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From: Fil Dickinson <Fil.Dickinson@optus.com.au>
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My understanding of this draft is that it proposes the Voice signalling and
Voice Bearer are transported in the same class with the exception being
"signaling flows between high capacity telephony call servers or soft
switches". 
I am concerned with this. During testing of our multi-vendor converged
network I have seen that oversubscribed voice bearer traffic can disrupt the
voice signalling traffic which results in abnormal operation of all voice
calls [re-registration (outage) of IP Phones]. I would like to see this
addressed by a draft that mandates All Voice Signalling and Voice Bearer
traffic operate in distinct classes. Ie, a Telephony Signalling Service
Class and a Telephony Bearer Service Class.

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Subject: [Tsvwg] RFC 3758 on Stream Control Transmission Protocol (SCTP)
	Partial Reliability Extension
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--NextPart


A new Request for Comments is now available in online RFC libraries.


        RFC 3758

        Title:      Stream Control Transmission Protocol (SCTP)
                    Partial Reliability Extension
        Author(s):  R. Stewart, M. Ramalho, Q. Xie, M. Tuexen,
                    P. Conrad
        Status:     Standards Track
        Date:       May 2004
        Mailbox:    rrs@cisco.com, mramalho@cisco.com,
                    qxie1@email.mot.com, tuexen@fh-muenster.de,
                    conrad@acm.org
        Pages:      22
        Characters: 50999
        Updates/Obsoletes/SeeAlso:    None

        I-D Tag:    draft-ietf-tsvwg-prsctp-03.txt

        URL:        ftp://ftp.rfc-editor.org/in-notes/rfc3758.txt


This memo describes an extension to the Stream Control Transmission
Protocol (SCTP) that allows an SCTP endpoint to signal to
its peer that it should move the cumulative ack point forward.  When
both sides of an SCTP association support this extension, it can be
used by an SCTP implementation to provide partially reliable data
transmission service to an upper layer protocol.  This memo describes
the protocol extensions, which consist of a new parameter for
INIT and INIT ACK, and a new FORWARD TSN chunk type, and provides one
example of a partially reliable service that can be provided to the
upper layer via this mechanism.

This document is a product of the Transport Area Working Group of the
IETF.

This is now a Proposed Standard Protocol.

This document specifies an Internet standards track protocol for
the Internet community, and requests discussion and suggestions
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Joyce K. Reynolds and Sandy Ginoza
USC/Information Sciences Institute

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_______________________________________________
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From tsvwg-bounces@ietf.org  Thu May 27 09:08:53 2004
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From: "Jozef Babiarz" <babiarz@nortelnetworks.com>
To: Fil Dickinson <Fil.Dickinson@optus.com.au>, tsvwg@ietf.org
Subject: RE: [Tsvwg] draft-baker-diffserv-basic-classes-02
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In the next revision (03) of the draft (should be put on the mailing list
shortly) we state that peer-to-peer signaling such as SIP and H.323 be
forwarded out of a different service class than telephony payload. I will
analyze the impact on speech and ring clipping at beginning of call when
interfacing to a TDM telephony switch if all telephony signaling would be
forwarded using a different queue (higher latency) than the voice bearer. My
motivation for mapping telephony signaling in to the same queue as voice
bearer was to eliminate the delta delay between audio and signaling,
therefore reducing the chance of ring and speech clipping at beginning of
call. Other suggestions to address this problem are welcome.

In the draft we recommend that for Telephony service, per DSCP traffic
conditioning, policing of EF and CS5 market packets interpedently be
performed, as well flow admission control needs to be performed so that
oversubscription within the Telephony services does not occur.
 
Regards, Joe.
-----Original Message-----
From: Fil Dickinson [mailto:Fil.Dickinson@optus.com.au] 
Sent: Tuesday, May 25, 2004 12:27 AM
To: tsvwg@ietf.org
Subject: [Tsvwg] draft-baker-diffserv-basic-classes-02

My understanding of this draft is that it proposes the Voice signalling and
Voice Bearer are transported in the same class with the exception being
"signaling flows between high capacity telephony call servers or soft
switches". 
I am concerned with this. During testing of our multi-vendor converged
network I have seen that oversubscribed voice bearer traffic can disrupt the
voice signalling traffic which results in abnormal operation of all voice
calls [re-registration (outage) of IP Phones]. I would like to see this
addressed by a draft that mandates All Voice Signalling and Voice Bearer
traffic operate in distinct classes. Ie, a Telephony Signalling Service
Class and a Telephony Bearer Service Class.

