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Date: Wed, 01 Dec 1999 17:10:17 -0500
From: GeeVarghese John <gvjohn@miel.mot.com>
Organization: Motorola India Electronics Ltd.
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To: Tmima Koren <tmima@cisco.com>, casner@cisco.com, micke@cdt.luth.se,
        brucet@cisco.com, dwing@cisco.com, pruddy@cisco.com, rem-conf@es.net,
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Subject: Re: Extensions to CRTP
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Tmima, team and Lars (and all u there in the rem list)

At this point one thing came to mind I am just presenting here
Could you please commnet on this ..

Apart from F_HDR, C_RTP & C_UDP packet
if we can have C_UDP with IP id transmitted as it is in the IP/UDP/RTP
packet ..
we may have the following advantages ..

1) Avoiding negative cache :
Compression can be at 3 stage  first start compressing a flow  with
CRTP at this state  when this state when the flow is detected as too
much inconsistent,
compression state can be moved to C_UDP if still inconsistent move to
C_UDP with
IP id send as it is (Say C_UDPi) packet.
For the inconsistent flow C_UDPi state is the last state and   can avoid
moving
to negative cache ..Advantage of this mechanism is

2) We use heuristics to find RTP flow ..
Even if the flow identified is not RTP which can be started with
C_RTPcomperssion and
if the flow is found to be too much inconsistent  move to the
sucsseeding states.
Still we get a compression benifit .

3) It is the experiance that CRTP is more processor intensive ..
When there has to be a balance, C_UDPi packet sending need not be that
processor intensive
and can improve the throughput. Because here the delta encoding is less
..
So compressor can decide on whether to send only C_UDPi packet for
particular flows
or all flows.

4) "Extension to CRTP" draft talks on running CRTP algo over ATM links,
In that case ..
Is the sequence number space 1-16 sufficient ?
C_UDPi packet need not be sequentially aligned .. So we can avoid
looking at the sequence number.

Compression rate will be as follows

C_RTP -> 2 (or 4 ) bytes from 40 bytes
C_UDP-> 2 (or 4) from 28
C_UDPi -> 4 or (6) from 28 but there will not encoding.

I feel tmima's change to the draft should be to indicate an explicit IP
id value ..
and if people agree on these points you may append these points .
Yes, In my previous mail I suggested as having one bit to indicate
deltaT and IP id.

Please comment ....

Rgds
GVJ

GeeVarghese John wrote:

> Tmima,
>
> Thanks for the reply and I have added some more comments on the CU+
> packets in this
> mail. I read the first mail from "Lars-Erik Jonsson", and your reply
> to that, Good
> you are changing the CU+ with IP ID but you may not need to set M bit
> there,
> by settting deltaT bit you can encode both deltaT and IP ID, because
> you need both values...
> I have added whatever comments I have about CU+,  please read that
> below
> According to me CU+ with IP ID also will not give a robust solution.
> Reason stated below.
>
> I think we should windup our discussion on CU + with this mail unless
> someone else is
> interested. If you want to reply to me please ..If you want more
> clarrification on my point
> I will reply back.
>
> My next mail will address few things about ACCEPT/REJECT
> packet use, you & team proposed. Please reply to that mail.
>
> My comments on CU+ are :
>
> Point 1.1:
>
> IP ID can be assigned by an host in three ways
> a) sequence number in ascending order with variace 1
> b) random number
> c) sequence number with variance n (n >1)
>
> I am not aware of any other implementation ..(May be possible)
> IF the IP id variance is not in the ascending order,
> CU+ will have problem. You expect that, host will implement this
> while transmitting the RTP packets. I dont know how many (non
> compliant) people
> will change this for the new proposal came in "extending CRTP".
> Or you are agreed for IP ID in CU+, it is a positive move.
>
> Point # 1.2
>
> Infact looking at the sequence number of CUDP packets helps to  get
> rid of
> IP id sync problem which we discussed .  (What I meant is if CUDP came
> in with a
> non expected link seqnumber, decompressor send out a CS packet to
> compressor to synchronise the context)
> This will take care all issues with the IP id variance (variance in
> any manner random, ascending order).
> If we extend CRTP with the new extension you proposed,
> we may loose this synchronization chance between compressor and
> decompressor,
> because we are NOT expected to look at CUDP+ (or CU+) link sequence
> number
> to synchronize anymore as far as IP ID variance is concerned.
> Otherwise CU+ packet should always carry IP ID field then we can avoid
>
> looking at sequnce number but not in the case with CU packet.
>
> What is your comment about CUDP with extension co-exist?
> Should it look for link seq number match further also  ?
>
> Point #2
>
> One solution to IP ID prolem you stated as "Twics Algo"
> Solution to this problem could be using twice algorithm.
> But "twice" algorithm is not mandatory. There are implementations
> without preserving UDP checksum. In this case I dont have an answer.
> May be we have to force a standard that everyone should implement
> UDP checksum while transmitting out. I dont know how many people will
> do that.
>
> Point #3.1
>
> In our discussion we have addressed two kinds of packet loss
>
> 1)  RTP packet loss in the network before reaching compessor (Say L1)
> 2) Compressed packet loss sent from compressor to decompressor (Say
> L2)
>
>
> In a lossy environment (L1), we end up in sending more number of
> encoded bytes
> with CU+ than what we used to send with CR+.
> Because in CU+ we encode deltaT of the next expected RTP packet, with
> the current packet.
> I can understand your point that source generate with same periodic
> time gap for
> each packet within a talkburst. But the carrier network is IP datagram
> service and
> (+ link lossy nature). So should we rely on the possible network
> behaviour or should we, on the source behaviour.
>
> If L2 is greater than L1 then you claimed advantage of CU+ sync
> between compressor
> and decompressor can be achieved.
>
> But What is the likely chance for L2 >  L1 ?
> Only for some specific lossy links.
>
> Point # 3.2
>
> As far as I know CRTP use some heuristic mechanism to identify a RTP
> flow.
> It is not a definite mechanism. In this case  does it make sense to
> send the deltaT of the next expected packet. Consider L1 case
> mentioned above.
> Not only that, assume the time gap between talkburst is (T1) and time
> gap between
> inter packet arrival of same session is T2. Consider a case of,
> If we have identified a non-RTP flow as RTP flow and started
> compression
> with our heuristics how do we determine when to send CU+.
> Will we end up in sending continuous CU+ ?  we may ..
> There need not be talkburst gap with some what high definite gap
> when  compare to interpacket gap of the same session...
> (See Pt # 4 on CU+ size comparison with CR+)
>
>
> Point #4
>
> Size of the CU+ is more than CR+ because of the compressed RTP header
> ..
> in CR+. In CU+ you send complete RTP header as it is (i.e 12 byts).
> In old CRTP, Assume that for some situation we have to encode RTP seq
> number and
> Timestamp, For ideal case encoded byte size will be 2 byts (1 for seq
> number
> + 1 for deltaT) or otherwise 4 or max 6 (still less than 12 bytes).
> Over a lossy link (L2), which is particular to the link, advantage of
> CU+ is missing.
>
> Continuous sending of CU+ for a talk burst may avoid send CS from
> decompressor to
> compressor. But What kind of loss is more likey L1 or L2.
>
>
> My personal comment to CU+ is
>
> May be we can claim some advantage only if
> the stated assumptions (about packet loss (L1 < L2),  twice algo..etc)
> are met ..
> That also with the modification of IP ID inside the CU+  packet.
>
> Thanks & Rgds
>
> GVJ
>
> Tmima Koren wrote:
>
>>  At 10:17 PM 11/23/99 -0500, GeeVarghese John wrote:
>>
>> >
>> >
>> >> >> >
>> >> >> > The decompressor will stay in sync as long as he received at
>> >> >> > least one of the CU+
>> >> >>
>> >> > I think any of the CU+ loss will lead to CONTEXT_STATE packet
>> >> > generation from
>> >> > decompressor to compressor because of the link seq number
>> >> > mismatch ..Is nt it ?
>> >>
>> >>  The decompressor should be in sync after receiving a CU+. There
>> >>  is no reason for a CS
>> >>  Tmima
>> >
>> > Tmima,
>> >
>> > Could you please explain, how IP id variance should be treated in
>> > this case ?
>> > For IP id field  first order difference is sent as delta in CUDP
>> > with I bit set.
>> > In this case, if there is a CU+ packet loss how IP ID delta
>> > encoding should be treated ..
>>
>>
>> Usually the IP id will not increment uniformly in packets of an RTP
>> stream (the host sends packets other then the RTP streams, silence
>> periods).
>> This means that if we want to compress a stream, we might have to
>> send a new deltaID often, maybe in each packet. This will increase
>> the size of the packets.
>> To optimize the compression, the host can assign IP id's to the
>> packets of each stream in a constant increment.
>> This keeps the deltaID constant for each stream.
>>
>> The advantage of a CU+ packet is that in one packet we restore all
>> RTP-related parameters at the decompressor.
>> The sequence of:  CU followed by a CR+  is very vulnerable: the CU
>> zeroes the deltaT, so if the CR+ was lost, the decompressor is out
>> of sync.
>> The CU+ does not restore the IP id. But if the implementation keeps
>> the deltaID constant..
>>
>> After a silence period, the sequence number will still increment by
>> 1 and the IP id by deltaID: there is no reason to skip RTP sequence
>> numbers or IP id's. deltaT is the only difference that is not
>> correct for the first packet after a silence period.
>>
>> To optimize CRTP over lossy links
>> 1. Use a constant deltaID
>> 2. Check udp checksum
>> 3. Use CU+ as necessary
>>
>>
>>
>> > and also I am not still convinced with the advantage of CU+ over
>> > CR+ after
>> > the  silence period. Could you please explain your point  ..(please
>> > ref my previous mail)
>>
>>
>> Let's say that the characteristics of the link is that we may lose
>> at most 1 packet in a row
>> If we use:
>>
>> 11      200       CR+    dt=100
>> 12      210       CR+    dt=10
>> 13      220       CR
>>
>> If 11 is lost, the decompressor does not have the correct RTP
>> timestamp for packet 12.
>> If 12 is lost, the decompressor has a wrong deltaT.
>> In both cases the decompressor is out of sync.
>> This is a very weak sequence to use after a silence period.
>>
>> If instead we use:
>>
>> 11"    200       CU+    dt=10
>> 12"    210       CU+    dt=10
>> 13"    220       CR
>>
>> In this case If one of 11" or 12" is lost, the decompressor remains
>> in sync. So this is a very strong sequence
>>
>> Tmima
>>
>>
>>
>> > Please reply .
>> >
>> > Rgds
>> >
>> > GVJ
>>
> -




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<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Tmima, team and Lars (and all u there in the rem list)
<p>At this point one thing came to mind I am just presenting here
<br>Could you please commnet on this ..
<p>Apart from F_HDR, C_RTP &amp; C_UDP packet
<br>if we can have C_UDP with IP id transmitted as it is in the IP/UDP/RTP
packet ..
<br>we may have the following advantages ..
<p>1) Avoiding negative cache :
<br>Compression can be at 3 stage&nbsp; first start compressing a flow&nbsp;
with
<br>CRTP at this state&nbsp; when this state when the flow is detected
as too much inconsistent,
<br>compression state can be moved to C_UDP if still inconsistent move
to C_UDP with
<br>IP id send as it is (Say C_UDPi) packet.
<br>For the inconsistent flow C_UDPi state is the last state and&nbsp;&nbsp;
can avoid moving
<br>to negative cache ..Advantage of this mechanism is
<p>2) We use heuristics to find RTP flow ..
<br>Even if the flow identified is not RTP which can be started with C_RTPcomperssion
and
<br>if the flow is found to be too much inconsistent&nbsp; move to the
sucsseeding states.
<br>Still we get a compression benifit .
<p>3) It is the experiance that CRTP is more processor intensive ..
<br>When there has to be a balance, C_UDPi packet sending need not be that
processor intensive
<br>and can improve the throughput. Because here the delta encoding is
less ..
<br>So compressor can decide on whether to send only C_UDPi packet for&nbsp;
particular flows
<br>or all flows.
<p>4) "Extension to CRTP" draft talks on running CRTP algo over ATM links,
In that case ..
<br>Is the sequence number space 1-16 sufficient ?
<br>C_UDPi packet need not be sequentially aligned .. So we can avoid looking
at the sequence number.
<p>Compression rate will be as follows
<p>C_RTP -> 2 (or 4 ) bytes from 40 bytes
<br>C_UDP-> 2 (or 4) from 28
<br>C_UDPi -> 4 or (6) from 28 but there will not encoding.
<p>I feel tmima's change to the draft should be to indicate an explicit
IP id value ..
<br>and if people agree on these points you may append these points .
<br>Yes, In my previous mail I suggested as having one bit to indicate
deltaT and IP id.
<p>Please comment ....
<p>Rgds
<br>GVJ
<p>GeeVarghese John wrote:
<blockquote TYPE=CITE>Tmima,
<p>Thanks for the reply and I have added some more comments on the CU+
packets in this
<br>mail. I read the first mail from "Lars-Erik Jonsson", and your reply
to that, Good
<br>you are changing the CU+ with IP ID but you may not need to set M bit
there,
<br>by settting deltaT bit you can encode both deltaT and IP ID, because
you need both values...
<br><b>I have added whatever comments I have about CU+,&nbsp; please read
that below</b>
<br><b>According to me CU+ with IP ID also will not give a robust solution.</b>
<br><b>Reason stated below.</b>
<p>I think we should windup our discussion on CU + with this mail unless
someone else is
<br>interested. If you want to reply to me please ..If you want more clarrification
on my point
<br>I will reply back.
<p>My next mail will address few things about ACCEPT/REJECT
<br>packet use, you &amp; team proposed. Please reply to that mail.
<p>My comments on CU+ are :
<p><b>Point 1.1:</b>
<p>IP ID can be assigned by an host in three ways
<br>a) sequence number in ascending order with variace 1
<br>b) random number
<br>c) sequence number with variance n (n >1)
<p>I am not aware of any other implementation ..(May be possible)
<br>IF the IP id variance is not in the ascending order,
<br>CU+ will have problem. You expect that, host will implement this
<br>while transmitting the RTP packets. I dont know how many (non compliant)
people
<br>will change this for the new proposal came in "extending CRTP".
<br>Or you are agreed for IP ID in CU+, it is a positive move.
<p><b>Point # 1.2</b>
<p>Infact looking at the sequence number of CUDP packets helps to&nbsp;
get rid of
<br>IP id sync problem which we discussed .&nbsp; (What I meant is if CUDP
came in with a
<br>non expected link seqnumber, decompressor send out a CS packet to
<br>compressor to synchronise the context)
<br>This will take care all issues with the IP id variance (variance in
any manner random, ascending order).
<br>If we extend CRTP with the new extension you proposed,
<br>we may loose this synchronization chance between compressor and decompressor,
<br>because we are NOT expected to look at CUDP+ (or CU+) link sequence
number
<br>to synchronize anymore as far as IP ID variance is concerned.
<br>Otherwise CU+ packet should always carry IP ID field then we can avoid
<br>looking at sequnce number but not in the case with CU packet.
<p>What is your comment about CUDP with extension co-exist?
<br>Should it look for link seq number match further also&nbsp; ?
<p><b>Point #2</b>
<p>One solution to IP ID prolem you stated as "Twics Algo"
<br>Solution to this problem could be using twice algorithm.
<br>But "twice" algorithm is not mandatory. There are implementations
<br>without preserving UDP checksum. In this case I dont have an answer.
<br>May be we have to force a standard that everyone should implement
<br>UDP checksum while transmitting out. I dont know how many people will
do that.
<p><b>Point #3.1</b>
<p>In our discussion we have addressed two kinds of packet loss
<p>1)&nbsp; RTP packet loss in the network before reaching compessor (Say
L1)
<br>2) Compressed packet loss sent from compressor to decompressor (Say
L2)
<br>&nbsp;
<p>In a lossy environment (L1), we end up in sending more number of encoded
bytes
<br>with CU+ than what we used to send with CR+.
<br>Because in CU+ we encode deltaT of the next expected RTP packet, with
the current packet.
<br>I can understand your point that source generate with same periodic
time gap for
<br>each packet within a talkburst. But the carrier network is IP datagram
service and
<br>(+ link lossy nature). So should we rely on the possible network behaviour
or should we, on the source behaviour.
<p>If L2 is greater than L1 then you claimed advantage of CU+ sync between
compressor
<br>and decompressor can be achieved.
<p>But What is the likely chance for L2 >&nbsp; L1 ?
<br>Only for some specific lossy links.
<p><b>Point # 3.2</b>
<p>As far as I know CRTP use some heuristic mechanism to identify a RTP
flow.
<br>It is not a definite mechanism. In this case&nbsp; does it make sense
to
<br>send the deltaT of the next expected packet. Consider L1 case mentioned
above.
<br>Not only that, assume the time gap between talkburst is (T1) and time
gap between
<br>inter packet arrival of same session is T2. Consider a case of,
<br>If we have identified a non-RTP flow as RTP flow and started compression
<br>with our heuristics how do we determine when to send CU+.
<br>Will we end up in sending continuous CU+ ?&nbsp; we may ..
<br>There need not be talkburst gap with some what high definite gap
<br>when&nbsp; compare to interpacket gap of the same session...
<br>(See Pt # 4 on CU+ size comparison with CR+)
<br>&nbsp;
<p><b>Point #4</b>
<p>Size of the CU+ is more than CR+ because of the compressed RTP header
..
<br>in CR+. In CU+ you send complete RTP header as it is (i.e 12 byts).
<br>In old CRTP, Assume that for some situation we have to encode RTP seq
number and
<br>Timestamp, For ideal case encoded byte size will be 2 byts (1 for seq
number
<br>+ 1 for deltaT) or otherwise 4 or max 6 (still less than 12 bytes).
<br>Over a lossy link (L2), which is particular to the link, advantage
of CU+ is missing.
<p>Continuous sending of CU+ for a talk burst may avoid send CS from decompressor
to
<br>compressor. But What kind of loss is more likey L1 or L2.
<br>&nbsp;
<p>My personal comment to CU+ is
<p>May be we can claim some advantage only if
<br>the stated assumptions (about packet loss (L1 &lt; L2),&nbsp; twice
algo..etc) are met ..
<br>That also with the modification of IP ID inside the CU+&nbsp; packet.
<p>Thanks &amp; Rgds
<p>GVJ
<p>Tmima Koren wrote:
<blockquote TYPE=CITE>&nbsp;At 10:17 PM 11/23/99 -0500, GeeVarghese John
wrote:
<blockquote type=cite cite>&nbsp;
<blockquote type=cite cite>
<blockquote type=cite cite>
<blockquote type=cite cite>
<blockquote type=cite cite>&nbsp;
<br>The decompressor will stay in sync as long as he received at least
one of the CU+</blockquote>
</blockquote>
I think any of the CU+ loss will lead to CONTEXT_STATE packet generation
from
<br>decompressor to compressor because of the link seq number mismatch
..Is nt it ?</blockquote>
The decompressor should be in sync after receiving a CU+. There is no reason
for a CS
<br>Tmima</blockquote>
Tmima,
<p>Could you please explain, how IP id variance should be treated in this
case ?
<br>For IP id field&nbsp; first order difference is sent as delta in CUDP
with I bit set.
<br>In this case, if there is a CU+ packet loss how IP ID delta encoding
should be treated ..</blockquote>

