Subject: Re: ����: QoS and Call Admission

From: "Mo Zonoun" <[email-masked]>
Date: Fri, 18 Sep 1998 10:37:58 -0700

Eric Dean wrote:
> 
> Wow, this is great.  I like the perspectives.  I also wanted to add that
> there is more to consider other than congestion such as delay, jitter,
> group delay,...  Actually, a basic telephony book sited about 15
> different QoS sources that have to be considered in the PSTN network.
> 
> I really like to think that there is nothing unique to IP that also
> doesn't have to be considered when delivering Voice over PSTN or ATM.  As
> a matter of fact, with IP being connectionless, I imagine that we have
> more to consider.

We need to be clear with our terminology when we talk about QoS. Some aspect of
QoS such as average throughput, response to congestion , reliability of
delivery, security and others are defined or in some cases being defined in IETF
and we need to make reference to the appropriate RFCs as require for IpTel. 

Other aspects of QoS, unique for real time payloads (voice and video), are
related to transmission impairments (for PSTN typically: loss/level, echo,delay,
packet loss and distortion; for packet network, packet size, packet loss,
Coder/decoder delay and jitter are of concerns). These transmission impairments
are well understood and defined by other organizations such as ITU-T. We need to
work jointly with ITU-T and use their expertise to determine the requirement for
IpTel. 

The first issue is, where in the network to do what?  

Mo Zonoun
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From: Eric Dean <[email-masked]>
Date: Sat, 19 Sep 1998 13:56:38 -0400 (EDT)

> The first issue is, where in the network to do what?  

I don't understand your question?
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This message came from the IETF IPTEL Working Group Mailing List.

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From: "Mo Zonoun" <[email-masked]>
Date: Mon, 21 Sep 1998 10:03:52 -0700

> I don't understand your question?

Any compensation for the impairment of voice over IP (echo cancellation, loss
and etc.) must be done as close as possible to the user where the conversion
from analog to digital takes place. The reason for this is that you want to
avoid multiple transcoding which will significantly impairs voice quality. On
the other hand, the closest you get to the user the less information you have
about QoS parameters from IP, and hence less control.

Mo Zonoun
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