_______________________________________________
tsvwg mailing list
tsvwg@ietf.org
https://www1.ietf.org/mailman/listinfo/tsvwg

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<TITLE>RE: [Tsvwg] draft-baker-diffserv-basic-classes-02</TITLE>
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<P><FONT SIZE=3D2>In the next revision (03) of the draft (should be put =
on the mailing list shortly) we state that peer-to-peer signaling such =
as SIP and H.323 be forwarded out of a different service class than =
telephony payload. I will analyze the impact on speech and ring =
clipping at beginning of call when interfacing to a TDM telephony =
switch if all telephony signaling would be forwarded using a different =
queue (higher latency) than the voice bearer. My motivation for mapping =
telephony signaling in to the same queue as voice bearer was to =
eliminate the delta delay between audio and signaling, therefore =
reducing the chance of ring and speech clipping at beginning of call. =
Other suggestions to address this problem are welcome.</FONT></P>

<P><FONT SIZE=3D2>In the draft we recommend that for Telephony service, =
per DSCP traffic conditioning, policing of EF and CS5 market packets =
interpedently be performed, as well flow admission control needs to be =
performed so that oversubscription within the Telephony services does =
not occur.</FONT></P>

<P><FONT SIZE=3D2>&nbsp;</FONT>
<BR><FONT SIZE=3D2>Regards, Joe.</FONT>
<BR><FONT SIZE=3D2>-----Original Message-----</FONT>
<BR><FONT SIZE=3D2>From: Fil Dickinson [<A =
HREF=3D"mailto:Fil.Dickinson@optus.com.au">mailto:Fil.Dickinson@optus.co=
m.au</A>] </FONT>
<BR><FONT SIZE=3D2>Sent: Tuesday, May 25, 2004 12:27 AM</FONT>
<BR><FONT SIZE=3D2>To: tsvwg@ietf.org</FONT>
<BR><FONT SIZE=3D2>Subject: [Tsvwg] =
draft-baker-diffserv-basic-classes-02</FONT>
</P>

<P><FONT SIZE=3D2>My understanding of this draft is that it proposes =
the Voice signalling and</FONT>
<BR><FONT SIZE=3D2>Voice Bearer are transported in the same class with =
the exception being</FONT>
<BR><FONT SIZE=3D2>&quot;signaling flows between high capacity =
telephony call servers or soft</FONT>
<BR><FONT SIZE=3D2>switches&quot;. </FONT>
<BR><FONT SIZE=3D2>I am concerned with this. During testing of our =
multi-vendor converged</FONT>
<BR><FONT SIZE=3D2>network I have seen that oversubscribed voice bearer =
traffic can disrupt the</FONT>
<BR><FONT SIZE=3D2>voice signalling traffic which results in abnormal =
operation of all voice</FONT>
<BR><FONT SIZE=3D2>calls [re-registration (outage) of IP Phones]. I =
would like to see this</FONT>
<BR><FONT SIZE=3D2>addressed by a draft that mandates All Voice =
Signalling and Voice Bearer</FONT>
<BR><FONT SIZE=3D2>traffic operate in distinct classes. Ie, a Telephony =
Signalling Service</FONT>
<BR><FONT SIZE=3D2>Class and a Telephony Bearer Service Class.</FONT>
</P>

<P><FONT =
SIZE=3D2>_______________________________________________</FONT>
<BR><FONT SIZE=3D2>tsvwg mailing list</FONT>
<BR><FONT SIZE=3D2>tsvwg@ietf.org</FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"https://www1.ietf.org/mailman/listinfo/tsvwg" =
TARGET=3D"_blank">https://www1.ietf.org/mailman/listinfo/tsvwg</A></FONT=
>
</P>

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Message-ID: <194.293a358c.2de8017b@aol.com>
Date: Thu, 27 May 2004 22:44:11 EDT
Subject: Re: [Tsvwg] draft-baker-diffserv-basic-classes-02
To: babiarz@nortelnetworks.com, Fil.Dickinson@optus.com.au, tsvwg@ietf.org
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In a message dated 5/27/2004 9:07:27 AM Eastern Standard Time, 
babiarz@nortelnetworks.com writes:


> In the next revision (03) of the draft (should be put on the mailing list 
> shortly) we state that peer-to-peer signaling such as SIP and H.323 be 
> forwarded out of a different service class than telephony payload. I will analyze 
> the impact on speech and ring clipping at beginning of call when interfacing to 
> a TDM telephony switch if all telephony signaling would be forwarded using a 
> different queue (higher latency) than the voice bearer. My motivation for 
> mapping telephony signaling in to the same queue as voice bearer was to 
> eliminate the delta delay between audio and signaling, therefore reducing the chance 
> of ring and speech clipping at beginning of call. Other suggestions to 
> address this problem are welcome.
> In the draft we recommend that for Telephony service, per DSCP traffic 
> 


I believe this remains as one of the major open questions for the application 
of SIP or H.323 in a large, managed network. Some comments:

I've heard of speech clipping (on answer), but I don't know what 
"ring-clipping" is. I presume you are referring to call setup delay (or delay to ring). 
The issue is signaling delay in general (call setup and release), but speech 
clipping has always been the most serious impact of delay.