<p><br>Usually the IP id will not increment uniformly in packets of an
RTP stream (the host sends packets other then the RTP streams, silence
periods).
<br>This means that if we want to compress a stream, we might have to send
a new deltaID often, maybe in each packet. This will increase the size
of the packets.
<br>To optimize the compression, the host can assign IP id's to the packets
of each stream in a constant increment.
<br>This keeps the deltaID constant for each stream.
<p>The advantage of a CU+ packet is that in one packet we restore all RTP-related
parameters at the decompressor.
<br>The sequence of:&nbsp; CU followed by a CR+&nbsp; is very vulnerable:
the CU zeroes the deltaT, so if the CR+ was lost, the decompressor is out
of sync.
<br>The CU+ does not restore the IP id. But if the implementation keeps
the deltaID constant..
<p>After a silence period, the sequence number will still increment by
1 and the IP id by deltaID: there is no reason to skip RTP sequence numbers
or IP id's. deltaT is the only difference that is not correct for the first
packet after a silence period.
<p>To optimize CRTP over lossy links
<br>1. Use a constant deltaID
<br>2. Check udp checksum
<br>3. Use CU+ as necessary
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>and also I am not still convinced with the advantage
of CU+ over CR+ after
<br>the&nbsp; silence period. Could you please explain your point&nbsp;
..(please ref my previous mail)</blockquote>

<p><br>Let's say that the characteristics of the link is that we may lose
at most 1 packet in a row
<br>If we use:
<p>11&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=100
<br>12&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR+&nbsp;&nbsp;&nbsp; dt=10
<br>13&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
CR
<p>If 11 is lost, the decompressor does not have the correct RTP timestamp
for packet 12.
<br>If 12 is lost, the decompressor has a wrong deltaT.
<br>In both cases the decompressor is out of sync.
<br>This is a very weak sequence to use after a silence period.
<p>If instead we use:
<p>11"&nbsp;&nbsp;&nbsp; 200&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>12"&nbsp;&nbsp;&nbsp; 210&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CU+&nbsp;&nbsp;&nbsp;
dt=10
<br>13"&nbsp;&nbsp;&nbsp; 220&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; CR
<p>In this case If one of 11" or 12" is lost, the decompressor remains
in sync. So this is a very strong sequence
<p>Tmima
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>Please reply .
<p>Rgds
<p>GVJ</blockquote>
</blockquote>
-</blockquote>

<br>&nbsp;
<br>&nbsp;</html>

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From: Tmima Koren <tmima@cisco.com>
Subject: CRTP and IP ID
In-Reply-To: <38459CC9.EE4B28BB@miel.mot.com>
References: <4.1.19991118024030.01aeaa50@jindo.cisco.com>
 <38358615.D8679DA0@miel.mot.com>
 <4.1.19991122104601.038605e0@jindo.cisco.com>
 <4.1.19991124103940.01ec4b60@jindo.cisco.com>
 <383DC57E.9D1FAA52@miel.mot.com>
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When the IP ID is changing arbitrarily, we have to change the deltaID very frequently - probably in every packet.
In such cases it might be more useful to include the full IP ID in a CR instead of the deltaID.
This can be done if we could change the meaning of the 'I' flag in the CR packet to mean either full IP ID or deltaID.
Maybe we can add an option to the CRTP negotiation about the usage of the 'I' flag in CR: full IP ID or deltaID
Then if a host requires preserving the IP ID and cannot assign them sequentially for the compressed RTP streams, the host will negotiate the 'I' flag in CR packets to mean full IP ID.
Or if we want this to be per stream, add a flag in the FH.

Are there cases when it's ok for the CRTP compressor to reasign the IP ID?

Tmima 



From rem-conf Thu Dec 02 02:05:14 1999 
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From: =?iso-8859-1?Q?Gunnar_Hellstr=F6m?= <Gunnar.Hellstrom@omnitor.se>
To: <rem-conf@es.net>
Subject: RE: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps 
Date: Thu, 2 Dec 1999 10:58:32 +0100
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A question on tones.
In the draft-ietf-avt-tones-03.txt, a set of modem and fax related tones =
are
specified.

I miss two modem related signals used by calling modems in the same way a=
s
the faxes use the CNG tone:

The V.25 CT, Calling tone. 1300 Hz for 0.5 s with 2 s. intervals ( more
precisely defined in V.25).
All modems not starting with V.8 CI starts with this tone nowadays (at le=
ast
in Europe).

The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could be
tranmitted as V.21 tones, but it might be of interest to have a specific
code for it with a parameter for the data contents of the CI.

Since their inclusion would complete the modem handshake support in the
tones spec, I suggest that they are added.

Regards
______________________________________________
Gunnar Hellstr=F6m
Omnitor AB
Alsn=F6gatan 7 ,  4 tr
S-116 41 Stockholm
Sweden

Tel +46 8 556 002 03
Mob: + 46 708 204 288
Txt and video +46 8 556 002 05
Fax +46 8 556 002 06

e-mail: gunnar.hellstrom@omnitor.se
web:    www.omnitor.se

> -----Original Message-----
> From: nsyracus@cnri.reston.va.us [mailto:nsyracus@cnri.reston.va.us]On
> Behalf Of Internet-Drafts@ietf.org
> Sent: Tuesday, November 30, 1999 12:57 PM
> To: IETF-Announce: ;
> Cc: rem-conf@es.net
> Subject: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
>
>
> A New Internet-Draft is available from the on-line
> Internet-Drafts directories.
> This draft is a work item of the Audio/Video Transport Working
> Group of the IETF.
>
> 	Title		: RTP Payload for DTMF Digits, Telephony Tones and
>                           Telephony Signals
> 	Author(s)	: H. Schulzrinne, S. Petrack
> 	Filename	: draft-ietf-avt-tones-03.txt,.ps
> 	Pages		: 28
> 	Date		: 29-Nov-99
>
> This memo describes how to carry dual-tone multifrequency (DTMF)
> signaling, other tone signals and telephony events in RTP packets.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-tones-03.txt
>
> Internet-Drafts are also available by anonymous FTP. Login with
> the username
> "anonymous" and a password of your e-mail address. After logging in,
> type "cd internet-drafts" and then
> 	"get draft-ietf-avt-tones-03.txt".
>
> A list of Internet-Drafts directories can be found in
> http://www.ietf.org/shadow.html
> or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
> Internet-Drafts can also be obtained by e-mail.
>
> Send a message to:
> 	mailserv@ietf.org.
> In the body type:
> 	"FILE /internet-drafts/draft-ietf-avt-tones-03.txt".
>
> NOTE:	The mail server at ietf.org can return the document in
> 	MIME-encoded form by using the "mpack" utility.  To use this
> 	feature, insert the command "ENCODING mime" before the "FILE"
> 	command.  To decode the response(s), you will need "munpack" or
> 	a MIME-compliant mail reader.  Different MIME-compliant mail readers
> 	exhibit different behavior, especially when dealing with
> 	"multipart" MIME messages (i.e. documents which have been split
> 	up into multiple messages), so check your local documentation on
> 	how to manipulate these messages.
>
>
> Below is the data which will enable a MIME compliant mail reader
> implementation to automatically retrieve the ASCII version of the
> Internet-Draft.
>




From rem-conf Thu Dec 02 06:15:56 1999 
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Date: Thu, 02 Dec 1999 19:39:10 -0500
From: GeeVarghese John <gvjohn@miel.mot.com>
Organization: Motorola India Electronics Ltd.
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To: Tmima Koren <tmima@cisco.com>
CC: casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
        pruddy@cisco.com, rem-conf@es.net, pazhynnr@cig.mot.com
Subject: Re: CRTP and IP ID
References: <4.1.19991118024030.01aeaa50@jindo.cisco.com>
	 <38358615.D8679DA0@miel.mot.com>
	 <4.1.19991122104601.038605e0@jindo.cisco.com>
	 <4.1.19991124103940.01ec4b60@jindo.cisco.com>
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Hi,

Tmima Koren wrote:

> When the IP ID is changing arbitrarily, we have to change the deltaID very frequently - probably in every packet.
> In such cases it might be more useful to include the full IP ID in a CR instead of the deltaID.

Exactly, you are correct ..
This is the thing I was mentioning by saying three stages of compression.
At the first stage we can start compressing a flow at CRTP then if we found the flow is
inconsistent then move the state to CUDP and send only CUDP packet then CUDPid packet.
Inconsistency can be interms of anything ..
One thing could be like what you said about IP id variance rate or other delta encoding rate or CONTEX_STATE packet received etc ..

>
> This can be done if we could change the meaning of the 'I' flag in the CR packet to mean either full IP ID or deltaID.
> Maybe we can add an option to the CRTP negotiation about the usage of the 'I' flag in CR: full IP ID or deltaID
> Then if a host requires preserving the IP ID and cannot assign them sequentially for the compressed RTP streams, the host will negotiate the 'I' flag in CR packets to mean full IP ID.
> Or if we want this to be per stream, add a flag in the FH.

It looks Very good. I am in favour of  negotation  at CRTP level not only for this point
I have something more on seq number extension also. May be later we will discuss this.
But Adding the information bit in the FH packet how do we ensure the reliable
delivery of the FH packet ..If Accept and Reject packet is the soln in your mind
That could be one possibility but Accept and Reject packet will have many other dsadvantes.
Instead of that, why cant we have a protocol negotiation packet which needs to be sent
only one time when the stack is initialised. For this packet we can have a design where
negotation parameters can be exteneded so that future requirement also will be taken care.



>
>
> Are there cases when it's ok for the CRTP compressor to reasign the IP ID?

Do you mean that CRTP will reassign a new IP id to the decompressed IP packet  ..?


Rgds
GeevJohn
--
-------------------------------------------------------------------------------
[ ]  Motorola Confidential Proprietory
[ ]  Motorola Internal Use Only
[x]  Not an MCP or MIU
-------------------------------------------------------------------------------
Geevarghese John,
DataCom Group,
Motorola India Electronics Ltd,
"The Senate", No-33 A,
Ulsoor Road,
Bangalore - 42.
Phone  : 91-80-5598615  Xnt-4005
E-mail : gvjohn@miel.mot.com
-------------------------------------------------------------------------------
-------------------------------------------------------------------------------







From rem-conf Thu Dec 02 16:04:00 1999 
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From rem-conf Fri Dec 03 05:54:22 1999 
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From: The IESG <iesg-secretary@ietf.org>
SUBJECT: Last Call: Real-Time Transport Protocol Management Information
	 Base to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Working
Group to consider Real-Time Transport Protocol Management Information
Base <draft-ietf-avt-rtp-mib-07.txt> as a Proposed Standard.

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by December 16, 1999.

Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mib-07.txt




From rem-conf Fri Dec 03 08:23:21 1999 
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To: rem-conf@es.net
Subject: RTP interoperability statements for draft standard
Date: Fri, 03 Dec 1999 16:19:49 +0000
Message-ID: <2929.944237989@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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To the AVT working group,

As noted in the AVT meeting at the Washington DC IETF, it is necessary to
collect interoperability reports for RTP implementations before we can
progress the RTP specification and audio/video profile to draft standard.
Two internet-drafts describe the type of interoperability statement required:
	draft-ietf-avt-rtp-interop-02.txt
	draft-ietf-avt-profile-interop-00.txt
and a third draft (draft-ietf-avt-rtptest-02.txt) describes possible
testing strategies.

This message is a plea for vendors with RTP implementations to make
available results of their interoperability tests, in order that we
can advance the specifications. Results of the tests to date are 
available from http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/RTPinterop/
Please send additions to this table to me (if you have a problem with
making this information publically available, please contact me privately
and we can discuss alternatives).

Thanks,
Colin



From rem-conf Fri Dec 03 14:23:52 1999 
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To: GeeVarghese John <gvjohn@miel.mot.com>, Tmima Koren <tmima@cisco.com>
From: Tmima Koren <tmima@cisco.com>
Subject: Re: CRTP and IP ID
Cc: casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
        pruddy@cisco.com, rem-conf@es.net, pazhynnr@cig.mot.com
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 <38358615.D8679DA0@miel.mot.com>
 <4.1.19991122104601.038605e0@jindo.cisco.com>
 <4.1.19991124103940.01ec4b60@jindo.cisco.com>
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At 07:39 PM 12/2/99 -0500, GeeVarghese John wrote:
>Hi,
>
>Tmima Koren wrote:
>
>> When the IP ID is changing arbitrarily, we have to change the deltaID very 
>frequently - probably in every packet.
>> In such cases it might be more useful to include the full IP ID in a CR 
>instead of the deltaID.
>
>Exactly, you are correct ..
>This is the thing I was mentioning by saying three stages of compression.
>At the first stage we can start compressing a flow at CRTP then if we found 
>the flow is
>inconsistent then move the state to CUDP and send only CUDP packet then 
>CUDPid packet.
>Inconsistency can be interms of anything ..
>One thing could be like what you said about IP id variance rate or other 
>delta encoding rate or CONTEX_STATE packet received etc ..

Yes. This is the same idea.
The CRid can be used when the RTP parameters are consistent, but the IP ID is not.

Mikael Degermark suggested to enhance the COMPRESSED_NON_TCP packet instead of the CUDP 
The COMPRESSED_NON_TCP packet includes the IP ID, the udp checksum (if nonzero) and also includes one byte of data which currently holds the 4-bit CRTP sequence# (see rfc2507). 2 of the 4 unused bits can be used to flag 'T' and 'I', so the packet may also include the deltaT and deltaID if necessary.
Such a packet will restore all parameters at the decompressor, same as CUDPid

>
>>
>> This can be done if we could change the meaning of the 'I' flag in the CR 
>packet to mean either full IP ID or deltaID.
>> Maybe we can add an option to the CRTP negotiation about the usage of the 
>'I' flag in CR: full IP ID or deltaID
>> Then if a host requires preserving the IP ID and cannot assign them 
>sequentially for the compressed RTP streams, the host will negotiate the 'I' 
>flag in CR packets to mean full IP ID.
>> Or if we want this to be per stream, add a flag in the FH.
>
>It looks Very good. I am in favour of  negotation  at CRTP level not only 
>for this point
>I have something more on seq number extension also. May be later we will 
>discuss this.
>But Adding the information bit in the FH packet how do we ensure the reliable
>delivery of the FH packet ..If Accept and Reject packet is the soln in your 
>mind
>That could be one possibility but Accept and Reject packet will have many 
>other dsadvantes.
>Instead of that, why cant we have a protocol negotiation packet which needs 
>to be sent
>only one time when the stack is initialised. For this packet we can have a 
>design where
>negotation parameters can be exteneded so that future requirement also will 
>be taken care.

I meant adding options to the IP header compression negotiation as described in rfc2509

>>
>>
>> Are there cases when it's ok for the CRTP compressor to reasign the IP ID?
>
>Do you mean that CRTP will reassign a new IP id to the decompressed IP 
>packet  ..?

Yes. I suggest a mode where CRTP does not preserve the IP ID.
Tmima


>
>
>Rgds
>GeevJohn
>--
>-------------------------------------------------------------------------------
>[ ]  Motorola Confidential Proprietory
>[ ]  Motorola Internal Use Only
>[x]  Not an MCP or MIU
>-------------------------------------------------------------------------------
>Geevarghese John,
>DataCom Group,
>Motorola India Electronics Ltd,
>"The Senate", No-33 A,
>Ulsoor Road,
>Bangalore - 42.
>Phone  : 91-80-5598615  Xnt-4005
>E-mail : gvjohn@miel.mot.com
>-------------------------------------------------------------------------------
>-------------------------------------------------------------------------------
>
>
>
>
>




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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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        "RTP interoperability statements for draft standard" (Dec  3,  4:19pm)
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Could someone please clarify the following "conflict" I see in the spec?

In the event of data loss, RFC 2250 has mechanisms to recover at the slice
level(basically by discarding RTP packets until you find one with the
BeginningOfSlice bit set to 1).

However, if you lose a few RTP packets, one is **really forced** to skip
until the beginning of the next GOP!

This is an incredible amount of loss or I have misunderstood the spec. Could
someone please clarify?


Details:
-------

Consider the following segment of MPEG-1 system stream.
...GOP_H P11 P12 P13 ... P1n GOP_H P21 P22 P23 ...

GOP_H refers to GOP Headers. Pij refers to picture j of the ith GOP.

For simplicity, assume that the stream and display order are the same. Thus,
Pij will have a "Temporal Reference" of j. For simplicity, assume each
picture is split into 5 RTP packets.

Suppose I receive RTP packets containing all of the first GOP header and
P11 and I receive 1 RTP packet containing the first "piece" of P12.

