Re: [rtcweb] RTCWEB needs an Internet Codec

Martin Taylor <> Wed, 29 August 2012 11:26 UTC

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From: Martin Taylor <>
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Thread-Topic: [rtcweb] RTCWEB needs an Internet Codec
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Date: Wed, 29 Aug 2012 11:26:36 +0000
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Subject: Re: [rtcweb] RTCWEB needs an Internet Codec
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An Internet codec is one which was designed specifically to handle a significant level of packet loss.  The series codecs were designed for circuit-switched environments where packet loss is a non-issue.

If you are not experiencing voice quality issues with G.722, then you are lucky enough to have a very low loss Internet connection.  We see frequent issues with G.722 when used for VoIP, especially over WiFi and upstream over ADSL.  Data traffic that competes with voice for bandwidth frequently causes choppy voice on these types of connection.  We have chosen to use SILK to deal with this issue, and this has resulted in major improvements in subjective voice quality.  SILK is one of the "ingredients" of Opus.

I believe that WebRTC applications which use Opus will deliver a far better user experience in general than those that use G.711 or G.722.  Whether that means Opus should be mandatory to implement is another matter.  In my view, it is only necessary to specify one MTI codec to ensure that there is a baseline for interop.  The market can decide what codecs it wishes to use to improve on this baseline interop.


From: [] On Behalf Of Roman Shpount
Sent: 29 August 2012 05:14
To: Alan Johnston
Subject: Re: [rtcweb] RTCWEB needs an Internet Codec

Not that I have anything against Opus, but what exactly makes Opus an internet codec? What is internet codec anyway? Makes this all kind of a meaningless argument.

I would argue that for my broadband internet connection G.722 is a perfect internet codec. I do not care about bandwidth savings of Opus, and quality wise, for the voice conversation, I cannot hear any difference.

I would argue G.711 should be the MTI codec. The rest can be left up to browser implementers. We can argue all we want, but the best royalty free low bitrate codec available will be the one everybody supports. We can force it to be Opus, but even if we don't, it will still be Opus on its merit alone. G.722 will probably end up being supported as well, since it is free, pretty good quality, and easy to implement.

P.S. Not that I am arguing for it, I am surprised no one made a case for iSAC, since it is also royalty free, low bit rate, and very high quality. It is event named "internet Speech Audio Codec".
Roman Shpount

On Tue, Aug 28, 2012 at 11:41 PM, Alan Johnston <<>> wrote:
The RTCWEB effort needs an Internet codec.  This is why Opus is the
right choice.  RTCWEB also needs one codec for backwards compatibility
with the VoIP world.  This is why G.711 is also the right choice.

Any G.mumble codec is NOT an Internet codec and will not have the same
performance on the Internet as one that was designed for the Internet!
 If anyone doesn't understand what that means, go back and examine the
CODEC Working Group archives to get educated.

If someone wants another codec instead of Opus, then they need to
propose another Internet codec.  Otherwise, we are not serving the

- Alan -
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