It seems to me that this issue is really not related to "interfacing to a TDM 
telephony switch", but occurs equally within a fully IP environment.

No matter what service class is used for signaling (above, below, or same as 
speech bearer) there must be guaranteed bandwidth for the expected signaling 
traffic to provide sufficient low latency and almost zero discard rate 
independent from the amount of speech bearer traffic. In effect, if signaling is in a 
separate class and is guaranteed, say 5% of the bandwidth on an outgoing link 
and speech bearer (EF) is guaranteeed 50%, then it isn't really necessary to 
think of one being above or below or the same as the other. The forwarding 
algorithm guarantees a minimum bandwidth to both in what is effectively 
"time-division multiplexing", but we don't call it that since it is not based on strict 
timing. The result is that the signaling does not experience "higher latency", 
but probably has "higher delay variation", mostly due to the varying packet 
sizes.

The problem with putting signaling in the same class (queue) as speech bearer 
seems to be that the proper management of the speech queue depends partly on 
all of the packets in the queue being roughly the same size. While packets 
from codecs are something like 20-40 bytes, signaling messages, especially SIP, 
can get very long and would require segmentation. If they are in the same queue 
as speech, it doesn't seem that they can be segmented to allow speech packets 
to be transmitted between the segments.

In addition, there must be different discard policies applied to signaling 
packets than for speech packets. While discard of speech packets is normal and 
expected under overload (queue overflow), signaling packets must not be 
discarded.

Mike Pierce


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<HTML><FONT FACE=3Darial,helvetica><FONT  SIZE=3D2 PTSIZE=3D10>In a message=20=
dated 5/27/2004 9:07:27 AM Eastern Standard Time, babiarz@nortelnetworks.com=
 writes:
<BR>
<BR>
<BR><BLOCKQUOTE TYPE=3DCITE style=3D"BORDER-LEFT: #0000ff 2px solid; MARGIN-=
LEFT: 5px; MARGIN-RIGHT: 0px; PADDING-LEFT: 5px">In the next revision (03) o=
f the draft (should be put on the mailing list shortly) we state that peer-t=
o-peer signaling such as SIP and H.323 be forwarded out of a different servi=
ce class than telephony payload. I will analyze the impact on speech and rin=
g clipping at beginning of call when interfacing to a TDM telephony switch i=
f all telephony signaling would be forwarded using a different queue (higher=
 latency) than the voice bearer. My motivation for mapping telephony signali=
ng in to the same queue as voice bearer was to eliminate the delta delay bet=
ween audio and signaling, therefore reducing the chance of ring and speech c=
lipping at beginning of call. Other suggestions to address this problem are=20=
welcome.
<BR>In the draft we recommend that for Telephony service, per DSCP traffic=20
<BR></BLOCKQUOTE>
<BR>
<BR>
<BR>I believe this remains as one of the major open questions for the applic=
ation of SIP or H.323 in a large, managed network. Some comments:
<BR>
<BR>I've heard of speech clipping (on answer), but I don't know what "ring-c=
lipping" is. I presume you are referring to call setup delay (or delay to ri=
ng). The issue is signaling delay in general (call setup and release), but s=
peech clipping has always been the most serious impact of delay.
<BR>
<BR>It seems to me that this issue is really not related to "interfacing to=20=
a TDM telephony switch", but occurs equally within a fully IP environment.
<BR>
<BR>No matter what service class is used for signaling (above, below, or sam=
e as speech bearer) there must be guaranteed bandwidth for the expected sign=
aling traffic to provide sufficient low latency and almost zero discard rate=
 independent from the amount of speech bearer traffic. In effect, if signali=
ng is in a separate class and is guaranteed, say 5% of the bandwidth on an o=
utgoing link and speech bearer (EF) is guaranteeed 50%, then it isn't really=
 necessary to think of one being above or below or the same as the other. Th=
e forwarding algorithm guarantees a minimum bandwidth to both in what is eff=
ectively "time-division multiplexing", but we don't call it that since it is=
 not based on strict timing. The result is that the signaling does not exper=
ience "higher latency", but probably has "higher delay variation", mostly du=
e to the varying packet sizes.
<BR>
<BR>The problem with putting signaling in the same class (queue) as speech b=
earer seems to be that the proper management of the speech queue depends par=
tly on all of the packets in the queue being roughly the same size. While pa=
ckets from codecs are something like 20-40 bytes, signaling messages, especi=
ally SIP, can get very long and would require segmentation. If they are in t=
he same queue as speech, it doesn't seem that they can be segmented to allow=
 speech packets to be transmitted between the segments.
<BR>
<BR>In addition, there must be different discard policies applied to signali=
ng packets than for speech packets. While discard of speech packets is norma=
l and expected under overload (queue overflow), signaling packets must not b=
e discarded.
<BR>
<BR>Mike Pierce
<BR></FONT></HTML>