Suppose the next RTP packet I received contains the "Temporal Reference"
of 3, how do I know which GOP this picture belongs to. In other words, how
do I know it belongs to P13 or P23 or in general Pi3.

If I cannot know, I will have to insert a broken GOP thus essentially
losing data until the next GOP header is received.

Thanks for your help.
Ram

-- 
Ram Kordale
Advanced Media Products



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From rem-conf Sun Dec 05 09:24:51 1999 
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Date: Sun, 05 Dec 1999 18:39:14 +0100
To: Tmima Koren <tmima@cisco.com>, GeeVarghese John <gvjohn@miel.mot.com>,
        Tmima Koren <tmima@cisco.com>
From: Mikael Degermark <micke@cdt.luth.se>
Subject: Re: CRTP and IP ID
Cc: casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
        pruddy@cisco.com, rem-conf@es.net, pazhynnr@cig.mot.com
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I feel that I have to inform you people that when we were pushing 2507 and
2508 to
Proposed Standard, we originally proposed not to maintain the IP ID. The IESG
would not agree to that due to potential problems and we had to back off. 

Before going too far along the path of not maintaining the IP ID, I would
recommend
sounding out the IESG to see if they changed their minds. 

Mikael Degermark

At 02:13 PM 99/12/03 -0800, Tmima Koren wrote:
>At 07:39 PM 12/2/99 -0500, GeeVarghese John wrote:
>>Hi,
>>
>>Tmima Koren wrote:
>>
>>> When the IP ID is changing arbitrarily, we have to change the deltaID
very 
>>frequently - probably in every packet.
>>> In such cases it might be more useful to include the full IP ID in a CR 
>>instead of the deltaID.
>>
>>Exactly, you are correct ..
>>This is the thing I was mentioning by saying three stages of compression.
>>At the first stage we can start compressing a flow at CRTP then if we found 
>>the flow is
>>inconsistent then move the state to CUDP and send only CUDP packet then 
>>CUDPid packet.
>>Inconsistency can be interms of anything ..
>>One thing could be like what you said about IP id variance rate or other 
>>delta encoding rate or CONTEX_STATE packet received etc ..
>
>Yes. This is the same idea.
>The CRid can be used when the RTP parameters are consistent, but the IP ID
is not.
>
>Mikael Degermark suggested to enhance the COMPRESSED_NON_TCP packet
instead of the CUDP 
>The COMPRESSED_NON_TCP packet includes the IP ID, the udp checksum (if
nonzero) and also includes one byte of data which currently holds the 4-bit
CRTP sequence# (see rfc2507). 2 of the 4 unused bits can be used to flag
'T' and 'I', so the packet may also include the deltaT and deltaID if
necessary.
>Such a packet will restore all parameters at the decompressor, same as CUDPid
>
>>
>>>
>>> This can be done if we could change the meaning of the 'I' flag in the CR 
>>packet to mean either full IP ID or deltaID.
>>> Maybe we can add an option to the CRTP negotiation about the usage of the 
>>'I' flag in CR: full IP ID or deltaID
>>> Then if a host requires preserving the IP ID and cannot assign them 
>>sequentially for the compressed RTP streams, the host will negotiate the
'I' 
>>flag in CR packets to mean full IP ID.
>>> Or if we want this to be per stream, add a flag in the FH.
>>
>>It looks Very good. I am in favour of  negotation  at CRTP level not only 
>>for this point
>>I have something more on seq number extension also. May be later we will 
>>discuss this.
>>But Adding the information bit in the FH packet how do we ensure the
reliable
>>delivery of the FH packet ..If Accept and Reject packet is the soln in your 
>>mind
>>That could be one possibility but Accept and Reject packet will have many 
>>other dsadvantes.
>>Instead of that, why cant we have a protocol negotiation packet which needs 
>>to be sent
>>only one time when the stack is initialised. For this packet we can have a 
>>design where
>>negotation parameters can be exteneded so that future requirement also will 
>>be taken care.
>
>I meant adding options to the IP header compression negotiation as
described in rfc2509
>
>>>
>>>
>>> Are there cases when it's ok for the CRTP compressor to reasign the IP ID?
>>
>>Do you mean that CRTP will reassign a new IP id to the decompressed IP 
>>packet  ..?
>
>Yes. I suggest a mode where CRTP does not preserve the IP ID.
>Tmima
>
>
>>
>>
>>Rgds
>>GeevJohn
>>--
>>--------------------------------------------------------------------------
-----
>>[ ]  Motorola Confidential Proprietory

>>[ ]  Motorola Internal Use Only
>>[x]  Not an MCP or MIU
>>--------------------------------------------------------------------------
-----
>>Geevarghese John,
>>DataCom Group,
>>Motorola India Electronics Ltd,
>>"The Senate", No-33 A,
>>Ulsoor Road,
>>Bangalore - 42.
>>Phone  : 91-80-5598615  Xnt-4005
>>E-mail : gvjohn@miel.mot.com
>>--------------------------------------------------------------------------
-----
>>--------------------------------------------------------------------------
-----
>>
>>
>>
>>
>>
> 




From rem-conf Sun Dec 05 12:21:55 1999 
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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To: rem-conf@es.net
Subject: CFP: JCN Special Issue on QoS in IP Networks
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         JOURNAL OF COMMUNICATIONS AND NETWORKING (JCN)

 CALL FOR PAPERS - SPECIAL ISSUE ON QoS IN IP NETWORKS
                        JUNE, 2000

A Special Issue of JCN dedicated to the realization of QoS-sensitive
services in IP networks will be published in June, 2000.  It will be
Guest Edited by Prof. Henning Schulzrinne of Columbia University,
Prof. Hideo Miyahara of Osaka University, and Prof. Luigi Fratta of
the University of Milan.  The topics include but are not limited to:

Differentiated Services
Integrated Services
MPLS
Real-time multicast
IP network traffic engineering
Performance measurement and evaluation
Pricing and billing
User perception of QOS
QOS policy management
Adaptive and other new service models
Light-weight reservation protocols and aggregation

Continuing JCN's tradition of fast turnaround together with full peer
reviews, the following schedule has been set:

Dec. 31, 1999   Submit manuscript via web page (see below)

Mar. 31, 2000   First reviews returned to author, revisions returned
                within three weeks

June, 2000      Special Issue published

The guest editors are
Prof. Luigi Fratta  <fratta@elet.polimi.it>
Prof. Hideo Miyahara <miyahara@mercury.nal.ics.es.osaka-u.ac.jp>
Prof. Henning Schulzrinne <hgs@cs.columbia.edu>

Papers should be submitted via the procedure described at

http://www.cs.columbia.edu/~hgs/edas/JCN

Only PostScript and PDF formats are accepted.

Further information about JCN is available at http://JCN.snu.ac.kr. 
JCN is a high-quality quarterly archival journal, published by the
Korean Institute of Communications Sciences with the technical
cosponsorship of the IEEE Communications Society, covering the fields of
Communication Theory and Systems, Wireless Communications, and Networks
and Services.  JCN began publication in March, 1999.




From rem-conf Sun Dec 05 15:41:41 1999 
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Date: Sun, 05 Dec 1999 18:34:59 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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Subject: Re: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
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Gunnar Hellstr=F6m wrote:
>=20
> A question on tones.
> In the draft-ietf-avt-tones-03.txt, a set of modem and fax related tone=
s are
> specified.
>=20
> I miss two modem related signals used by calling modems in the same way=
 as
> the faxes use the CNG tone:
>=20
> The V.25 CT, Calling tone. 1300 Hz for 0.5 s with 2 s. intervals ( more
> precisely defined in V.25).
> All modems not starting with V.8 CI starts with this tone nowadays (at =
least
> in Europe).
>=20

I tentatively added this, but since it is not quite clear where the spec
is in the pipeline right now, it may be too late for this round. Adding
additional tones, however, can be done at any time, by normal IANA
registration procedures.

> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could be
> tranmitted as V.21 tones, but it might be of interest to have a specifi=
c
> code for it with a parameter for the data contents of the CI.

We currently don't really have tone parameters, so this would be
somewhat difficult to do.

>=20
> Since their inclusion would complete the modem handshake support in the
> tones spec, I suggest that they are added.
>=20
> Regards
> ______________________________________________
> Gunnar Hellstr=F6m
> Omnitor AB
> Alsn=F6gatan 7 ,  4 tr
> S-116 41 Stockholm
> Sweden
>=20


--=20
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Dec 06 03:14:05 1999 
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Subject: Re: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
In-Reply-To: Message from Henning Schulzrinne <schulzrinne@cs.columbia.edu> of "Sun, 05 Dec 1999 18:34:59 EST." <384AF6A3.66C0EE3E@cs.columbia.edu>
Date: Mon, 06 Dec 1999 11:04:19 +0000
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--> Henning Schulzrinne writes:
>Gunnar Hellström wrote:
>> 
>> A question on tones.
>> In the draft-ietf-avt-tones-03.txt, a set of modem and fax related tones are
>> specified.
>> 
>> I miss two modem related signals used by calling modems in the same way as
>> the faxes use the CNG tone:
>> 
>> The V.25 CT, Calling tone. 1300 Hz for 0.5 s with 2 s. intervals ( more
>> precisely defined in V.25).
>> All modems not starting with V.8 CI starts with this tone nowadays (at least
>> in Europe).
>
>I tentatively added this, but since it is not quite clear where the spec
>is in the pipeline right now, it may be too late for this round. Adding
>additional tones, however, can be done at any time, by normal IANA
>registration procedures.

We were taking the latest draft as being a revision to address working group
last call comments, and something we could send to the IESG. If this is just 
a new tone, which has no impact on the rest of the document, it is appropriate 
to add it now and make one final revision to the document. 

Any additional tones after this should, in my opinion, be registered with IANA
rather than added to this draft.

>> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could be
>> tranmitted as V.21 tones, but it might be of interest to have a specific
>> code for it with a parameter for the data contents of the CI.
>
>We currently don't really have tone parameters, so this would be
>somewhat difficult to do.

Does this imply a need for parameters? Or are the cases which would require
them sufficiently unusual that they can always be defined as sequences of
the existing tones? I'm hesitent about making this draft more complex,
unless there is a definite need.

Colin



From rem-conf Mon Dec 06 05:37:34 1999 
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From: =?iso-8859-1?Q?Gunnar_Hellstr=F6m?= <Gunnar.Hellstrom@omnitor.se>
To: "Colin Perkins" <c.perkins@cs.ucl.ac.uk>,
        "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: <rem-conf@es.net>
Subject: RE: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
Date: Mon, 6 Dec 1999 14:34:44 +0100
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OK,
1. For V.25 Calling tone:

Here is the definition of Calling Tone from V.25:  "The calling tone
consists of a series of interrupted bursts of binary 1 signal or 1300 Hz,=
 ON
for a duration of not less than 0.5 s and not more than 0.7 s and OFF for=
 a
duration of not less than 1.5 s and not more than 2.0 s."

In practice you can ignore the option of sending it as the "binary 1"
signal. All modems I know, transmit Calling Tone as 1300 Hz.
I believe the tones draft would gain by adding this definition, but I do =
not
want to hold up progress by requiring it.


2. V.8 CI.
This signal is meaningful only if communicated with its data contents.
It starts with a general CI identifier, but that is followed by contents
declaring possible modes and modulations. It is V.21 (1) modulated at 300
bits/s.
So, it may be true that the solution is to transmit it with the codes for
V.21 transmission.


Regards
Gunnar Hellstr=F6m


______________________________________________
Gunnar Hellstr=F6m
Omnitor AB
Alsn=F6gatan 7 ,  4 tr
S-116 41 Stockholm
Sweden

Tel +46 8 556 002 03
Mob: + 46 708 204 288
Txt and video +46 8 556 002 05
Fax +46 8 556 002 06

e-mail: gunnar.hellstrom@omnitor.se
web:    www.omnitor.se

> -----Original Message-----
> From: Colin Perkins [mailto:c.perkins@cs.ucl.ac.uk]
> Sent: Monday, December 06, 1999 12:04 PM
> To: Henning Schulzrinne
> Cc: m; rem-conf@es.net
> Subject: Re: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
>
>
> --> Henning Schulzrinne writes:
> >Gunnar Hellstr=F6m wrote:
> >>
> >> A question on tones.
> >> In the draft-ietf-avt-tones-03.txt, a set of modem and fax
> related tones are
> >> specified.
> >>
> >> I miss two modem related signals used by calling modems in the
> same way as
> >> the faxes use the CNG tone:
> >>
> >> The V.25 CT, Calling tone. 1300 Hz for 0.5 s with 2 s. intervals ( m=
ore
> >> precisely defined in V.25).
> >> All modems not starting with V.8 CI starts with this tone
> nowadays (at least
> >> in Europe).
> >
> >I tentatively added this, but since it is not quite clear where the sp=
ec
> >is in the pipeline right now, it may be too late for this round. Addin=
g
> >additional tones, however, can be done at any time, by normal IANA
> >registration procedures.
>
> We were taking the latest draft as being a revision to address
> working group
> last call comments, and something we could send to the IESG. If
> this is just
> a new tone, which has no impact on the rest of the document, it
> is appropriate
> to add it now and make one final revision to the document.
>
> Any additional tones after this should, in my opinion, be
> registered with IANA
> rather than added to this draft.
>
> >> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could =
be
> >> tranmitted as V.21 tones, but it might be of interest to have
> a specific
> >> code for it with a parameter for the data contents of the CI.
> >
> >We currently don't really have tone parameters, so this would be
> >somewhat difficult to do.
>
> Does this imply a need for parameters? Or are the cases which
> would require
> them sufficiently unusual that they can always be defined as sequences =
of
> the existing tones? I'm hesitent about making this draft more complex,
> unless there is a definite need.
>
> Colin
>
>




From rem-conf Mon Dec 06 08:20:05 1999 
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From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
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Colin Perkins wrote:
> 

> 
> We were taking the latest draft as being a revision to address working group
> last call comments, and something we could send to the IESG. If this is just
> a new tone, which has no impact on the rest of the document, it is appropriate
> to add it now and make one final revision to the document.

I'll await the outcome of this discussion and then submit -04 with this
change. This is indeed a trivial change.

> 
> Any additional tones after this should, in my opinion, be registered with IANA
> rather than added to this draft.

Agreed; or added when the draft progresses along the standards track,
assuming there's demonstrable need and usage.

> 
> >> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could be
> >> tranmitted as V.21 tones, but it might be of interest to have a specific
> >> code for it with a parameter for the data contents of the CI.
> >
> >We currently don't really have tone parameters, so this would be
> >somewhat difficult to do.
> 
> Does this imply a need for parameters? Or are the cases which would require
> them sufficiently unusual that they can always be defined as sequences of
> the existing tones? I'm hesitent about making this draft more complex,
> unless there is a definite need.

Adding a parameter would indeed require significant changes and
complexity. Also, this type of information is not really a tone anymore,
but basically a text or octet string of variable length.

I've had recent discussions about creating a third payload format (in
addition to the two in the current draft) suitable for caller-id. I
suspect that the V.8 CI fits there. My intent is to make this a separate
document, although many of the reliability and timing discussions from
the current draft will apply there, too. One of the items to be
discussed is whether caller-id is to be sent all at once or in the
"incremental" fashion we've used for DTMF. Caller-id consists of a
relatively long preamble, which consists of invalid characters, for
example, followed by roughly 30 to 40 information-bearing octets.

A summary of caller id can be found at
http://www-s.ti.com/sc/psheets/spra462/spra462.pdf

> 
> Colin

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Mon Dec 06 13:12:19 1999 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
cc: m <Gunnar.Hellstrom@omnitor.se>, rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
In-Reply-To: Message from Henning Schulzrinne <schulzrinne@cs.columbia.edu> of "Mon, 06 Dec 1999 11:08:04 EST." <384BDF64.645436B0@cs.columbia.edu>
Date: Mon, 06 Dec 1999 17:32:18 +0000
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--> Henning Schulzrinne writes:
>Colin Perkins wrote:
>> We were taking the latest draft as being a revision to address working group
>> last call comments, and something we could send to the IESG. If this is just
>> a new tone, which has no impact on the rest of the document, it is appropriate
>> to add it now and make one final revision to the document.
>
>I'll await the outcome of this discussion and then submit -04 with this
>change. This is indeed a trivial change.

Okay.

>> Any additional tones after this should, in my opinion, be registered with IANA
>> rather than added to this draft.
>
>Agreed; or added when the draft progresses along the standards track,
>assuming there's demonstrable need and usage.
>
>> 
>> >> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could be
>> >> tranmitted as V.21 tones, but it might be of interest to have a specific
>> >> code for it with a parameter for the data contents of the CI.
>> >
>> >We currently don't really have tone parameters, so this would be
>> >somewhat difficult to do.
>> 
>> Does this imply a need for parameters? Or are the cases which would require
>> them sufficiently unusual that they can always be defined as sequences of
>> the existing tones? I'm hesitent about making this draft more complex,
>> unless there is a definite need.
>
>Adding a parameter would indeed require significant changes and
>complexity. Also, this type of information is not really a tone anymore,
>but basically a text or octet string of variable length.
>
>I've had recent discussions about creating a third payload format (in
>addition to the two in the current draft) suitable for caller-id. I
>suspect that the V.8 CI fits there. My intent is to make this a separate
>document, although many of the reliability and timing discussions from
>the current draft will apply there, too. One of the items to be
>discussed is whether caller-id is to be sent all at once or in the
>"incremental" fashion we've used for DTMF. Caller-id consists of a
>relatively long preamble, which consists of invalid characters, for
>example, followed by roughly 30 to 40 information-bearing octets.

This seems to be a good approach, and one that won't hold up the current
tones payload format unnecessarily.

Colin



From rem-conf Mon Dec 06 15:28:43 1999 
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Colin Perkins" <c.perkins@cs.ucl.ac.uk>,
        "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: <rem-conf@es.net>
Subject: RE: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps
Date: Tue, 7 Dec 1999 00:26:36 +0100
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Caller ID is sent in PSTN before or between the Rings. Some networks use
DTMF, other use V.22, I think. Usually consists of the caller=B4s telepho=
ne
number.

