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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Cc: "codec@ietf.org" <codec@ietf.org>
Subject: Re: [codec] #18: Frame Sizes
Precedence: list

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 efficie=
ncy gains made with such large payloads and more on their ability to carry =
said traffic cheaply as lower priority traffic over non-QoS controlled netw=
ork hops.=20

In any case, the ability to carry multiple codec frames within an RTP packe=
t is already standard voip system implementation practice and, with the exc=
eption of jitter buffer support for unpacking these composite RTP packets w=
hen 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=0A=
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=20
> be supporting two-way communications in a human interactive sense.

Because it might work better over your GSM data network which in itself=20
has 1s delay.

And you're wrong, I communicate over skype all the time with 2+ second=20
RTT. It's free, the 500ms RTT version costs quite a lot of money. Doing=20
bidirectional talk over long RTT is a matter of practice.

I'd rather have 2s RTT for free than paying 1USD / minute.

--=20
Mikael Abrahamsson    email: swmike@swm.pp.se