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Date: Fri, 28 May 2004 08:15:08 -0400
To: Mpierce1@aol.com
From: Kwok Ho Chan <khchan@nortelnetworks.com>
Subject: Re: [Tsvwg] draft-baker-diffserv-basic-classes-02
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References: <194.293a358c.2de8017b@aol.com>
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Cc: Jozef Babiarz <babiarz@nortelnetworks.com>, khchan@nortelnetworks.com,
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The co-authors have considered the variable packet sizes of the signaling 
packets as
compared to the more fixed size of the speech packets.  Especially wrt SIP 
and its
variable payload.
Hence in 
http://www.ietf.org/internet-drafts/draft-baker-diffserv-basic-classes-02.txt
Section 4.1.1 on page 23 where CS5 marking recommendation for telephony 
service class
excludes SIP.
And have SIP be in Low Latency Data Service Class described in Section 
4.2.2 on page 30.
One of the reason for this is because of increasing use of SIP as a 
transport for many things
(some encripted).
IMHO, I think some of the concerns of this E-Mail thread are alone the same 
line.
Do you think the current text in those sections are sufficient to resolve 
the issue?
Thank you very much for all the comments.
-- Kwok Ho Chan --


At 10:44 PM 5/27/2004, Mpierce1@aol.com wrote:
>In a message dated 5/27/2004 9:07:27 AM Eastern Standard Time, 
>babiarz@nortelnetworks.com writes:
>
>
>>In the next revision (03) of the draft (should be put on the mailing list 
>>shortly) we state that peer-to-peer signaling such as SIP and H.323 be 
>>forwarded out of a different service class than telephony payload. I will 
>>analyze the impact on speech and ring clipping at beginning of call when 
>>interfacing to a TDM telephony switch if all telephony signaling would be 
>>forwarded using a different queue (higher latency) than the voice bearer. 
>>My motivation for mapping telephony signaling in to the same queue as 
>>voice bearer was to eliminate the delta delay between audio and 
>>signaling, therefore reducing the chance of ring and speech clipping at 
>>beginning of call. Other suggestions to address this problem are welcome.
>>In the draft we recommend that for Telephony service, per DSCP traffic
>
>
>
>I believe this remains as one of the major open questions for the 
>application of SIP or H.323 in a large, managed network. Some comments:
>
>I've heard of speech clipping (on answer), but I don't know what 
>"ring-clipping" is. I presume you are referring to call setup delay (or 
>delay to ring). The issue is signaling delay in general (call setup and 
>release), but speech clipping has always been the most serious impact of 
>delay.
>
>It seems to me that this issue is really not related to "interfacing to a 
>TDM telephony switch", but occurs equally within a fully IP environment.
>
>No matter what service class is used for signaling (above, below, or same 
>as speech bearer) there must be guaranteed bandwidth for the expected 
>signaling traffic to provide sufficient low latency and almost zero 
>discard rate independent from the amount of speech bearer traffic. In 
>effect, if signaling is in a separate class and is guaranteed, say 5% of 
>the bandwidth on an outgoing link and speech bearer (EF) is guaranteeed 
>50%, then it isn't really necessary to think of one being above or below 
>or the same as the other. The forwarding algorithm guarantees a minimum 
>bandwidth to both in what is effectively "time-division multiplexing", but 
>we don't call it that since it is not based on strict timing. The result 
>is that the signaling does not experience "higher latency", but probably 
>has "higher delay variation", mostly due to the varying packet sizes.
>
>The problem with putting signaling in the same class (queue) as speech 
>bearer seems to be that the proper management of the speech queue depends 
>partly on all of the packets in the queue being roughly the same size. 
>While packets from codecs are something like 20-40 bytes, signaling 
>messages, especially SIP, can get very long and would require 
>segmentation. If they are in the same queue as speech, it doesn't seem 
>that they can be segmented to allow speech packets to be transmitted 
>between the segments.
>
>In addition, there must be different discard policies applied to signaling 
>packets than for speech packets. While discard of speech packets is normal 
>and expected under overload (queue overflow), signaling packets must not 
>be discarded.
>
>Mike Pierce
>_______________________________________________
>tsvwg mailing list
>tsvwg@ietf.org
>https://www1.ietf.org/mailman/listinfo/tsvwg