V.8 CI is V.21(1) modulated. Can be as little as two bytes. Starts with a=
 CI
identifier. Still, your mechanism for V.21 transmission could carry it.

Regards
Gunnar

-----Original Message-----
From: Colin Perkins [mailto:c.perkins@cs.ucl.ac.uk]
Sent: Monday, December 06, 1999 6:32 PM
To: Henning Schulzrinne
Cc: m; rem-conf@es.net
Subject: Re: I-D ACTION:draft-ietf-avt-tones-03.txt,.ps


--> Henning Schulzrinne writes:
>Colin Perkins wrote:
>> We were taking the latest draft as being a revision to address working
group
>> last call comments, and something we could send to the IESG. If this i=
s
just
>> a new tone, which has no impact on the rest of the document, it is
appropriate
>> to add it now and make one final revision to the document.
>
>I'll await the outcome of this discussion and then submit -04 with this
>change. This is indeed a trivial change.

Okay.

>> Any additional tones after this should, in my opinion, be registered w=
ith
IANA
>> rather than added to this draft.
>
>Agreed; or added when the draft progresses along the standards track,
>assuming there's demonstrable need and usage.
>
>>
>> >> The V.8 CI. That is a V.21 (1) modulated signal. Therefore it could=
 be
>> >> tranmitted as V.21 tones, but it might be of interest to have a
specific
>> >> code for it with a parameter for the data contents of the CI.
>> >
>> >We currently don't really have tone parameters, so this would be
>> >somewhat difficult to do.
>>
>> Does this imply a need for parameters? Or are the cases which would
require
>> them sufficiently unusual that they can always be defined as sequences=
 of
>> the existing tones? I'm hesitent about making this draft more complex,
>> unless there is a definite need.
>
>Adding a parameter would indeed require significant changes and
>complexity. Also, this type of information is not really a tone anymore,
>but basically a text or octet string of variable length.
>
>I've had recent discussions about creating a third payload format (in
>addition to the two in the current draft) suitable for caller-id. I
>suspect that the V.8 CI fits there. My intent is to make this a separate
>document, although many of the reliability and timing discussions from
>the current draft will apply there, too. One of the items to be
>discussed is whether caller-id is to be sent all at once or in the
>"incremental" fashion we've used for DTMF. Caller-id consists of a
>relatively long preamble, which consists of invalid characters, for
>example, followed by roughly 30 to 40 information-bearing octets.

This seems to be a good approach, and one that won't hold up the current
tones payload format unnecessarily.

Colin





From rem-conf Mon Dec 06 19:55:58 1999 
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Date: Mon, 06 Dec 1999 19:47:30 -0800
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Bill Foster <bfoster@cisco.com>
Subject: Named signaling events - minor request.
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One minor comment/request for "draft-ietf-avt-tones-03.txt".

It would be nice to have a separate named signaling event (NSE) for ringing versus ring-back. 

The place where a NSE for ringback tone comes in useful is in a device control situation using MGCP or Megacop. The particular case is where the Media Gateway Controller (MGC) requests the media gateway (MG) to do ringback from the far end as follows:

Call between two residential media gateways MG A and MG B
- MG A originates a call to MG B
- once the MGC sets up a half duplex connection from MG B it requests ringing and ringback to MGB (in MGCP S: rg, rt@connection)
- if during the capabilities exchange earlier when the half duplex connection was set up (SDP passed between endpoints), the endpoints realized that they supported ringback tone using NSEs, they might use the NSE rather than a tone (some advantages in terms of use of bandwidth prior to call, as well as implementation advantages - MIPS required etc.).

On the other hand, the same MG may be used in other RTP trunking applications where an MGC may set up the trunk and NSE's used for signaling (e.g. ringing request passed across the RTP stream).

If we leave ringing and ringback as the same event that depends on context (like hook status) there are problems in the trunking application. If one end sends a ringing request to the other end - but when it gets there the person just happens to have gone off-hook (because they wanted to originate a call just as a call was coming in) - if the MG is left to interpret the meaning of the ringing NSE event based on hook-status - they will suddenly hear ringback - not quite what was intended.

In Summary:

Any chance of getting to separate NSE events - one for ringing and another for ringback - they are both useful but only if the are separate events.

Bill 
-----------------------------
Bill Foster
Phone: (408) 527-8791
Page:   (800) 365-4578
Fax:     (781) 998-6492



From rem-conf Tue Dec 07 09:31:39 1999 
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Tmima Koren wrote:

> At 07:39 PM 12/2/99 -0500, GeeVarghese John wrote:
> >Hi,
> >
> >Tmima Koren wrote:
> >
> >> When the IP ID is changing arbitrarily, we have to change the deltaID very
> >frequently - probably in every packet.
> >> In such cases it might be more useful to include the full IP ID in a CR
> >instead of the deltaID.
> >
> >Exactly, you are correct ..
> >This is the thing I was mentioning by saying three stages of compression.
> >At the first stage we can start compressing a flow at CRTP then if we found
> >the flow is
> >inconsistent then move the state to CUDP and send only CUDP packet then
> >CUDPid packet.
> >Inconsistency can be interms of anything ..
> >One thing could be like what you said about IP id variance rate or other
> >delta encoding rate or CONTEX_STATE packet received etc ..
>
> Yes. This is the same idea.
> The CRid can be used when the RTP parameters are consistent, but the IP ID is not.
>
> Mikael Degermark suggested to enhance the COMPRESSED_NON_TCP packet instead of the CUDP
> The COMPRESSED_NON_TCP packet includes the IP ID, the udp checksum (if nonzero) and also includes one byte of data which currently holds the 4-bit CRTP sequence# (see rfc2507). 2 of the 4 unused bits can be used to flag 'T' and 'I', so the packet may also include the deltaT and deltaID if necessary.
> Such a packet will restore all parameters at the decompressor, same as CUDPid

What I read about COMPRESSED_NON_TCP packet is, it is optional for CRTP and
mainly meant for fragmented packet etc ..Are we nt restricting the advantages here ..
(CUDP with IP ID can compress more (UDP etc ) fields than NON_TCP packet)
But having a mechanis to send IP ID will definitely give some sort of improvement in CUDP.

>
>
> >
> >>
> >> This can be done if we could change the meaning of the 'I' flag in the CR
> >packet to mean either full IP ID or deltaID.
> >> Maybe we can add an option to the CRTP negotiation about the usage of the
> >'I' flag in CR: full IP ID or deltaID
> >> Then if a host requires preserving the IP ID and cannot assign them
> >sequentially for the compressed RTP streams, the host will negotiate the 'I'
> >flag in CR packets to mean full IP ID.
> >> Or if we want this to be per stream, add a flag in the FH.
> >
> >It looks Very good. I am in favour of  negotation  at CRTP level not only
> >for this point
> >I have something more on seq number extension also. May be later we will
> >discuss this.
> >But Adding the information bit in the FH packet how do we ensure the reliable
> >delivery of the FH packet ..If Accept and Reject packet is the soln in your
> >mind
> >That could be one possibility but Accept and Reject packet will have many
> >other dsadvantes.
> >Instead of that, why cant we have a protocol negotiation packet which needs
> >to be sent
> >only one time when the stack is initialised. For this packet we can have a
> >design where
> >negotation parameters can be exteneded so that future requirement also will
> >be taken care.
>
> I meant adding options to the IP header compression negotiation as described in rfc2509
>
> >>
> >>
> >> Are there cases when it's ok for the CRTP compressor to reasign the IP ID?
> >
> >Do you mean that CRTP will reassign a new IP id to the decompressed IP
> >packet  ..?
>
> Yes. I suggest a mode where CRTP does not preserve the IP ID.
> Tmima
>
> >
> >
> >Rgds
> >GeevJohn
> >--
> >-------------------------------------------------------------------------------
> >[ ]  Motorola Confidential Proprietory
> >[ ]  Motorola Internal Use Only
> >[x]  Not an MCP or MIU
> >-------------------------------------------------------------------------------
> >Geevarghese John,
> >DataCom Group,
> >Motorola India Electronics Ltd,
> >"The Senate", No-33 A,
> >Ulsoor Road,
> >Bangalore - 42.
> >Phone  : 91-80-5598615  Xnt-4005
> >E-mail : gvjohn@miel.mot.com
> >-------------------------------------------------------------------------------
> >-------------------------------------------------------------------------------
> >
> >
> >
> >
> >

--





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Tmima,

I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP
extension draft..
It is a continuation of the old mail. In you answer below, though you
wrote the
ACCPET/REJECT packe is optional, I think definition of these packet are
bit confusing.
First of all I feel no need of a ACCEPT packet
Second Why to have a REJECT packet as separate packet to invalidate the
context ..
Why cant we use CONTEXT_STATE packet for the same ..
We can have a bit in CS packet  indicating that context need to be
invalidated or
resynchronised using FH packet. If there is no CS packet it is assumed
that
flow ACCEPT.

Here the issue of interoperability also arise when an implementation
interopeate with the
old CRTP (2508) implemnetation new implemetation expected ACCEPT or
REJECT packet
for each FH packet .. If it does not receive either packet what action
compressor is expected to take. To make interoperation possible there is
a need for some sort of negotiation, here also a protocol level
negotiation is expected ..
So I feel the approach with ACCEPT/REJECT packet is not optimal.
But the solution I suggested  does not cause any interoptation issue
also.
No extra packet overhead, no protocol modification etc  ...

Could please answer your points on this ..

Thanks & Rgds

GVJ

>
>
>> Q3) Section III : Acknowledgement to a new compressed stream
>>
>> Context Invalidation was one point which I had pointed out to
>> Steve/tmima long time back ..
>> I am having those mails with me ..
>>
>> But, (i) Do we need ACCEPT and REJECT packet  ??
>> Having a NACK mechnism sounds reasonable ..
>> I feel there shouldnt be both ..
>> But what compressor is supposed to do after sending the first
>> FULL_HEADER packet
>> when compressor waiting for the ACCEPT or REJECT
>> should it queue the packets till the reply/or timeout
>> or Should it send the compressed packets assuming the decompressor
>> will drop if there are
>> REJECT state at the decompressor side.
>>
>> But (ii) what about the backword compatibility issue because this
>> ACCEPT/REJECT mechanism has dependancy and it will not work with
>> CRTP-2508
>>
>> As per the current proposal we need to send ACCEPT/REJECT packet for
>> every
>> FULL_HEADER packet ... (iii)Is it not adding extra bandwidth
>> consuption ..
>> Draft talks about FULL_HEADER retransmit as well ..
>
>
> The REJECT packet is useful when the decompressor runs out of
> resources. The compressor initiates compression of a stream by sending
> a FH. With CRTP today the decompressor can send a CS to invalidate the
> context, but the compressor does not know the reason and will attempt
> to compress this stream again. A REJECT will notify the compressor
> that the decompressor cannot handle compresstion of additional streams
> at this moment.
> An implementation that was not enhanced to use REJECT will still use
> CS to invalidate the context each time the compressor attempts to
> compress the stream.
> The compressor continues to send the compressed packets of the stream
> until he receives CS or REJECT
> The intent is to make the ACCEPT/REJECT packet optional - an
> optimization.
>
> Tmima
>

--------------15D6398F85975468E05872C5
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Tmima,
<p>I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP extension
draft..
<br>It is a continuation of the old mail. In you answer below, though you
wrote the
<br>ACCPET/REJECT packe is optional, I think definition of these packet
are bit confusing.
<br>First of all I feel no need of a ACCEPT packet
<br>Second Why to have a REJECT packet as separate packet to invalidate
the context ..
<br>Why cant we use CONTEXT_STATE packet for the same ..
<br>We can have a bit in CS packet&nbsp; indicating that context need to
be invalidated or
<br>resynchronised using FH packet. If there is no CS packet it is assumed
that
<br>flow ACCEPT.
<p>Here the issue of interoperability also arise when an implementation
interopeate with the
<br>old CRTP (2508) implemnetation new implemetation expected ACCEPT or
REJECT packet
<br>for each FH packet .. If it does not receive either packet what action
compressor is expected to take. To make interoperation possible there is
a need for some sort of negotiation, here also a protocol level negotiation
is expected ..
<br>So I feel the approach with ACCEPT/REJECT packet is not optimal.
<br>But the solution I suggested&nbsp; does not cause any interoptation
issue also.
<br>No extra packet overhead, no protocol modification etc&nbsp; ...
<p>Could please answer your points on this ..
<p>Thanks &amp; Rgds
<p>GVJ
<blockquote TYPE=CITE>&nbsp;
<blockquote type=cite cite><b>Q3) Section III : Acknowledgement to a new
compressed stream</b>
<p>Context Invalidation was one point which I had pointed out to Steve/tmima
long time back ..
<br>I am having those mails with me ..
<p>But, (i) <i>Do we need ACCEPT and REJECT packet&nbsp; ??</i>
<br>Having a NACK mechnism sounds reasonable ..
<br>I feel there shouldnt be both ..
<br>But what compressor is supposed to do after sending the first FULL_HEADER
packet
<br>when compressor waiting for the ACCEPT or REJECT
<br>should it queue the packets till the reply/or timeout
<br>or Should it send the compressed packets assuming the decompressor
will drop if there are
<br>REJECT state at the decompressor side.
<p>But (ii) <i>what about the backword compatibility issue </i>because
this ACCEPT/REJECT mechanism has dependancy and it will not work with CRTP-2508
<p>As per the current proposal we need to send ACCEPT/REJECT packet for
every
<br>FULL_HEADER packet ... (iii)<i>Is it not adding extra bandwidth consuption
</i>..
<br>Draft talks about FULL_HEADER retransmit as well ..</blockquote>

<p><br>The REJECT packet is useful when the decompressor runs out of resources.
The compressor initiates compression of a stream by sending a FH. With
CRTP today the decompressor can send a CS to invalidate the context, but
the compressor does not know the reason and will attempt to compress this
stream again. A REJECT will notify the compressor that the decompressor
cannot handle compresstion of additional streams at this moment.
<br>An implementation that was not enhanced to use REJECT will still use
CS to invalidate the context each time the compressor attempts to compress
the stream.
<br>The compressor continues to send the compressed packets of the stream
until he receives CS or REJECT
<br>The intent is to make the ACCEPT/REJECT packet optional - an optimization.
<p>Tmima
<br>&nbsp;</blockquote>
</html>

--------------15D6398F85975468E05872C5--




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                         =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
                         PACKET VIDEO 2000
                         =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

           The 10th International Packet Video Workshop
                           =20
                            1-2 May 2000
            Forte Village Resort, Cagliari, Sardinia, Italy
                   http://www.diee.unica.it/pv2000/

     ********** Submission Date Extended to December 31 **********
 ------------------------------------------------------------------------
 PV 2000 will be held in the middle of the Mediterranean Sea, at the Forte
 Village Resort near Cagliari (Sardinia island), Italy. The workshop is
 devoted to present technological advancements and innovations in video
 communications over packet networks, in particular, the Internet.
 PVs have been unique in providing a common ground for people from video
 coding and networking fields; presentations on theory and practice,
 standards activities, and business and consumer applications are=
 encouraged.
 We cordially invite you to take part in this workshop by submitting
 your work and look forward to welcoming you in Sardinia in May 2000
 for what will be a rewarding and exciting experience!
 ------------------------------------------------------------------------
 SUBMISSIONS

 Please submit an electronic manuscript written in Word or HTML,
 not exceeding 10 printed pages. We will produce a CD-ROM (pdf
 format) containing all the accepted papers.
 Please note that the author of the best paper will receive a $250=20
 prize and a diploma suitable for framing.=20
 Additionally, SELECTED PAPERS ARE LIKELY TO BE PUBLISHED, AFTER REVISION,
 IN THE FOLLOWING JOURNALS:
 - European Transactions on Telecommunications
 - Image Communication

 DEADLINES
 Paper submission: DECEMBER 31, 1999,=20
 Notification of acceptance: February 29, 2000
 Final paper delivery: March 31, 2000=20

 SUBMISSIONS PROCEDURES
 Submit your work in a Word document OR a set of HTML=20
 files organized in a single directory to: pv2000@diee.unica.it
 Detailed instructions for authors and help with manuscript
 preparation can be found on the conference web page.
 ----------------------------------------------------------------------
 GENERAL CHAIRS:
 Francesco G.B. De Natale, University of Trento, Italy
 Daniel D. Giusto, University of Cagliari, Italy

 TECHNICAL COMMITTEE CHAIRS:
 Leonardo Chiariglione, CSELT, Italy
 Maurizio D=E8cina, Cefriel and Politecnico di Milano, Italy

 TECHNICAL COMMITTEE (members):
 John Arnold, Univ. New South Wales, Australia
 Andrea Basso, AT&T, USA
 Stephen Casner, CISCO, USA
 Shih-Fu Chang, Columbia University, USA
 M. Reha Civanlar, AT&T, USA
 Jon Crowcroft, Univ. College London, UK
 Edward Delp, Purdue Univ., USA
 Touradj Ebrahimi, EPFL, Switzerland
 Mohamed Ghanbari, Univ. Essex, UK
 Barry G. Haskell, AT&T, USA
 Yu Hen Hu, Univ. Wisconsin, USA
 Aggelos Katsaggelos, Northwestern Univ., USA
 Jae-Kyoon Kim, Kaist, Korea
 Faouzi Kossentini, Univ. British Columbia, Canada
 C.-C. Jay Kuo, Univ. Southern California, USA
 Steven McCanne, U.C. Berkeley, USA
 James Modestino, Rensselaer Polytechnic Institute, USA
 Geoff Morrison, BT, UK
 Hans-Georg Musmann, TU Hannover, Germany
 Naohisa Ohta, Sony, Japan
 Sakae Okubo, Telecommunications Adv. Org., Japan
 Joerg Ott, Univ. Bremen, Germany
 Fernando Pereira, Instituto Superiore T=E9cnico, Portugal
 Majid Rabbani, Eastman Kodak, USA
 Amy Reibman, AT&T, USA
 Philippe Salembier, UPC, Spain
 Henning Schulzrinne, Columbia Univ., USA
 Gary Sullivan, Microsoft, USA
 A. Murat Tekalp, Univ. Rochester, USA
 Thierry Turletti, INRIA, France
 Stephan Wenger, TU Berlin, Germany
 John Woods, Rensselaer Polytechnic Institute, USA
 Hiroshi Yasuda, Tokyo Univ., Japan

