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Re: [codec] #18: Frame Sizes

Michael Knappe <mknappe@juniper.net> Thu, 13 May 2010 00:41 UTC

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From: Michael Knappe <mknappe@juniper.net>
To: "swmike@swm.pp.se" <swmike@swm.pp.se>
Date: Wed, 12 May 2010 17:37:43 -0700
Thread-Topic: [codec] #18: Frame Sizes
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Subject: Re: [codec] #18: Frame Sizes
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Fair enough, although a service provider's decision to offer a free service with long round trip delays is likely less dependent on the header efficiency gains made with such large payloads and more on their ability to carry said traffic cheaply as lower priority traffic over non-QoS controlled network hops. 

In any case, the ability to carry multiple codec frames within an RTP packet is already standard voip system implementation practice and, with the exception of jitter buffer support for unpacking these composite RTP packets when received, is outside the scope of the codec wg.

Mike

----- Original Message -----
From: Mikael Abrahamsson <swmike@swm.pp.se>
To: Michael Knappe
Cc: codec@ietf.org <codec@ietf.org>
Sent: Wed May 12 18:27:11 2010
Subject: Re: [codec] #18: Frame Sizes

On Wed, 12 May 2010, Michael Knappe wrote:

> Why would you need frame sizes approaching 250 ms? This would no longer 
> be supporting two-way communications in a human interactive sense.

Because it might work better over your GSM data network which in itself 
has 1s delay.

And you're wrong, I communicate over skype all the time with 2+ second 
RTT. It's free, the 500ms RTT version costs quite a lot of money. Doing 
bidirectional talk over long RTT is a matter of practice.

I'd rather have 2s RTT for free than paying 1USD / minute.

-- 
Mikael Abrahamsson    email: swmike@swm.pp.se

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