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<html>
<body>
The co-authors have considered the variable packet sizes of the signaling
packets as<br>
compared to the more fixed size of the speech packets.&nbsp; Especially
wrt SIP and its<br>
variable payload.<br>
Hence in
<a href="http://www.ietf.org/internet-drafts/draft-baker-diffserv-basic-classes-02.txt" eudora="autourl">http://www.ietf.org/internet-drafts/draft-baker-diffserv-basic-classes-02.txt</a><br>
Section 4.1.1 on page 23 where CS5 marking recommendation for telephony
service class<br>
excludes SIP.<br>
And have SIP be in Low Latency Data Service Class described in Section
4.2.2 on page 30.<br>
One of the reason for this is because of increasing use of SIP as a
transport for many things<br>
(some encripted).<br>
IMHO, I think some of the concerns of this E-Mail thread are alone the
same line.<br>
Do you think the current text in those sections are sufficient to resolve
the issue?<br>
Thank you very much for all the comments.<br>
-- Kwok Ho Chan --<br><br>
<br>
At 10:44 PM 5/27/2004, Mpierce1@aol.com wrote:<br>
<blockquote type=cite class=cite cite><font face="arial" size=2>In a
message dated 5/27/2004 9:07:27 AM Eastern Standard Time,
babiarz@nortelnetworks.com writes: <br><br>
<br>
<blockquote type=cite class=cite cite>In the next revision (03) of the
draft (should be put on the mailing list shortly) we state that
peer-to-peer signaling such as SIP and H.323 be forwarded out of a
different service class than telephony payload. I will analyze the impact
on speech and ring clipping at beginning of call when interfacing to a
TDM telephony switch if all telephony signaling would be forwarded using
a different queue (higher latency) than the voice bearer. My motivation
for mapping telephony signaling in to the same queue as voice bearer was
to eliminate the delta delay between audio and signaling, therefore
reducing the chance of ring and speech clipping at beginning of call.
Other suggestions to address this problem are welcome. <br>
In the draft we recommend that for Telephony service, per DSCP traffic
</blockquote><br><br>
<br>
I believe this remains as one of the major open questions for the
application of SIP or H.323 in a large, managed network. Some comments:
<br><br>
I've heard of speech clipping (on answer), but I don't know what
&quot;ring-clipping&quot; is. I presume you are referring to call setup
delay (or delay to ring). The issue is signaling delay in general (call
setup and release), but speech clipping has always been the most serious
impact of delay. <br><br>
It seems to me that this issue is really not related to &quot;interfacing
to a TDM telephony switch&quot;, but occurs equally within a fully IP
environment. <br><br>
No matter what service class is used for signaling (above, below, or same
as speech bearer) there must be guaranteed bandwidth for the expected
signaling traffic to provide sufficient low latency and almost zero
discard rate independent from the amount of speech bearer traffic. In
effect, if signaling is in a separate class and is guaranteed, say 5% of
the bandwidth on an outgoing link and speech bearer (EF) is guaranteeed
50%, then it isn't really necessary to think of one being above or below
or the same as the other. The forwarding algorithm guarantees a minimum
bandwidth to both in what is effectively &quot;time-division
multiplexing&quot;, but we don't call it that since it is not based on
strict timing. The result is that the signaling does not experience
&quot;higher latency&quot;, but probably has &quot;higher delay
variation&quot;, mostly due to the varying packet sizes. <br><br>
The problem with putting signaling in the same class (queue) as speech
bearer seems to be that the proper management of the speech queue depends
partly on all of the packets in the queue being roughly the same size.
While packets from codecs are something like 20-40 bytes, signaling
messages, especially SIP, can get very long and would require
segmentation. If they are in the same queue as speech, it doesn't seem
that they can be segmented to allow speech packets to be transmitted
between the segments. <br><br>
In addition, there must be different discard policies applied to
signaling packets than for speech packets. While discard of speech
packets is normal and expected under overload (queue overflow), signaling
packets must not be discarded. <br><br>
Mike Pierce <br>
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