 PUBLICATIONS AND DEMOS:
 Luigi Atzori, University of Cagliari, Italy=20
 ------------------------------------------------------------------------=20
 TECHNICAL PROGRAM

 The technical program of Packet Video 2000 will consist of invited
 talks, submitted paper presentations, poster sessions and demos.=20
 Topics of interest include, but are not limited to:
 Video processing and transmission
 Video streaming over the Internet
 Network adaptive video coding and transport
 Packetized video for home LANs
 Packetized video for wireless/mobile systems
 Packet video protocols and storage formats
 Layered coding for error resilience and heterogeneous networks
 Packet loss resilient coding and transport=20
 Terminal and server architectures for Internet TV
 Efficient transcoding for heterogeneous networks=20
 Congestion control
 Error concealment
 Pre and post-processing for picture quality enhancement
 Statistical multiplexing for greater network and terminal utilization
 Traffic shaping for efficient network and terminal utilization=20
 Interstream synchronization for multiple video presentations=20
 Packet network performance modeling and evaluation
 Rate control for VBR video
 Standards: MPEG4, MPEG7, H.26L, H.323, RTP, RTSP, SIP, SDP
 Multicasting, MBONE applications
 Implementations and commercial applications
 ------------------------------------------------------------------------
 LOCAL ARRANGEMENTS

 PV2000 Organizing Committee
 Dept. of Electrical and Electronic Engineering
 University of Cagliari
 Piazza d'Armi
 09123 Cagliari, Italy
 pv2000@diee.unica.it





From rem-conf Tue Dec 07 11:11:09 1999 
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To: GeeVarghese John <gvjohn@miel.mot.com>, Tmima Koren <tmima@cisco.com>
From: Mikael Degermark <micke@cdt.luth.se>
Subject: Re: CRTP and IP ID
Cc: casner@cisco.com, micke@cdt.luth.se, brucet@cisco.com, dwing@cisco.com,
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At 10:42 PM 99/12/07 -0500, GeeVarghese John wrote:
>> Mikael Degermark suggested to enhance the COMPRESSED_NON_TCP packet
instead of the CUDP
>> The COMPRESSED_NON_TCP packet includes the IP ID, the udp checksum (if
nonzero) and also includes one byte of data which currently holds the 4-bit
CRTP sequence# (see rfc2507). 2 of the 4 unused bits can be used to flag
'T' and 'I', so the packet may also include the deltaT and deltaID if
necessary.
>> Such a packet will restore all parameters at the decompressor, same as
CUDPid
>
>What I read about COMPRESSED_NON_TCP packet is, it is optional for CRTP and
>mainly meant for fragmented packet etc.

CNTCP can be used for any header. It does not use delta-encoding. For a
UDP/IP header,
the difference from CUDP is that the IP identifier is not sent using delta
encoding. Instead,
it is sent as an absolute value. It is not true that CUDP compresses more
fields than 
CNTCP (except for IP ID of course). What fields did you have in mind?

> ..Are we nt restricting the advantages here ..
>(CUDP with IP ID can compress more (UDP etc ) fields than NON_TCP packet)
>But having a mechanis to send IP ID will definitely give some sort of
improvement in CUDP.

The real question here is whether one wants to have more than one way to do
things.
If one decides to go with CUDP with the option of sending the IP ID as-is,
there will 
be two ways. Moreover, when we want to support IPv6, where there is no IP
ID field,
CUDP becomes more or less obsolete.

>> >> This can be done if we could change the meaning of the 'I' flag in
the CR
>> >packet to mean either full IP ID or deltaID.

The case where the IP ID increments as expected would then not be covered.
That would be bad for the well-behaved stack which increments IP ID by one
for each packet. 

>> >> Maybe we can add an option to the CRTP negotiation about the usage of
the
>> >'I' flag in CR: full IP ID or deltaID
>> >> Then if a host requires preserving the IP ID and cannot assign them
>> >sequentially for the compressed RTP streams, the host will negotiate
the 'I'
>> >flag in CR packets to mean full IP ID.

Won't work when the host is on the receiving end of the RTP stream.

>> >> Or if we want this to be per stream, add a flag in the FH.

Negotiating on that level will add considerably to the complexity of the
protocol
in the presence of dropped packets, etc. 
Do we really want that?

>> I meant adding options to the IP header compression negotiation as
described in rfc2509

Please make sure that what you propose makes sense for the case where none
of the endpoints of the narrow link is the sender. It does not seem to me
that the 
information needed to negotiate for example the meaning of the 'I' bit is
necessarily
available to either side of the link. 

>> >> Are there cases when it's ok for the CRTP compressor to reasign the
IP ID?
>> >
>> >Do you mean that CRTP will reassign a new IP id to the decompressed IP
>> >packet  ..?
>>
>> Yes. I suggest a mode where CRTP does not preserve the IP ID.

Please make sure that the IESG has changed their mind before you pursue that
any further. Might save you some work & trouble. 

Micke Degermark





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From: Tmima Koren <tmima@cisco.com>
Subject: Re: Extensions to CRTP
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At 10:45 PM 12/7/99 -0500, GeeVarghese John wrote: 
>
> Tmima, 
>
> I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP extension
> draft.. 
> It is a continuation of the old mail. In you answer below, though you wrote
> the 
> ACCPET/REJECT packe is optional, I think definition of these packet are bit
> confusing. 
> First of all I feel no need of a ACCEPT packet 
> Second Why to have a REJECT packet as separate packet to invalidate the
> context .. 
> Why cant we use CONTEXT_STATE packet for the same .. 
> We can have a bit in CS packet  indicating that context need to be
> invalidated or 
> resynchronised using FH packet. If there is no CS packet it is assumed that 
> flow ACCEPT. 


Yes, we suggested two formats for the REJECT packet: one is using a new code,
and the other is using the CS with an extra bit
See attached presentation (presented in DC)
We also prefer to use the 'CS + extra bit' format 
Tmima


>
> Here the issue of interoperability also arise when an implementation
> interopeate with the 
> old CRTP (2508) implemnetation new implemetation expected ACCEPT or REJECT
> packet 
> for each FH packet .. If it does not receive either packet what action
> compressor is expected to take. To make interoperation possible there is a
> need for some sort of negotiation, here also a protocol level negotiation is
> expected .. 
> So I feel the approach with ACCEPT/REJECT packet is not optimal. 
> But the solution I suggested  does not cause any interoptation issue also. 
> No extra packet overhead, no protocol modification etc  ... 
>
> Could please answer your points on this .. 


All the enhancements are not backward compatible so we'll have to change the
negotiation: maybe negotiate a version of CRTP, or negotiate the new options
separately
Tmima


>
> Thanks & Rgds 
>
> GVJ 
>>
>>   
>>>
>>> Q3) Section III : Acknowledgement to a new compressed stream 
>>>
>>> Context Invalidation was one point which I had pointed out to Steve/tmima
>>> long time back .. 
>>> I am having those mails with me .. 
>>>
>>> But, (i) Do we need ACCEPT and REJECT packet  ?? 
>>> Having a NACK mechnism sounds reasonable .. 
>>> I feel there shouldnt be both .. 
>>> But what compressor is supposed to do after sending the first FULL_HEADER
>>> packet 
>>> when compressor waiting for the ACCEPT or REJECT 
>>> should it queue the packets till the reply/or timeout 
>>> or Should it send the compressed packets assuming the decompressor will
>>> drop if there are 
>>> REJECT state at the decompressor side. 
>>>
>>> But (ii) what about the backword compatibility issue because this
>>> ACCEPT/REJECT mechanism has dependancy and it will not work with CRTP-2508 
>>>
>>> As per the current proposal we need to send ACCEPT/REJECT packet for every 
>>> FULL_HEADER packet ... (iii)Is it not adding extra bandwidth consuption .. 
>>> Draft talks about FULL_HEADER retransmit as well ..
>>
>>
>>
>> The REJECT packet is useful when the decompressor runs out of resources. The
>> compressor initiates compression of a stream by sending a FH. With CRTP
>> today the decompressor can send a CS to invalidate the context, but the
>> compressor does not know the reason and will attempt to compress this stream
>> again. A REJECT will notify the compressor that the decompressor cannot
>> handle compresstion of additional streams at this moment. 
>> An implementation that was not enhanced to use REJECT will still use CS to
>> invalidate the context each time the compressor attempts to compress the
>> stream. 
>> The compressor continues to send the compressed packets of the stream until
>> he receives CS or REJECT 
>> The intent is to make the ACCEPT/REJECT packet optional - an optimization. 
>>
>> Tmima 
>>  
>



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<html>
At 10:45 PM 12/7/99 -0500, GeeVarghese John wrote: <br>
<blockquote type=cite cite>Tmima, <br>
<br>
I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP extension
draft.. <br>
It is a continuation of the old mail. In you answer below, though you
wrote the <br>
ACCPET/REJECT packe is optional, I think definition of these packet are
bit confusing. <br>
First of all I feel no need of a ACCEPT packet <br>
Second Why to have a REJECT packet as separate packet to invalidate the
context .. <br>
Why cant we use CONTEXT_STATE packet for the same .. <br>
We can have a bit in CS packet&nbsp; indicating that context need to be
invalidated or <br>
resynchronised using FH packet. If there is no CS packet it is assumed
that <br>
flow ACCEPT. </blockquote><br>
Yes, we suggested two formats for the REJECT packet: one is using a new
code, and the other is using the CS with an extra bit<br>
See attached presentation (presented in DC)<br>
We also prefer to use the 'CS + extra bit' format <br>
Tmima<br>
<br>
<br>
<blockquote type=cite cite>Here the issue of interoperability also arise
when an implementation interopeate with the <br>
old CRTP (2508) implemnetation new implemetation expected ACCEPT or
REJECT packet <br>
for each FH packet .. If it does not receive either packet what action
compressor is expected to take. To make interoperation possible there is
a need for some sort of negotiation, here also a protocol level
negotiation is expected .. <br>
So I feel the approach with ACCEPT/REJECT packet is not optimal. <br>
But the solution I suggested&nbsp; does not cause any interoptation issue
also. <br>
No extra packet overhead, no protocol modification etc&nbsp; ... <br>
<br>
Could please answer your points on this .. </blockquote><br>
All the enhancements are not backward compatible so we'll have to change
the negotiation: maybe negotiate a version of CRTP, or negotiate the new
options separately<br>
Tmima<br>
<br>
<br>
<blockquote type=cite cite>Thanks &amp; Rgds <br>
<br>
GVJ <br>
<blockquote type=cite cite>&nbsp; <br>
<b><blockquote type=cite cite>Q3) Section III : Acknowledgement to a new
compressed stream</b> <br>
<br>
Context Invalidation was one point which I had pointed out to Steve/tmima
long time back .. <br>
I am having those mails with me .. <br>
<br>
But, (i) <i>Do we need ACCEPT and REJECT packet&nbsp; ??</i> <br>
Having a NACK mechnism sounds reasonable .. <br>
I feel there shouldnt be both .. <br>
But what compressor is supposed to do after sending the first FULL_HEADER
packet <br>
when compressor waiting for the ACCEPT or REJECT <br>
should it queue the packets till the reply/or timeout <br>
or Should it send the compressed packets assuming the decompressor will
drop if there are <br>
REJECT state at the decompressor side. <br>
<br>
But (ii) <i>what about the backword compatibility issue </i>because this
ACCEPT/REJECT mechanism has dependancy and it will not work with
CRTP-2508 <br>
<br>
As per the current proposal we need to send ACCEPT/REJECT packet for
every <br>
FULL_HEADER packet ... (iii)<i>Is it not adding extra bandwidth
consuption </i>.. <br>
Draft talks about FULL_HEADER retransmit as well ..</blockquote><br>
<br>
The REJECT packet is useful when the decompressor runs out of resources.
The compressor initiates compression of a stream by sending a FH. With
CRTP today the decompressor can send a CS to invalidate the context, but
the compressor does not know the reason and will attempt to compress this
stream again. A REJECT will notify the compressor that the decompressor
cannot handle compresstion of additional streams at this moment. <br>
An implementation that was not enhanced to use REJECT will still use CS
to invalidate the context each time the compressor attempts to compress
the stream. <br>
The compressor continues to send the compressed packets of the stream
until he receives CS or REJECT <br>
The intent is to make the ACCEPT/REJECT packet optional - an
optimization. <br>
<br>
Tmima <br>
&nbsp;</blockquote></blockquote><br>
</html>

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From rem-conf Tue Dec 07 19:10:27 1999 
From rem-conf-request@es.net Tue Dec 07 19:10:26 1999
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Date: Tue, 07 Dec 1999 22:05:35 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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CC: rem-conf@es.net
Subject: Re: Named signaling events - minor request.
References: <4.2.0.58.19991206192731.01c805e0@redale.cisco.com>
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Bill Foster wrote:
> 
> One minor comment/request for "draft-ietf-avt-tones-03.txt".
> 
> It would be nice to have a separate named signaling event (NSE) for ringing versus ring-back.

This seems reasonable. The only minor complications are naming (since
there's the potential for confusion between ringing tone = ringback and
"ring") and that this tone doesn't have a canonical representation. To
minimize the confusion, I used the term "alert". My suggested wording
is:

--- begin spec excerpt ---

\item[Alert:] This named signal event causes the recipient to generate
an alert signal (``ringing''). The
tone rendition is up to the receiver. (This differs from the ringing
tone, below, heard by the {\bf caller}.)

--- end excerpt ---

Please let me know if this definition is clear enough.

A meta comment: since the last call periods are over, I will now declare
a last and final chance. Any tone request, however reasonable, that
doesn't reach me by tomorrow 5 pm EST will not be included in this round
so that I can package up -04 and get this to the IESG, finally.
Additional requests will have to wait

> 
> The place where a NSE for ringback tone comes in useful is in a device control situation using MGCP or Megacop. The particular case is where the Media Gateway Controller (MGC) requests the media gateway (MG) to do ringback from the far end as follows:
> 
> Call between two residential media gateways MG A and MG B
> - MG A originates a call to MG B
> - once the MGC sets up a half duplex connection from MG B it requests ringing and ringback to MGB (in MGCP S: rg, rt@connection)
> - if during the capabilities exchange earlier when the half duplex connection was set up (SDP passed between endpoints), the endpoints realized that they supported ringback tone using NSEs, they might use the NSE rather than a tone (some advantages in terms of use of bandwidth prior to call, as well as implementation advantages - MIPS required etc.).
> 
> On the other hand, the same MG may be used in other RTP trunking applications where an MGC may set up the trunk and NSE's used for signaling (e.g. ringing request passed across the RTP stream).
> 
> If we leave ringing and ringback as the same event that depends on context (like hook status) there are problems in the trunking application. If one end sends a ringing request to the other end - but when it gets there the person just happens to have gone off-hook (because they wanted to originate a call just as a call was coming in) - if the MG is left to interpret the meaning of the ringing NSE event based on hook-status - they will suddenly hear ringback - not quite what was intended.
> 
> In Summary:
> 
> Any chance of getting to separate NSE events - one for ringing and another for ringback - they are both useful but only if the are separate events.
> 
> Bill
> -----------------------------
> Bill Foster
> Phone: (408) 527-8791
> Page:   (800) 365-4578
> Fax:     (781) 998-6492

-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



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Tmima,

It is a good news for me that you added the bit solution in CS packet
for contex invalidation.
I remeber on 15 April 1999 when I pointed out this problem and this soln
to Steven Casener
the response I got from him was not in favour of this idea. He just
replied ::
>>> you can't just have an error flag.
May be I didnot communicate properly.
Anyway this solution is in place though the ownership is for different
people, No problem.

Could you please send me the latest draft ...

