RE: [avt] Use of redundancy in rfc2793bis text transmission - optimizing interoperability

["Arnoud van Wijk" <[e-mail]>]

Hmm.
When looking at an Interactive text stream, even with full redundancy, the
used bandwidth is a fraction of the bandwidth used by video and audio.

So knowing this, it is then desirable to cope with the packetloss by using
the full redundancy. The effect of reducing the sending rate is nihil when
combined with audio and/or video.

Also, the minimal sending speed is equal to the typing speed, it is
real-time text after all on a character by character basis.

With full redundancy on, you ensure that interactive text arrived complete,
while the audio and video may stutter. I would not mind to switch off the
audio then and continue with the video and interactive text.

That is how I see it.

Arnoud

Drs. Arnoud A. T. van Wijk
Viataal
Research & Development
Afdeling RDS
Theerestraat 42
5271 GD Sint-Michielsgestel
The Netherlands. 
Mobile: [phone]
International text telephone: [phone]

-----Original Message-----
From: [e-mail] [mailto:[e-mail]] On Behalf Of Colin
Perkins
Sent: donderdag 15 april 2004 11:55
To: Gunnar Hellström
Cc: [e-mail]; [e-mail]
Subject: Re: [avt] Use of redundancy in rfc2793bis text transmission -
optimizing interoperability

On 14 Apr 2004, at 21:59, Gunnar Hellström wrote:
> I often get comments on the real time interactive text conversation 
> transport
> RFC2793bis, that it must more clearly require the use of redundancy to 
> assure
> good success rate in text transmission even in bad network conditions.
>
> Next version of draft-ietf-avt-rfc2793bis is about to be published, 
> and I would like to have  agreeable wording on this issue.
>
> A traditional requirement for basic text conversation quality is that 
> no more
> than 1% characters may be lost in conditions where voice 
> communications is
> barely usable. Where characters are dropped marks for missing text 
> should be
> inserted in the received text. A higher quality level, called good 
> text quality requires no more than 0.2% characters to be dropped. This 
> is of course no exact scientific measure and many factors influence 
> both the loss and the perception of usability of voice conversation, 
> but it gives a fair design goal.
>
> With voice coding and transmission schemes prevailing today, voice 
> gets barely
> usable around 20% packet loss.
>
> Therefore, in order to assure proper interoperability with the 
> required quality level for text at 20% packet loss, we need to require 
> one original and two redundant transmissions according to RFC2198. 
> (lowering text loss to 0.8%)

Why do you think it acceptable to be sending at full rate when there is 
20% packet loss, rather than reducing the sending rate to reduce packet 
loss? Do you have some analysis to show that, with typical text 
conversation packet rates and network RTTs, transmission with 20% 
packet loss is TCP friendly (or otherwise)?

The requirements you list may be appropriate from a media quality 
viewpoint, however you also need to demonstrate that the payload format 
is safe to deploy. This is particularly important for a format which 
adds a large amount of FEC to allow it to operate in the presence of 
large amounts of congestion, but doesn't specify a congestion control 
response.

-- 
Colin Perkins
http://csperkins.org/

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Audio/Video Transport Working Group
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