wg-voip - Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes)
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- From: Guy T Almes <>
- To: Ben Teitelbaum <>, , VoIP Working Group <>
- Subject: Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes)
- Date: Thu, 23 Oct 2003 08:34:43 -0400
Ben et al.,
This is a great thread, and I agree with almost all of your comments specifically.
For too long the TCP experts have operated purely in the context of how to optimize large bulk transfers. As a recent example of a success, the CERN/Caltech team have succeeded in pushing 5.6 Gb/s end-to-end between CERN/Geneve and Caltech/LosAngeles.
When this is done with Reno-based techniques, they very intentionally build up (and after packet loss events, tear down) massive buffers at the bottleneck router. This cannot be good for VoIP etc.
Since late 2002 we've been working with the FAST TCP folks (Steven Low and his colleagues) at Caltech. Their control algorithms for TCP are grounded in control theory and attempt to maintain moderate-sized queues at the bottleneck. Though they were at first focused entirely on throughput and TCP fairness, we pointed out to them that both the moderate-size and the consistency of size of queues would be good for VoIP etc.
I'd love to hear more from you and the others in wg-pic and wg-voip about the specifics of how amount of queueing delay and stability of queueing delay impinge on their work.
One quibble: I would not want to burden your effort too early, but I would argue that congestion control will be needed if/when large-scale deployment occurs. The cute way of saying it is that you only need to do it if your application succeeds.
Regards,
-- Guy
--On Wednesday, October 22, 2003 22:25:07 -0400 Ben Teitelbaum <> wrote:
Jeremy George
<>
writes:
One of the things we can do is feedback to network engineers the
necessity of substantially eliminating congestion (and not buffering)
when real time applications are running. I think this is non-trivial
feedback and one I've pushed at yale. A really good data network is
not necessarily automatically good for real time apps. I think this is
a topic worth some exploration, but probably not by this group.
I think this *is* a topic of which both the PIC and VoIP groups should
be mindful and which we should discuss from time to time. For now,
outside of pathological environments where hundreds of people are
sharing the ether, real-time applications with adaptive, loss-tolerant
codecs work well. In the longer term, we need to worry about
co-existence with non-real-time bulk transfers.
Let us not forget that TCP is *designed* to congest. With bigger
MTUs, "tuned" socket buffering, and tricks like zero-copy, TCP Reno
will fill the pipe and, worse, build a queue at some interface. That
interface will have a delay-bandwidth product of memory adding
something like 150ms of delay to the comingled VoIP traffic. On top
of codec delay and propagation delay, this could push a lot of voice
traffic into uncomfortable territory (>250ms).
Lightweight QoS techniques (e.g. alternative best-effort (ABE)) could
help isolate real-time traffic. Well-tuned RED could avoid
tail-dropping and therefore the consecutive loss patterns that destroy
even clever voice codecs. Finally, new transport protocols like FAST
(from CalTech) or XCP (from MIT) could improve bulk data throughputs,
while reducing the need to have so much memory in the routers.
All of this is of critical long-term importance to us.
Finally, the concerns raised by Sally Floyd in
http://www.ietf.org/internet-drafts/draft-iab-congestion-01.txt don't
worry me much in the Internet2 environment. Non-adaptive or
coarsely-adaptive applications like voice should be just fine as long
as the streams are "small". A single 128kbps voice stream is tiny in
comparison to the slowest interface on a "typical Internet2 path"
(ignoring wireless, lets define this as switched 10Mbps). It would be
interesting to do a back-of-the-envelope worst-case analysis assuming
complete POTS replacement and the current Internet2 topology and see
if we could estimate the feasibility or improbability of congestion
collapse due to VoIP.
Does anyone have busy-hour call volume figures from a campus voice
switch they could share? With a few datapoints, I'd do this
back-of-the-envelope analysis.
What worries me in the Internet2 environment are the non-adaptive
video applications we are seeing that are not at all "small".
-- ben
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- Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Ben Teitelbaum, 10/22/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Guy T Almes, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Jeremy George, 10/23/2003
- Re: [WG-PIC:366] Real and Non-Real Time Traffic (was Re: FMM demo notes), Chris Peabody, 10/23/2003
- Re: [WG-PIC:367] Re: Real and Non-Real Time Traffic, Ben Teitelbaum, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Jon Zeeff, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Dennis Baron, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Christine Moe, 10/23/2003
- <Possible follow-up(s)>
- RE: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Enyeart, Mike, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Chris Peabody, 10/23/2003
- RE: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Enyeart, Mike, 10/23/2003
- RE: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Erik Dobbelsteijn, 10/23/2003
- Re: Real and Non-Real Time Traffic (was [WG-PIC:352] Re: FMM demo notes), Guy T Almes, 10/23/2003
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