Thanks & Rgds
GVJ

Tmima Koren wrote:

>  At 10:45 PM 12/7/99 -0500, GeeVarghese John wrote:
>
>> Tmima,
>>
>> I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP
>> extension draft..
>> It is a continuation of the old mail. In you answer below, though
>> you wrote the
>> ACCPET/REJECT packe is optional, I think definition of these packet
>> are bit confusing.
>> First of all I feel no need of a ACCEPT packet
>> Second Why to have a REJECT packet as separate packet to invalidate
>> the context ..
>> Why cant we use CONTEXT_STATE packet for the same ..
>> We can have a bit in CS packet  indicating that context need to be
>> invalidated or
>> resynchronised using FH packet. If there is no CS packet it is
>> assumed that
>> flow ACCEPT.
>
>
> Yes, we suggested two formats for the REJECT packet: one is using a
> new code, and the other is using the CS with an extra bit
> See attached presentation (presented in DC)
> We also prefer to use the 'CS + extra bit' format
> Tmima
>
>
>
>> Here the issue of interoperability also arise when an implementation
>> interopeate with the
>> old CRTP (2508) implemnetation new implemetation expected ACCEPT or
>> REJECT packet
>> for each FH packet .. If it does not receive either packet what
>> action compressor is expected to take. To make interoperation
>> possible there is a need for some sort of negotiation, here also a
>> protocol level negotiation is expected ..
>> So I feel the approach with ACCEPT/REJECT packet is not optimal.
>> But the solution I suggested  does not cause any interoptation issue
>> also.
>> No extra packet overhead, no protocol modification etc  ...
>>
>> Could please answer your points on this ..
>
>
> All the enhancements are not backward compatible so we'll have to
> change the negotiation: maybe negotiate a version of CRTP, or
> negotiate the new options separately
> Tmima
>
>
>
>> Thanks & Rgds
>>
>> GVJ
>>
>> >
>> >
>> >>  Q3) Section III : Acknowledgement to a new compressed stream
>> >>
>> >>  Context Invalidation was one point which I had pointed out to
>> >>  Steve/tmima long time back ..
>> >>  I am having those mails with me ..
>> >>
>> >>  But, (i) Do we need ACCEPT and REJECT packet  ??
>> >>  Having a NACK mechnism sounds reasonable ..
>> >>  I feel there shouldnt be both ..
>> >>  But what compressor is supposed to do after sending the first
>> >>  FULL_HEADER packet
>> >>  when compressor waiting for the ACCEPT or REJECT
>> >>  should it queue the packets till the reply/or timeout
>> >>  or Should it send the compressed packets assuming the
>> >>  decompressor will drop if there are
>> >>  REJECT state at the decompressor side.
>> >>
>> >>  But (ii) what about the backword compatibility issue because this
>> >>  ACCEPT/REJECT mechanism has dependancy and it will not work with
>> >>  CRTP-2508
>> >>
>> >>  As per the current proposal we need to send ACCEPT/REJECT packet
>> >>  for every
>> >>  FULL_HEADER packet ... (iii)Is it not adding extra bandwidth
>> >>  consuption ..
>> >>  Draft talks about FULL_HEADER retransmit as well ..
>> >
>> > The REJECT packet is useful when the decompressor runs out of
>> > resources. The compressor initiates compression of a stream by
>> > sending a FH. With CRTP today the decompressor can send a CS to
>> > invalidate the context, but the compressor does not know the reason
>> > and will attempt to compress this stream again. A REJECT will
>> > notify the compressor that the decompressor cannot handle
>> > compresstion of additional streams at this moment.
>> > An implementation that was not enhanced to use REJECT will still
>> > use CS to invalidate the context each time the compressor attempts
>> > to compress the stream.
>> > The compressor continues to send the compressed packets of the
>> > stream until he receives CS or REJECT
>> > The intent is to make the ACCEPT/REJECT packet optional - an
>> > optimization.
>> >
>> > Tmima
>>
--
-------------------------------------------------------------------------------

[ ]  Motorola Confidential Proprietory
[ ]  Motorola Internal Use Only
[x]  Not an MCP or MIU
-------------------------------------------------------------------------------

Geevarghese John,
DataCom Group,
Motorola India Electronics Ltd,
"The Senate", No-33 A,
Ulsoor Road,
Bangalore - 42.
Phone  : 91-80-5598615  Xnt-4005
E-mail : gvjohn@miel.mot.com
-------------------------------------------------------------------------------

-------------------------------------------------------------------------------



--------------2469912007B0B7FB30D56393
Content-Type: text/html; charset=us-ascii
Content-Transfer-Encoding: 7bit

<!doctype html public "-//w3c//dtd html 4.0 transitional//en">
<html>
Tmima,
<p>It is a good news for me that you added the bit solution in CS packet
for contex invalidation.
<br>I remeber on 15 April 1999 when I pointed out this problem and this
soln to Steven Casener
<br>the response I got from him was not in favour of this idea. He just
replied ::
<br>>>> you can't just have an error flag.
<br>May be I didnot communicate properly.
<br>Anyway this solution is in place though the ownership is for different
people, No problem.
<p>Could you please send me the latest draft ...
<p>Thanks &amp; Rgds
<br>GVJ
<p>Tmima Koren wrote:
<blockquote TYPE=CITE>&nbsp;At 10:45 PM 12/7/99 -0500, GeeVarghese John
wrote:
<blockquote type=cite cite>Tmima,
<p>I wanted to ask you some doubts on ACCEPT/REJECT packet in CRTP extension
draft..
<br>It is a continuation of the old mail. In you answer below, though you
wrote the
<br>ACCPET/REJECT packe is optional, I think definition of these packet
are bit confusing.
<br>First of all I feel no need of a ACCEPT packet
<br>Second Why to have a REJECT packet as separate packet to invalidate
the context ..
<br>Why cant we use CONTEXT_STATE packet for the same ..
<br>We can have a bit in CS packet&nbsp; indicating that context need to
be invalidated or
<br>resynchronised using FH packet. If there is no CS packet it is assumed
that
<br>flow ACCEPT.</blockquote>

<p><br>Yes, we suggested two formats for the REJECT packet: one is using
a new code, and the other is using the CS with an extra bit
<br>See attached presentation (presented in DC)
<br>We also prefer to use the 'CS + extra bit' format
<br>Tmima
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>Here the issue of interoperability also arise
when an implementation interopeate with the
<br>old CRTP (2508) implemnetation new implemetation expected ACCEPT or
REJECT packet
<br>for each FH packet .. If it does not receive either packet what action
compressor is expected to take. To make interoperation possible there is
a need for some sort of negotiation, here also a protocol level negotiation
is expected ..
<br>So I feel the approach with ACCEPT/REJECT packet is not optimal.
<br>But the solution I suggested&nbsp; does not cause any interoptation
issue also.
<br>No extra packet overhead, no protocol modification etc&nbsp; ...
<p>Could please answer your points on this ..</blockquote>

<p><br>All the enhancements are not backward compatible so we'll have to
change the negotiation: maybe negotiate a version of CRTP, or negotiate
the new options separately
<br>Tmima
<br>&nbsp;
<br>&nbsp;
<blockquote type=cite cite>Thanks &amp; Rgds
<p>GVJ
<blockquote type=cite cite>&nbsp;
<blockquote type=cite cite><b>Q3) Section III : Acknowledgement to a new
compressed stream</b>
<p>Context Invalidation was one point which I had pointed out to Steve/tmima
long time back ..
<br>I am having those mails with me ..
<p>But, (i) <i>Do we need ACCEPT and REJECT packet&nbsp; ??</i>
<br>Having a NACK mechnism sounds reasonable ..
<br>I feel there shouldnt be both ..
<br>But what compressor is supposed to do after sending the first FULL_HEADER
packet
<br>when compressor waiting for the ACCEPT or REJECT
<br>should it queue the packets till the reply/or timeout
<br>or Should it send the compressed packets assuming the decompressor
will drop if there are
<br>REJECT state at the decompressor side.
<p>But (ii) <i>what about the backword compatibility issue </i>because
this ACCEPT/REJECT mechanism has dependancy and it will not work with CRTP-2508
<p>As per the current proposal we need to send ACCEPT/REJECT packet for
every
<br>FULL_HEADER packet ... (iii)<i>Is it not adding extra bandwidth consuption
</i>..
<br>Draft talks about FULL_HEADER retransmit as well ..</blockquote>
The REJECT packet is useful when the decompressor runs out of resources.
The compressor initiates compression of a stream by sending a FH. With
CRTP today the decompressor can send a CS to invalidate the context, but
the compressor does not know the reason and will attempt to compress this
stream again. A REJECT will notify the compressor that the decompressor
cannot handle compresstion of additional streams at this moment.
<br>An implementation that was not enhanced to use REJECT will still use
CS to invalidate the context each time the compressor attempts to compress
the stream.
<br>The compressor continues to send the compressed packets of the stream
until he receives CS or REJECT
<br>The intent is to make the ACCEPT/REJECT packet optional - an optimization.
<p>Tmima</blockquote>
</blockquote>
</blockquote>
--
<br>-------------------------------------------------------------------------------
<br>[ ]&nbsp; Motorola Confidential Proprietory
<br>[ ]&nbsp; Motorola Internal Use Only
<br>[x]&nbsp; Not an MCP or MIU
<br>-------------------------------------------------------------------------------
<br>Geevarghese John,
<br>DataCom Group,
<br>Motorola India Electronics Ltd,
<br>"The Senate", No-33 A,
<br>Ulsoor Road,
<br>Bangalore - 42.
<br>Phone&nbsp; : 91-80-5598615&nbsp; Xnt-4005
<br>E-mail : gvjohn@miel.mot.com
<br>-------------------------------------------------------------------------------
<br>-------------------------------------------------------------------------------
<br>&nbsp;</html>

--------------2469912007B0B7FB30D56393--




From rem-conf Wed Dec 08 06:44:44 1999 
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To: rem-conf@es.net
Subject: WG last call announcement
Date: Wed, 08 Dec 1999 14:42:22 +0000
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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This message is to announce the start of a two week working group last call
on the following drafts:

	Civanlar and Cash, "RTP Payload Format for Real-Time Pointers",
	draft-ietf-avt-pointer-00.txt 

	Hellstrom, "RTP Payload for Text Conversation",
	draft-ietf-avt-rtp-text-02.txt

If no comments requiring substantive changes are received in the next two
weeks (by 22nd December), these drafts will be submitted for publication as
standards track RFCs.

Colin



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The IESG has approved the Internet-Draft 'Sampling of the Group
Membership in RTP' <draft-ietf-avt-rtpsample-05.txt> as an
Experimental RFC.  This document is the product of the Audio/Video
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From rem-conf Wed Dec 08 08:55:15 1999 
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To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Bill Foster <bfoster@cisco.com>
Subject: Re: Named signaling events - minor request.
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Usually alerting is coming from the far end (callee to caller) as in Q.931.

How about:

Caller => Callee:        Ringing
Caller <= Callee:        Alerting


At 10:05 PM 12/7/99 -0500, Henning Schulzrinne wrote:
>Bill Foster wrote:
> > 
> > One minor comment/request for "draft-ietf-avt-tones-03.txt".
> > 
> > It would be nice to have a separate named signaling event (NSE) for ringing versus ring-back.
>
>This seems reasonable. The only minor complications are naming (since
>there's the potential for confusion between ringing tone = ringback and
>"ring") and that this tone doesn't have a canonical representation. To
>minimize the confusion, I used the term "alert". My suggested wording
>is:
>
>--- begin spec excerpt ---
>
>\item[Alert:] This named signal event causes the recipient to generate
>an alert signal (``ringing''). The
>tone rendition is up to the receiver. (This differs from the ringing
>tone, below, heard by the {\bf caller}.)
>
>--- end excerpt ---
>
>Please let me know if this definition is clear enough.
>
>A meta comment: since the last call periods are over, I will now declare
>a last and final chance. Any tone request, however reasonable, that
>doesn't reach me by tomorrow 5 pm EST will not be included in this round
>so that I can package up -04 and get this to the IESG, finally.
>Additional requests will have to wait
>
> > 
> > The place where a NSE for ringback tone comes in useful is in a device control situation using MGCP or Megacop. The particular case is where the Media Gateway Controller (MGC) requests the media gateway (MG) to do ringback from the far end as follows:
> > 
> > Call between two residential media gateways MG A and MG B
> > - MG A originates a call to MG B
> > - once the MGC sets up a half duplex connection from MG B it requests ringing and ringback to MGB (in MGCP S: rg, rt@connection)
> > - if during the capabilities exchange earlier when the half duplex connection was set up (SDP passed between endpoints), the endpoints realized that they supported ringback tone using NSEs, they might use the NSE rather than a tone (some advantages in terms of use of bandwidth prior to call, as well as implementation advantages - MIPS required etc.).
> > 
> > On the other hand, the same MG may be used in other RTP trunking applications where an MGC may set up the trunk and NSE's used for signaling (e.g. ringing request passed across the RTP stream).
> > 
> > If we leave ringing and ringback as the same event that depends on context (like hook status) there are problems in the trunking application. If one end sends a ringing request to the other end - but when it gets there the person just happens to have gone off-hook (because they wanted to originate a call just as a call was coming in) - if the MG is left to interpret the meaning of the ringing NSE event based on hook-status - they will suddenly hear ringback - not quite what was intended.
> > 
> > In Summary:
> > 
> > Any chance of getting to separate NSE events - one for ringing and another for ringback - they are both useful but only if the are separate events.
> > 
> > Bill
> > -----------------------------
> > Bill Foster
> > Phone: (408) 527-8791
> > Page:   (800) 365-4578
> > Fax:     (781) 998-6492
>
>-- 
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-----------------------------
Bill Foster
Phone: (408) 527-8791
Page:   (800) 365-4578
Fax:     (781) 998-6492



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 			   Call for Papers

		     ACM SIGCOMM 2000 Conference

	      Applications, Technologies, Architectures
	      and Protocols for Computer Communications

		    August 28 - September 1, 2000
		    Grand Hotel, Stockholm, Sweden
		http://www.acm.org/sigcomm/sigcomm2000

	       Sponsored by Ericsson, Sprint, and Telia

Important dates

	Paper submission: 	January 28, 2000
	Tutorial proposals: 	February 28, 2000
	Paper acceptance: 	April 21, 2000

	http://www.acm.org/sigcomm/sigcomm2000
	sigcomm2000-info@acm.org

The SIGCOMM 2000 conference seeks papers describing significant
research contributions to the field of computer and data communication
networks. Authors are invited to submit full papers concerned with
both theory and practice. Areas of interest include, but are not
limited to:

- Distributed application networking infrastructure.
- Distributed common application services, 
  middleware protocols, and signaling.
- Routing, switching, and addressing.
- Resource sharing, quality of service, multimedia networks, 
  and OS support. 
- Multimedia networking.
- Networking aspects of the WWW.
- Heterogeneous internetworking, large-scale networks.
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- Important experimental results from operational networks 
  and lessons learned from prototype implementations. 
- Wireless networking and support for nomadic computing.
- Analysis and design of computer network architectures and algorithms.

SIGCOMM 2000 is a single-track, highly selective conference at which
successful submissions typically report results firmly substantiated
by experiment, implementation, simulation, or mathematical analysis.
In addition to the technical program (paper presentations), SIGCOMM
2000 will offer tutorials by noted instructors on the two days
preceding the actual conference, and a session during the conference
at which speakers may present speculative results and outrageous
opinions.

Submission Instructions: 
------------------------

Papers must be less than 20 double-spaced pages long (formatted for
printing in the Proceedings, papers may not be longer than 12 pages),
have an abstract of 100-150 words, and be original material that has
not been previously published nor is currently under review by another
conference or journal.

Authors must submit papers electronically, using the instructions at
http://www.acm.org/sigcomm/sigcomm2000/submit.  Authors not able to
comply with these instructions should contact the Program Co-Chairs,
or send mail to sigcomm2000-info@acm.org for more information.  Papers
submitted after the deadline will not be considered without an
ahead-of-time extension from the Program Co-Chairs.

All submitted papers will be judged based on their quality and
relevance through double-blind reviewing, where the identities of the
authors are withheld from the reviewers. Consult the on-line
submission instructions for information on preparing a manuscript for
double-blind review.  Authors of accepted papers will need to sign an
ACM copyright release form and present their paper at the
conference. The Proceedings of the conference will be published as a
special issue of ACM SIGCOMM Computer Communication Review. The
Program Committee may also select a few papers for possible
publication in the IEEE/ACM Transactions on Networking. Electronic
copies of the accepted papers will be published on the SIGCOMM 2000
web site prior to the conference unless authors specifically request
that this not be done.

Tutorials: 
----------

SIGCOMM 2000 will begin with two days of full-day and half-day
tutorials covering single topics in detail, at both the introductory
or advanced level. Individuals interested in submitting tutorial
proposals are encouraged to contact the Tutorial Chair before the
deadline to discuss the proposed content.

Student Paper Award: 
--------------------

Papers submitted by students may be considered for a student-paper
award, which includes full conference registration and a partial
travel grant. To be eligible, the student must be the sole author of
the paper, or the first author and primary contributor. A cover letter
or email to the Program Chairs must identify the paper as a candidate
for this competition.

SIGCOMM Award: 
--------------

The keynote speaker at SIGCOMM 2000 will be the 2000 winner of the ACM
SIGCOMM Award for lifetime contributions to the field of computer
communication. Procedures for nominating candidates for the SIGCOMM
Award can be obtained from Craig Partridge (craig@bbn.com).

