen Vos <koen.vos@skype.net<mailto: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