Subject: RE: [Ieprep] any commercial usage for "emergency QoS"?

From: "Nguyen, An" <[email-masked]>
Date: Wed, 27 Nov 2002 05:25:49 -0500

Bill,

Sorry I was not there for the demos. I wondered if you took into account the
peak internet usage time when you gave the demos. I heard the demos were
given around 4:00PM EST( please correct me if I am wrong). That would have
been around 3:00 or 4:00 AM in Nairobi??? Do you consider those two time
zones are in the peak usage intervals? In other words, was the highway
basically empty when your traveled? What is the utilization factor of your
pipes when you VoiceIP calls traverse from one point to another? 

It would be nice if you could show us that given the traffic utilization
factor of , say, from .1 to 0.5, your voiceIP can still be good or something
to that extent. 

I truly believe that quality of the call is related to traffic utilization
and congestion of the network.

Regards,

An
-----Original Message-----
From: Bill Woodcock
To: RJ Atkinson
Cc: [email-masked]
Sent: 11/26/02 9:31 PM
Subject: Re: [Ieprep] any commercial usage for "emergency QoS"?

    > By the way, during the Nairobi VoIP demo in Atlanta, was there any
    > use of IP ToS bits, DiffServ, or other QoS ?

Nope.  We wanted it to work.  Demos which fail aren't as impressive.

                                -Bill

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---

From: RJ Atkinson <[email-masked]>
Date: Wed, 27 Nov 2002 06:43:09 -0500

On Wednesday, Nov 27, 2002, at 05:25 America/Montreal, Nguyen, An wrote:
> been around 3:00 or 4:00 AM in Nairobi???

	Bill gave the same demo on multiple occasions during that week, with 
the same
results each time.  The time of day for the various demos varied pretty 
widely
(e.g. the IEPG demo was mid-morning US EST).  The results of the demo 
did
not vary in terms of voice quality, though there were differences in 
the amount
of packet loss along the path at the different times.  [As to the 
choice of
the demo time for IEprep, Bill did not have any choice since he was 
asked to
do the demo during the IEprep meeting slot (and Bill wasn't involved in 
scheduling
that meeting time).]

> Do you consider those two time

	There was 40% packet loss along the path (mostly inside Kenya) during 
the
IEPG demo.  The demo quality remained unchanged (as compared with 
IEprep demo)

> factor of , say, from .1 to 0.5, your voiceIP can still be good or 

	What is a "traffic utilisation factor" ?  I'm thinking you have a 
specific
equation in mind, but am not sure what that equation might be.

> and congestion of the network.

	Well maybe not really closely related.  VoIP is a UDP service that 
lacks
congestion avoidance mechanisms.[1]  TCP and SCTP have congestion 
avoidance.  So if UDP
flows were to consume a lot of bandwidth, then the TCP and SCTP 
sessions sharing
that path will tend to back off, which tends to yield more bandwidth 
available
for those UDP flows.  (As Sally Floyd pointed out during IEprep, this 
also means
that IETF should probably put more work into adding congestion 
avoidance mechanisms
into VoIP applications.[2])

	Dave Oran, who knows a fair bit about the internals of the IP phone 
that
Bill was using for the demos, said that the particular IP phone that 
was used
for the demo always adds substantial implementation-specific FEC 
beneath the
G.711 codec output and above UDP.  That is, this FEC is a normal 
property of that
particular IP phone in its default configuration.  Understanding that 
there was
FEC added (rather than pure G.711) does help me understand why an IP 
phone using G.711
was able to work so well despite 40% packet loss inside Kenya.  As 
noted earlier,
I'm not tremendously impressed by the robustness of the G.711 codec -- 
CELP would
seem like a better choice for VoIP.

Ran
[email-masked]

[1] If VoIP were a TCP or SCTP service, then there would likely be more 
issues
with VoIP over a path with high packet loss (or very high delay).

[2] This is why in many WGs, folks hear the Transport ADs insist that 
TCP-like
congestion avoidance get deployed.  To pick a current example, NFSv4 
runs over
TCP *only*, not over UDP, and congestion avoidance is specifically 
listed as the
reason that UDP is disallowed.  Originally NFSv{1,2} ran over UDP.  
NFSv3 supported
both TCP and UDP, but preferred TCP transport.

---

From: Bill Woodcock <[email-masked]>
Date: Wed, 27 Nov 2002 09:36:06 -0800 (PST)

> been around 3:00 or 4:00 AM in Nairobi?

I don't remember the exact times, I do remember that, Brian's time, one
was about 8pm, and the other about 10pm.  However, that's not terribly
relevant, since those were just two phone calls, and these phones are used
day in and day out, during business hours, evening hours, et cetera.  The
point was that this works really well, not that two specific calls worked
really well.

    > factor of , say, from .1 to 0.5, your voiceIP can still be good or something

It's not "my VoIP," it's an open standard.  It's _your_ VoIP!  :-)  If you
have doubts about it, why don't you actually try it yourself, and then the
doubts will go away, and you'll also start tying to avoid using the PSTN
whenever you can.  I'm not selling anything here, I don't need to.  I'm
just pointing out the obvious.

                                -Bill

---

From: Sean Donelan <[email-masked]>
Date: Wed, 27 Nov 2002 17:27:12 -0500 (EST)

> factor of , say, from .1 to 0.5, your voiceIP can still be good or something

I have an idea, why not buy an IP Phone and try it yourself?

---

From: RJ Atkinson <[email-masked]>
Date: Fri, 29 Nov 2002 17:41:07 -0500

On Wednesday, Nov 27, 2002, at 17:27 America/Montreal, Sean Donelan 
> I have an idea, why not buy an IP Phone and try it yourself?

The DISA Skyline building is full of cisco IP phones.  I'll check model
numbers the next time I'm over there to compare with Bill, but I'd be
startled if they don't already have a pile of phones interoperable
with Bill's (likely even the same model number).  Last time I was at
Courthouse Road, they still had real phones there, however.

So running the test ought not require money to buy a phone even.

Ran

---

From: "Steve Silverman" <[email-masked]>
Date: Fri, 29 Nov 2002 22:32:46 -0500

While advertised as VoIP, the Skyline building, like most
implementations is actually VoLAN.  The IP is along for the ride.
With a 100Mb Ethernet, the line utilization is rarely as high as
.01.  In addition, voice has priority over data.  You would have a
hard time achieving a 10% loading on such a LAN.  At the WAN
interface, they convert to TDM.

>From what I've heard, very few users are using VoIP on a WAN where
the IP has to contend with moderately loaded (> 30% loaded lines).
I would very much like to hear from anyone with real experience with
VoIP over a typically loaded WAN.

Steve Silverman

> From: [email-masked]
> [mailto:[email-masked]]On Behalf Of
> RJ Atkinson
> Sent: Friday, November 29, 2002 5:41 PM
> To: Sean Donelan
> Cc: [email-masked] '
> Subject: Re: Try it, you might like it RE: [Ieprep] any
> commercial usage
> for "emergency QoS"?
>
>
>
> >> factor of , say, from .1 to 0.5, your voiceIP can
> still be good or
> >> something
> >> to that extent.
> >

---

From: Henning Schulzrinne <[email-masked]>
Date: Sat, 30 Nov 2002 07:13:06 -0500

The University of Kentucky uses a VoIP system in many of its buildings, 
including the desk I'm currently sitting at. It's converted to PRI 
on-site. From my experience so far, the quality is at least equivalent 
to a traditional PBX. One should note, however, that the LAN was 
installed maybe three years ago, so it might be in better shape than 
many. (See recent reports on the LAN outages at a Boston-area teaching 
hospital.)