General Co-Chairs

	Per Gunningberg			Steve Pink
	Uppsala U., Sweden	      	Lulea U. Tech., Sweden
	perg@docs.uu.se		      	steve@cdt.luth.se
	+46 18 471 3171			+46 920 72529

Program Co-Chairs

	Christophe Diot			Jim Kurose
	Sprint ATL, USA			U. Massachusetts, USA
	cdiot@sprintlabs.com		kurose@cs.umass.edu
	+1 650 375-4539			+1 413 545-2742

Publicity Chair

	Joe Touch
	USC/ISI, USA
	touch@isi.edu (also sigcomm2000-info@acm.org)
	+1 310 448-9151

Local Arrangements Co-Chairs

	Bengt Ahlgren			Christian Tschudin
	SICS, Sweden			Uppsala U, Sweden
	Bengt.Ahlgren@sics.se		tschudin@docs.uu.se
	+46 8 633 1562			+46 18 471 1066

Tutorials Chair

	Steve Pink
	Lulea U Tech., Sweden
	steve@cdt.luth.se
	+46 920 72529
--------------F90EB79E2531E0F3907217D3
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   <meta name="Author" content="Joe Touch">
   <meta name="GENERATOR" content="Mozilla/4.7 [en]C-CCK-MCD {Sony}  (Win98; U) [Netscape]">
   <title>Sigcomm 2000 CFP - long</title>
</head>
<body>

<ul>
<center>
<h1>
<b><font color="#000000">Call for Papers</font></b></h1></center>

<center>
<h1>
<b><a href="http://www.acm.org/sigcomm/sigcomm2000">ACM SIGCOMM 2000 Conference</a></b></h1></center>

<center>
<h2>
Applications, Technologies, Architectures&nbsp;<br>
and Protocols for Computer Communications</h2></center>

<center>
<address>
August 28 - September 1, 2000</address></center>

<center>
<address>
<a href="http://www.grandhotel.se/eng/index.html">Grand Hotel, Stockholm,
Sweden</a></address></center>

<center>
<address>
<a href="http://www.acm.org/sigcomm/sigcomm2000">http://www.acm.org/sigcomm/sigcomm2000</a></address></center>

<center>
<address>
<a href="mailto:sigcomm2000-info@acm.org">sigcomm2000-info@acm.org</a></address></center>
</ul>

<center><table COLS=2 WIDTH="50%" >
<tr>
<td><b>Paper submission:</b></td>

<td><b>January 28, 2000</b></td>
</tr>

<tr>
<td><b>Tutorial proposals:</b></td>

<td><b>February 28, 2000</b></td>
</tr>

<tr>
<td><b>Paper acceptance:</b></td>

<td><b>April 21, 2000</b></td>
</tr>
</table></center>

<ul>
<center>
<address>
<font size=+1>Sponsored by Ericsson, Sprint, and Telia</font></address></center>

<p><br>Click here for:
<ul>
<li>
<a href="http://www.acm.org/sigcomm/sigcomm2000">Conference URL (http://www.acm.org/sigcomm/sigcomm2000)</a></li>

<li>
<a href="#Submission">Submission Instructions</a></li>

<li>
<a href="#Tutorials">Tutorials</a></li>

<li>
<a href="#Student">Student Paper Award</a></li>

<li>
<a href="#Award">Sigcomm Award</a></li>

<li>
<a href="#Committee">Conference Committee</a></li>

<li>
<a href="mailto:sigcomm2000-info@acm.org">information via e-mail (sigcomm2000-info@acm.org)</a></li>
</ul>
</ul>
The SIGCOMM 2000 conference seeks papers describing significant research
contributions to the field of computer and data communication networks.
Authors are invited to submit full papers concerned with both theory and
practice. Areas of interest include, but are not limited to:
<ul>
<li>
Distributed application networking infrastructure.</li>

<li>
Distributed common application services, middleware protocols, and signaling.</li>

<li>
Routing, switching, and addressing.</li>

<li>
Resource sharing, quality of service, multimedia networks, and OS support.</li>

<li>
Multimedia networking.</li>

<li>
Networking aspects of the WWW.</li>

<li>
Heterogeneous internetworking, large-scale networks.</li>

<li>
Network management.</li>

<li>
Active network architectures and protocols.</li>

<li>
Important experimental results from operational networks and lessons learned
>from prototype implementations.</li>

<li>
Wireless networking and support for nomadic computing.</li>

<li>
Analysis and design of computer network architectures and algorithms.</li>
</ul>
SIGCOMM 2000 is a single-track, highly selective conference at which successful
submissions typically report results firmly substantiated by experiment,
implementation, simulation, or mathematical analysis.In addition to the
technical program (paper presentations), SIGCOMM 2000 will offer tutorials
by noted instructors on the two days preceding the actual conference, and
a session during the conference at which speakers may present speculative
results and outrageous opinions.
<h2>
<a NAME="Submission"></a>Submission Instructions:</h2>
Papers must be less than 20 double-spaced pages long (formatted for printing
in the Proceedings, papers may not be longer than 12 pages), have an abstract
of 100-150 words, and be original material that has
<br>not been previously published nor is currently under review by another
conference or journal.
<p>Authors must submit papers electronically, using the instructions at
<a href="http://www.acm.org/sigcomm/sigcomm2000/submit">http://www.acm.org/sigcomm/sigcomm2000/submit</a>.
Authors not able to comply with these instructions should <a href="mailto:cdiot@sprintlabs.com,kurose@cs.umass.edu">contact
the Program Co-Chairs</a>, or send mail to <a href="mailto:sigcomm2000-info@acm.org">sigcomm2000-info@acm.org</a>
for more information.&nbsp; Papers submitted after the deadline will not
be considered without an ahead-of-time extension from the Program Co-Chairs.
<p>All submitted papers will be judged based on their quality and relevance
through double-blind reviewing, where the identities of the authors are
withheld from the reviewers. Consult the on-line submission instructions
for information on preparing a manuscript for double-blind review.&nbsp;
Authors of accepted papers will need to sign an ACM copyright release form
and present their paper at the conference. The Proceedings of the conference
will be published as a special issue of ACM SIGCOMM Computer Communication
Review. The Program Committee may also select a few papers for possible
publication in the IEEE/ACM Transactions on Networking. Electronic copies
of the accepted papers will be published on the SIGCOMM 2000 web site prior
to the conference unless authors specifically request that this not be
done.
<h2>
<a NAME="Tutorials"></a>Tutorials:</h2>

<dl>SIGCOMM 2000 will begin with two days of full-day and half-day tutorials
covering single topics in detail, at both the introductory or advanced
level. Individuals interested in submitting tutorial proposals are encouraged
to contact the Tutorial Chair before the deadline to discuss the proposed
content.</dl>

<h2>
<a NAME="Student"></a>Student Paper Award:</h2>
Papers submitted by students may be considered for a student-paper award,
which includes full conference registration and a partial travel grant.
To be eligible, the student must be the sole author of
<br>the paper, or the first author and primary contributor. A cover letter
or email to the Program Chairs must identify the paper as a candidate for
this competition.
<h2>
<a NAME="Award"></a>SIGCOMM Award:</h2>
The keynote speaker at SIGCOMM 2000 will be the 2000 winner of the ACM
SIGCOMM Award for lifetime contributions to the field of computer communication.
Procedures for nominating candidates for the SIGCOMM Award can be obtained
>from Craig Partridge (<a href="mailto:craig@bbn.com">craig@bbn.com</a>).
<br>&nbsp;
<h2>
<a NAME="Committee"></a>Conference Committee:</h2>

<dl>
<h3>
General Co-Chairs</h3>
</dl>

<center><table COLS=2 WIDTH="75%" >
<tr>
<td>
<address>
Per Gunningberg</address>

<address>
Uppsala U., Sweden</address>

<address>
<a href="mailto:perg@docs.uu.se">perg@docs.uu.se</a></address>

<address>
+46 18 471 3171</address>
</td>

<td>
<address>
Steve Pink</address>

<address>
Lule&aring; U. Tech., Sweden</address>

<address>
<a href="mailto:steve@cdt.luth.se">steve@cdt.luth.se</a></address>

<address>
+46 920 72529</address>
</td>
</tr>
</table></center>

<dl>
<h3>
Program Co-Chairs</h3>
</dl>

<center><table COLS=2 WIDTH="75%" >
<tr>
<td>
<address>
Christophe Diot</address>

<address>
Sprint ATL, USA</address>

<address>
<a href="mailto:cdiot@sprintlabs.com">cdiot@sprintlabs.com</a></address>

<address>
+1 650 375-4539</address>
</td>

<td>
<address>
Jim Kurose</address>

<address>
U. Massachusetts, USA</address>

<address>
<a href="mailto:kurose@cs.umass.edu">kurose@cs.umass.edu</a></address>

<address>
+1 413 545-2742</address>
</td>
</tr>
</table></center>

<dl>
<h3>
Publicity Chair</h3>
</dl>

<center><table COLS=1 WIDTH="75%" >
<tr>
<td>
<address>
Joe Touch</address>

<address>
USC/ISI, USA</address>

<address>
<a href="mailto:touch@isi.edu">touch@isi.edu</a> (also <a href="mailto:sigcomm2000-info@acm.org">sigcomm2000-info@acm.org</a>)</address>

<br>+1 310 448-9151</td>
</tr>
</table></center>

<dl>
<h3>
Local Arrangements Co-Chairs</h3>
</dl>

<center><table COLS=2 WIDTH="75%" >
<tr>
<td>
<address>
Bengt &Aring;hlgren</address>

<address>
SICS, Sweden</address>

<address>
<a href="mailto:Bengt.Ahlgren@sics.se">Bengt.Ahlgren@sics.se</a></address>

<address>
+46 8 633 1562</address>
</td>

<td>
<address>
Christian Tschudin</address>

<address>
Uppsala U, Sweden</address>

<address>
<a href="mailto:tschudin@docs.uu.se">tschudin@docs.uu.se</a></address>

<address>
+46 18 471 1066</address>
</td>
</tr>
</table></center>

<h3>
Tutorials Chair</h3>

<center><table COLS=1 WIDTH="75%" >
<tr>
<td>
<address>
Steve Pink</address>

<address>
Lule&aring; U. Tech., Sweden</address>

<address>
<a href="mailto:steve@cdt.luth.se">steve@cdt.luth.se</a></address>

<address>
+46 920 72529</address>
</td>
</tr>
</table></center>

<br>&nbsp;
</body>
</html>

--------------F90EB79E2531E0F3907217D3--




From rem-conf Wed Dec 08 18:54:40 1999 
From rem-conf-request@es.net Wed Dec 08 18:54:38 1999
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Date: Wed, 08 Dec 1999 21:49:43 -0500
From: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
Organization: Columbia University, Dept. of Computer Science
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To: Bill Foster <bfoster@cisco.com>
CC: rem-conf@es.net
Subject: Re: Named signaling events - minor request.
References: <4.2.0.58.19991206192731.01c805e0@redale.cisco.com> <4.2.0.58.19991208083306.018d8230@redale.cisco.com>
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Bill Foster wrote:
> 
> Usually alerting is coming from the far end (callee to caller) as in Q.931.
> 
> How about:
> 
> Caller => Callee:        Ringing
> Caller <= Callee:        Alerting
> 
Unfortunately, the ITU specs (E.182, p. 2, item 7) seem to insist on
"ringing tone" as the tone heard by the caller, so I think this would be
rather confusing. Other suggestions are welcome.

Henning
-- 
Henning Schulzrinne   http://www.cs.columbia.edu/~hgs



From rem-conf Thu Dec 09 01:31:25 1999 
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From: Chris van der Merwe <chris@arnes.si>
Reply-To: "chris@arnes.si" <chris@arnes.si>
To: "rem-conf@es.net" <rem-conf@es.net>
Subject: Creating Simultaneous Mbone and RealMedia Streams...
Date: Thu, 9 Dec 1999 10:22:26 +0100
Organization: Arnes
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Hi;

We're interested in creating presentations broadcast to the Mbone and to 
RealPlayers. Has anyone had any experience doing this and maybe have some 
advice or cautionary warnings to share?

Also is there a way to convert incoming mbone streams to RealMedia streams 
on the fly?

Regards
  Chris




From rem-conf Thu Dec 09 05:22:52 1999 
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Subject: Re: Creating Simultaneous Mbone and RealMedia Streams...
In-Reply-To: <01BF422F.4A9BE4D0.chris@arnes.si> from Chris van der Merwe at "Dec
 9, 1999 10:22:26 am"
To: "chris@arnes.si" <chris@arnes.si>
Date: Thu, 9 Dec 1999 08:18:39 -0500 (EST)
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The first task would be to get the real* thing to accept audio either on a
socket or on stdin.  If that is possible, then it is relatively simple to
do what you wish.

What about windoze media player?

_J

Chris van der Merwe said:
> Hi;
> 
> We're interested in creating presentations broadcast to the Mbone and to 
> RealPlayers. Has anyone had any experience doing this and maybe have some 
> advice or cautionary warnings to share?
> 
> Also is there a way to convert incoming mbone streams to RealMedia streams 
> on the fly?
> 
> Regards
>   Chris
> 
> 




From rem-conf Thu Dec 09 06:51:47 1999 
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Subject: Re: Creating Simultaneous Mbone and RealMedia Streams...
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actually, i think realplayer supports multicast; and some common
datatypes used in Mbone transmissions (PCM for audio, h.261 for video,
i believe). So you don't need to do conversion, unless you want to
reduce bandwidth.


> Chris van der Merwe said:
> > Hi;
> >
> > We're interested in creating presentations broadcast to the Mbone and to
> > RealPlayers. Has anyone had any experience doing this and maybe have some
> > advice or cautionary warnings to share?
> >
> > Also is there a way to convert incoming mbone streams to RealMedia streams
> > on the fly?
> >
> > Regards
> >   Chris
> >
> >



From rem-conf Thu Dec 09 09:32:02 1999 
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Date: Thu, 09 Dec 1999 09:21:56 -0800
To: Henning Schulzrinne <schulzrinne@cs.columbia.edu>
From: Bill Foster <bfoster@cisco.com>
Subject: Re: Named signaling events - minor request.
Cc: rem-conf@es.net
In-Reply-To: <384F18C7.7184802@cs.columbia.edu>
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 <4.2.0.58.19991208083306.018d8230@redale.cisco.com>
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What about:

Caller => Callee:        Power Ringing
Caller <= Callee:        Audible Ringing or Ringing Tone



At 09:49 PM 12/8/99 -0500, Henning Schulzrinne wrote:
>Bill Foster wrote:
> > 
> > Usually alerting is coming from the far end (callee to caller) as in Q.931.
> > 
> > How about:
> > 
> > Caller => Callee:        Ringing
> > Caller <= Callee:        Alerting
> > 
>Unfortunately, the ITU specs (E.182, p. 2, item 7) seem to insist on
>"ringing tone" as the tone heard by the caller, so I think this would be
>rather confusing. Other suggestions are welcome.
>
>Henning
>-- 
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-----------------------------
Bill Foster
Phone: (408) 527-8791
Page:   (800) 365-4578
Fax:     (781) 998-6492



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From: almeroth@cs.ucsb.edu (Kevin C. Almeroth)
Message-Id: <199912092131.NAA28580@jackson.cs.ucsb.edu>
To: almeroth@cs.ucsb.edu, nonnen@research.bell-labs.com
Subject: CFP:  Integrating Multicast into the Internet (Special Issue of Computer Communications)
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CALL FOR PAPERS:

Special Issue of Computer Communications on
Integrating Multicast into the Internet


CFP URL:      http://www.cs.ucsb.edu/~almeroth/COMCOM:cfp.html
Journal URL:  http://www.elsevier.com/inca/publications/store/5/2/5/4/4/0/


TOPICS:

The amount of bandwidth available in the Internet is increasing
dramatically, both in the backbone networks, as well as in the last mile
(broadband access). One consequence is that the delivery of high-quality
multimedia data will become feasible, and multimedia data, including audio
and video, will become the dominant traffic. As more users gain access to
broadband services, new applications and services will become possible. The
result will be a growing demand for large-scale multimedia applications and
services.

Multicasting as a pure networking solution to the transport of media is
recognized as offering economies-of-scale for large-scale applications.
Multicast is also believed to enable applications which provide service to
thousands, even millions of customers. While there has been significant
research work on multicast, efforts to successfully deploy a production
service have lagged. Reasons range from frequently changing protocols,
management and engineering problems, legacy hardware limitations, traffic
conflicts with unicast, etc.. This special issue focuses on research issues
dealing with the challenges of deploying multicast in the network. Specific
topics include, but are not limited to:

    * Multicast Congestion Control
    * Multicast Overlay Networks
    * Next-Generation Multicast Routing Algorithms
    * Multicast Media Transport
    * Security for Multicast-Based Services                      
    * Multicast Traffic Engineering and Management
    * Multicast Pricing and Quality of Service

GUEST EDITORS:

Kevin Almeroth                    Jorg Nonnenmacher
Department of Computer Science    Network Research Dept, Lucent Technologies
University of California          600-700 Mountain Avenue
Santa Barbara, CA 93106-5110      Murray Hill, NJ 07974-0636
Phone: +1(805) 893-2777           Phone: +1(908) 582-1707
Fax: +1(805) 893-8553             Fax: +1(908) 582-6632
Email: almeroth@cs.ucsb.edu       Email: nonnen@research.bell-labs.com


IMPORTANT DATES:

    Paper Submission Deadline:    February 1, 2000
    Feedback to Authors:          May 1, 2000            
    Final Manuscripts Due:        June 1, 2000
    Publication Date:             Fall 2000


SUBMISSION INSTRUCTIONS:

Authors are requested to send e-mail to one or both of the Guest Editors
(almeroth@cs.ucsb.edu, nonnen@research.bell-labs.com) giving a URL from
where a postscript or PDF file can be downloaded. Successfully received
papers will be acknowledged.


AUTHOR INFORMATION:

Submissions made to the special issue should not have appeared in, or been
submitted to other archival publications. All papers will be subjected to
the journal's usual refereeing process.




From rem-conf Thu Dec 09 14:28:42 1999 
From rem-conf-request@es.net Thu Dec 09 14:28:41 1999
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "Bill Foster" <bfoster@cisco.com>,
        "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
Cc: <rem-conf@es.net>
Subject: RE: Named signaling events - minor request.
Date: Thu, 9 Dec 1999 23:26:02 +0100
Message-ID: <NDBBKHLAILCBFIHAKIKPGEGBCCAA.gunnar.hellstrom@omnitor.se>
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I suggest to keep as close as possible to already established nomenclatur=
e.
Proposal
Caller =3D> Callee:        Ring
Caller <=3D Callee:        Ringing Tone
Motivation:
"RING" is the standardised message from a modem experiencing an incoming
ring. It also matches normal language. I.e. "I heard a ring".

"Ringing Tone" is the E.182 name for the tone back to the caller.
"Alerting" could be used for the same thing since it is the name of the
Q.931 message in that situation.
But we are talking about tones from the PSTN here, aren't we?

Gunnar Hellstr=F6m


-----Original Message-----
From: Bill Foster [mailto:bfoster@cisco.com]
Sent: Thursday, December 09, 1999 6:22 PM
To: Henning Schulzrinne
Cc: rem-conf@es.net
Subject: Re: Named signaling events - minor request.


What about:

Caller =3D> Callee:        Power Ringing
Caller <=3D Callee:        Audible Ringing or Ringing Tone



At 09:49 PM 12/8/99 -0500, Henning Schulzrinne wrote:
>Bill Foster wrote:
> >
> > Usually alerting is coming from the far end (callee to caller) as in
Q.931.
> >
> > How about:
> >
> > Caller =3D> Callee:        Ringing
> > Caller <=3D Callee:        Alerting
> >
>Unfortunately, the ITU specs (E.182, p. 2, item 7) seem to insist on
>"ringing tone" as the tone heard by the caller, so I think this would be
>rather confusing. Other suggestions are welcome.
>
>Henning
>--
>Henning Schulzrinne   http://www.cs.columbia.edu/~hgs

-----------------------------
Bill Foster
Phone: (408) 527-8791
Page:   (800) 365-4578
Fax:     (781) 998-6492





From rem-conf Thu Dec 09 15:12:27 1999 
From rem-conf-request@es.net Thu Dec 09 15:12:26 1999
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To: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>,
        "Henning Schulzrinne" <schulzrinne@cs.columbia.edu>
From: Bill Foster <bfoster@cisco.com>
Subject: RE: Named signaling events - minor request.
Cc: <rem-conf@es.net>
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Sounds fine.

