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

Colin Perkins <csp@csperkins.org> Thu, 15 April 2004 12:51 UTC

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Cc: "\"'Gunnar Hellström'\"" <gunnar.hellstrom@omnitor.se>, toip@snowshore.com, avt@ietf.org
From: Colin Perkins <csp@csperkins.org>
Subject: Re: [avt] Use of redundancy in rfc2793bis text transmission - optimizing interoperability
Date: Thu, 15 Apr 2004 12:23:07 +0100
To: Arnoud van Wijk <a.vwijk@viataal.nl>
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Arnoud,

On 15 Apr 2004, at 11:30, Arnoud van Wijk wrote:
> 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.

The drafts states that text may be generated by automated systems at 
high rate in some scenarios.

> 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.

Consider the cases where there is no parallel audio/visual session; 
where the link capacity is low and is shared between real-time text and 
TCP traffic; or where there are large numbers of real-time text 
sessions.

RFC 3551 says:

       If best-effort service is being used, RTP receivers SHOULD monitor
       packet loss to ensure that the packet loss rate is within
       acceptable parameters.  Packet loss is considered acceptable if a
       TCP flow across the same network path and experiencing the same
       network conditions would achieve an average throughput, measured
       on a reasonable timescale, that is not less than the RTP flow is
       achieving.  This condition can be satisfied by implementing
       congestion control mechanisms to adapt the transmission rate (or
       the number of layers subscribed for a layered multicast session),
       or by arranging for a receiver to leave the session if the loss
       rate is unacceptably high.

RFC 2198 says:

    Whilst the addition of low-bandwidth redundancy to an audio stream is
    an effective means by which that stream may be protected against
    packet loss, application designers should be aware that the addition
    of large amounts of redundancy will increase network congestion, and
    hence packet loss, leading to a worsening of the problem which the
    use of redundancy was intended to solve.  At its worst, this can lead
    to excessive network congestion and may constitute a denial of
    service attack.

The text conversation draft does not implement congestion control, nor 
does it mandate that receivers leave the session when the loss rate 
becomes unacceptably high. Instead, it mandates the addition of large 
amounts of redundancy to hide the packet loss, and recommends that the 
session continue.

Behaviour that directly contradicts the other RFCs may be appropriate, 
but this draft has to justify why it is acceptable, and why the 
recommendations in the other RFCs cannot be followed.

Colin


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