At 11:26 PM 12/9/99 +0100, Gunnar Hellstrom wrote:
>I suggest to keep as close as possible to already established nomenclature.
>Proposal
>Caller =3D> Callee:        Ring
>Caller <=3D Callee:        Ringing Tone
>Motivation:
>"RING" is the standardised message from a modem experiencing an incoming
>ring. It also matches normal language. I.e. "I heard a ring".
>
>"Ringing Tone" is the E.182 name for the tone back to the caller.
>"Alerting" could be used for the same thing since it is the name of the
>Q.931 message in that situation.
>But we are talking about tones from the PSTN here, aren't we?
>
>Gunnar Hellstr=F6m
>
>
>-----Original Message-----
>From: Bill Foster [mailto:bfoster@cisco.com]
>Sent: Thursday, December 09, 1999 6:22 PM
>To: Henning Schulzrinne
>Cc: rem-conf@es.net
>Subject: Re: Named signaling events - minor request.
>
>
>What about:
>
>Caller =3D> Callee:        Power Ringing
>Caller <=3D Callee:        Audible Ringing or Ringing Tone
>
>
>
>At 09:49 PM 12/8/99 -0500, Henning Schulzrinne wrote:
> >Bill Foster wrote:
> > >
> > > Usually alerting is coming from the far end (callee to caller) as in
>Q.931.
> > >
> > > How about:
> > >
> > > Caller =3D> Callee:        Ringing
> > > Caller <=3D Callee:        Alerting
> > >
> >Unfortunately, the ITU specs (E.182, p. 2, item 7) seem to insist on
> >"ringing tone" as the tone heard by the caller, so I think this would be
> >rather confusing. Other suggestions are welcome.
> >
> >Henning
> >--
> >Henning Schulzrinne   http://www.cs.columbia.edu/~hgs
>
>-----------------------------
>Bill Foster
>Phone: (408) 527-8791
>Page:   (800) 365-4578
>Fax:     (781) 998-6492
>
>

-----------------------------
Bill Foster
Phone: (408) 527-8791
Page:   (800) 365-4578
Fax:     (781) 998-6492



From rem-conf Thu Dec 09 20:25:22 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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Date: Thu, 9 Dec 1999 20:17:01 -0800
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To: rem-conf@es.net
Subject: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
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I am surprised at the silence on this list regarding this question. Is my
question way off or does this point to an obvious limitation of the RFC?

I forwarded my message to one of the authors of RFC 2250. He suggested that the
solution to my example scenario is that P13 and P23 would have different
timestamps. I disagreed because 1) they need not have different timestamps and
2) even if they are different there is no mechanism in RTP headers to
distinguish them. I have not heard from him since!

Suggestions?

Thanks
Ram
--- Forwarded mail from <kordale> ("Ram Kordale")

From: "Ram Kordale" <kordale>
Date: Fri, 3 Dec 1999 15:26:06 -0800
To: rem-conf@es.net
Subject: MPEG-1 over RTP (RFC 2250) conflict?

Could someone please clarify the following "conflict" I see in the spec?

In the event of data loss, RFC 2250 has mechanisms to recover at the slice
level(basically by discarding RTP packets until you find one with the
BeginningOfSlice bit set to 1).

However, if you lose a few RTP packets, one is **really forced** to skip
until the beginning of the next GOP!

This is an incredible amount of loss or I have misunderstood the spec. Could
someone please clarify?


Details:
-------

Consider the following segment of MPEG-1 system stream.
...GOP_H P11 P12 P13 ... P1n GOP_H P21 P22 P23 ...

GOP_H refers to GOP Headers. Pij refers to picture j of the ith GOP.

For simplicity, assume that the stream and display order are the same. Thus,
Pij will have a "Temporal Reference" of j. For simplicity, assume each
picture is split into 5 RTP packets.

Suppose I receive RTP packets containing all of the first GOP header and
P11 and I receive 1 RTP packet containing the first "piece" of P12.

Suppose the next RTP packet I received contains the "Temporal Reference"
of 3, how do I know which GOP this picture belongs to. In other words, how
do I know it belongs to P13 or P23 or in general Pi3.

If I cannot know, I will have to insert a broken GOP thus essentially
losing data until the next GOP header is received.

Thanks for your help.
Ram

--
Ram Kordale
Advanced Media Products


---End of forwarded mail from <kordale> ("Ram Kordale")

-- 
Ram Kordale
Advanced Media Products



From rem-conf Fri Dec 10 01:20:15 1999 
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Message-ID: <01BF42F6.F3115290.chris@arnes.si>
From: Chris van der Merwe <chris@arnes.si>
Reply-To: "chris@arnes.si" <chris@arnes.si>
To: "'rem-conf@es.net'" <rem-conf@es.net>
Subject: AGAIN: Creating Simultaneous Mbone and RealMedia Streams...
Date: Fri, 10 Dec 1999 10:11:38 +0100
Organization: Arnes
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Sorry, I just checked my original message (reprinted below just to show 
what a pig's ear I made of my explaination). I meant to say that we'd like 
to broadcast to Mbone and REALSERVER at the same time - I guess we could 
just split the original source to two servers, but I'm wandering if there 
is a software solution.

Also: We're considering taking existing Mbone sessions and feeding them to 
RealServer to be transmitted as RealMedia broadcasts - has anyone had any 
experience doing this?
Basically looking at using RealServer with the Mbone.

Sorry for my lack of detail yesterday.

Regards
  Chris

-----Original Message-----
From:	Chris van der Merwe [SMTP:chris@arnes.si]
Sent:	Thursday, December 09, 1999 10:22 AM
To:	rem-conf@es.net
Subject:	Creating Simultaneous Mbone and RealMedia Streams...

Hi;

We're interested in creating presentations broadcast to the Mbone and to 
RealPlayers. Has anyone had any experience doing this and maybe have some 
advice or cautionary warnings to share?



Also is there a way to convert incoming mbone streams to RealMedia streams 
on the fly?

Regards
  Chris




From rem-conf Fri Dec 10 07:09:43 1999 
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To: minutes@ietf.org, rem-conf@es.net
Subject: AVT minutes
cc: casner@cisco.com
Date: Fri, 10 Dec 1999 14:58:52 +0000
Message-ID: <4251.944837932@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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Folks,

Attached are the AVT minutes from the Washington DC meeting, our apologies
for the late submission of these. In addition, copies of the slides from
the meeting may be found at
	http://www-mice.cs.ucl.ac.uk/multimedia/misc/avt/IETF46/slides/

Colin & Steve

-------------------------------------------------------------------------------
Audio/Video Transport Working Group Minutes

Reported by Colin Perkins and Steve Casner

The audio/video transport working group held two full sessions in Washington 
DC. The first session discussed the ongoing revision to the RTP specification 
and audio/video profile, together with a number of  payload formats. The second 
session dealt with transport of MPEG-4 and multiplexing. In addition, a design 
team met to discuss the transport of MPEG-4.

The meeting started with a review of work in progress and the status of the 
working group's documents. The RTP payload format for generic FEC (draft-ietf-
avt-fec-08.txt) has been approved for publication as a proposed standard, and 
the guidelines for writers of RTP payload format specifications (draft-ietf-
avt-rtp-format-guidelines-04.txt) has been approved for publication as a BCP. 
At the time of writing, neither of these RFCs have been issued yet.

The RTP MIB (draft-ietf-avt-rtp-mib-07.txt) has been sent to the IESG for
consideration as a proposed standard. The draft on sampling group
membership in RTP (draft-ietf-avt-rtpsample-05.txt) was published as an
Experimental RFC after the meeting. In addition, the payload format for
DTMF and related tones (draft-ietf-avt-tones-02.txt) was noted as having
completed working group last call and will be submitted for publication
after a final revision is posted. 

An update on the the RTP specification and audio/video profile was presented by 
Steve Casner. The recent changes to the RTP specification include:
	- Collision/loop detection algorithm nesting syntax changed to pseudo-C
	- Added SHOULD keep packets from new address on third-party collision 
	  for apps that may change addresses (e.g., where some endpoints are 
	  mobile)
	- Reverse reconsideration algorithm SHOULD be performed to possibly 
 	  reduce the delay before sending first SR packet after starting to 
 	  send data
	- Timing out a participant is based on inactivity for a number of RTCP
 	  report intervals calculated using the receiver RTCP bw fraction even 
 	  for active senders
There are three open issues with this specification:
	- It currently says that sources which change their source transport 
 	  address MUST change SSRC - is this inconsistent with "SHOULD follow 
 	  new address"?  It was agreed to change MUST to MAY.
	- Should the length of compound RTCP packets with information about N 
	  sources be divided by N for average?  Analysis since the meeting 
	  indicates this is not a problem.
	- It is necessary to clarify that a mixer mixing N sources may use N 
	  times the single-source RTCP bandwidth.

The recent changes to the audio/video profile include:
	- CN (comfort noise) static payload type changed back from 19 to 13 for 
	  consistency with ITU H.225.0 and VoIP Forum documents
	- RTP clock rate G722 changed back to 8000 even though sampling rate is 
	  16000 (a note of explanation was added)
	- Note that MPEG audio RTP clock rate is always 90000 regardless of 
	  sampling rate
Once again, there are a number of minor open issues:
	- Should the profile recommend the use of the H263-1998 payload format 
	  over the old payload format for H263? It was decided yes.
	- Did AIFF-C list of channels change from what is in RFC 1890?
	  No, but the set used for DAT and DV video are different and
	  so may require separate specification in a payload format.
	- Change title of Section 3 to "IANA Considerations".
	- Fix bungled edit of CN from 13 to 19 in Table 5.

The question of how to deal with proposed enhancements to the comfort noise 
payload format to include spectral shape parameters was raised. This work is 
ongoing in the ITU-T SG16, but will take time to complete - we don't want to 
wait for this, before we advance RTP. There are a number of options:
	- Leave the current comfort noise payload format as is, and define a 
	  new payload format for the extension
	- Enhance this format, so if it is longer than 1 byte it includes extra 
	  spectral information
	- Remove this payload format from the profile, so as not to hold the 
	  profile up, but leave the payload type assigned (with a note being 
	  added to the profile that payload type 13 is reserved for a comfort 
	  noise payload format which will be specified later).
The chairs proposed taking this latter option, but this raised a number of
questions about the status of the revised comfort noise draft (which is in
use in a number of products) and the ability of the profile to reference
it. After some discussion, a hybrid of these options was chosen: the
comfort noise payload format will be extracted from the audio/video profile
and submitted as a proposed standard RFC, referenced by the profile. Steve
Casner will do this as part of editing the profile.  The ITU-T will extend
this, in the same manner as any other payload format is extended, with the
expectation that it will cycle at proposed standard.

The MIME registration for the payload types defined in the audio/video
profile has been updated (draft-ietf-avt-rtp-mime-01.txt). The changes
include clarification of the mapping from MIME parameters to SDP "a=fmtp"
attributes and the addition of the RED (redundant audio) payload and "type"
parameter for MPEG. It was noted that the payload format for generic FEC,
which is already in the hands of the RFC editor, did not include a MIME
registration for the name "parityfec"; Jonathan Rosenberg volunteered to
write a separate draft for this.

This meeting marked the completion of working group last call on the set of 
drafts that have been prepared for advancement of RTP specification and A/V 
profile to Draft Standard status.  The working group accepted the changes that 
had been made in the drafts posted for last call as well as the few changes 
proposed in response to comments received during the last call.  With those 
changes, the working group agreed that the following drafts should be submitted 
for IESG Last Call:
	draft-ietf-avt-rtp-new-05.txt		(Draft standard)
	draft-ietf-avt-profile-new-06.txt	(Draft standard)
	draft-ietf-avt-rtp-mime-00.txt		(Proposed standard)
	draft-ietf-avt-rtcp-bw-00.txt		(Proposed standard)
This will be done immediately.

The need for interoperability statements from implementors was noted.
Implementors, both academic and vendors, are strongly encouraged to study
draft-ietf-avt-rtp-interop-02.txt and draft-ietf-avt-profile-interop-00.txt, 
and to submit a statement of interoperability to the group. Scott Bradner
(the transport area co-director) noted that it is okay for a neutral third
party to present the results, hiding the identity of the vendors if they do
not wish to make this information available. The chairs will set up a web
page, summarizing the results of interoperability tests conducted.

The next topic was a potential new task for AVT: a unicast RTCP-based
retransmission scheme, presented by Matthew Podolsky. The motivation for
this work (draft-podolsky-avt-rtprx-00.txt) is that there are a number of
proprietry retransmission schemes in use, hindering interoperability, and
no standard currently exists. 

Why unicast? Because unicast retransmission schemes are widely used, and 
because they are well understood, unlike multicast retransmission schemes which 
are difficult to make work in a scalable and timely manner. Further, reliable 
multicast is outside the scope of this working group.

The proposal is for a new RTCP packet type, the "multipurpose ACK", which can 
be used to ACK or NACK a packet with a particular sequence number and up to 16 
surrounding packets. This is flexible, specifying a basic framework for 
requests which can be modified or extended by a protocol subtype field.

A number of items were raised for discussion:
	- Is it necessary to include the SSRC in retransmission requests? This 
	  is good for compatibility with the other RTCP packet types and allows 
	  us to groups NACKs for multiple sessions in one request. Anders Klemets 
	  noted that this allows one to send multiple NACKs in a single packet, 
	  allowing one to NACK multiple unicast RTP sessions. This may be of use 
	  when working with an RTSP server, since the server can ensure that the 
	  SSRCs used in each session are unique. It is unclear how useful this is, 
	  in practise.
	- Is it best to include a protocol subtype field, or to rely on the RTCP 
	  packet type field and out-of-band signalling? The feeling of the 
	  group was that the RTCP packet type was the appropriate place to 
	  demultiplex, and there was no need to include an additional subtype. 
	  Future versions of this draft should specify behaviour profiles, for 
	  example ACK or NACK based retransmission.
	- Should the bit mask of other packets which are being ACKed/NACKed
	  cover preceeding or succeeding packets? It was suggested to do as
	  in the FEC format, with an offset from the first sequence number.
	- How does this work with silence suppression? How long must one wait 
	  for the next packet before deciding it is loss, rather than silence?
It was noted that, in order to make this work, it is necessary to eliminate the 
current limitations on minimum time between RTCP feedback. This may have 
adverse effects of the scalability and congestion control of the protocol and 
probably requires the definition of a new RTP profile, to specify these timers 
and other changes that may be required.

Mark Handley noted that there is an urgent need to deploy congestion
controlled RTP to prevent problems in the wide area network. If the group
is going to work on a new unicast RTP profile including retransmission, it
would be advisable if we could also include congestion control.  In a
discussion after the AVT session, the transport area directors agreed that
it was appropriate for AVT to undertake this work, but that congestion
management must be included.  Since the ECM working group won't have output
right away, Mark Handley and Sally Floyd agreed to write an informational
document that the work on retransmission can refer to.

The remainder of the first day was spent in a discussion of various RTP
payload formats.  The first was an update on the payload format for Real
Time Pointers was presented by Reha Civanlar. The new draft resolves the
issues raised at the previous meeting, and also adds the ability to convey
an indication of the pointer icon (the mapping from code-points to icons is
to be specified out of band). This is now ready for working group last
call.

The RTP payload formats for DV audio (draft-ietf-avt-dv-audio-00.txt) and video 
(draft-ietf-avt-dv-video-01.txt) were presented by Katsushi Kobayashi. Changes 
relating to the audio format include:
	- addition of 20 and 24 bit linear sampling modes, in addition to 12 
	  bit non-linear
	- change MIME encoding from audio/NL12 to audio/DAT12 and add a MIME 
	  registration template
	- remove pseudo-code, since it was obvious
	- define SDP fmtp parameter for analogue pre-emphasis; this may be
	  required for L16 audio within DV as well, and the working
	  group generally agreed that adding an fmtp parameter to the
	  existing L16 payload format was acceptable
The DV video payload has seen the addition of support for D-7 and D-9 
"professional" formats, a simplification of the SDP description and the removal 
of IEEE1394 specific timestamp issues to an appendix, since the D-7 and D-9 
formats suppose a different interface. The MIME registration has yet to be 
done. Two interoperable implementations exist (from the WIDE project and KTH).
These drafts are believed ready for working group last call.

The payload format for MPEG2 AAC was presented by Jim Snyder (standing in for 
Mathias Kretschmer who was ill). Changes have been made relating to 
	- priority vectors
	- fragmentation/grouping
	- a type field for the repair information
This payload format has not yet been implemented, but this is in progress 
(probably not be done by next meeting, but by the one after). The draft is not 
yet ready for last call, since the authors need to do more development and 
testing, and desire feedback from the group (especially relating to MPEG-4).

Ross Finlayson presented an update to the more loss-tolerant payload format
for MP3 audio (draft-ietf-avt-rtp-mp3-01.txt). This payload format is more
robust to packet loss than RFC2250, but implies more knowledge of MP3.
Recent additions to the draft include a specification of how to handle
streams containing a mix of layer I or II frames with layer III (this is an
uncommon situation) and pseudo-code to convert between "MP3 frames" and
"ADU frames" as required by this format.  A plea was made for the group to
check the pseudo- code, since it's somewhat complex. More information,
including a performance comparison with RFC2250, is available from
http://www.live.com/rtp-mp3/. Dave Singer noted that this is a great idea,
but doesn't go far enough: maybe it should include support for interleaving
too? His group has demonstrated improved performance with interleaving,
reusing the MPEG sync byte so no overhead is added (though doing this
precludes mixing in layer I or II frames).  The addition of interleaving
will be considered before the draft is progressed. 

An extension to the payload format for MPEG-2 transport streams (draft-ietf-
avt-rtp-mp2t-00.txt) was presented by Humphrey Liu. The issue here is that
MP2T is a packet- based transport format which can contain multiple programs 
with an aggregate bandwidth which can be as high as 50-60Mbps (=40Kpps).
The 90KHz RTP timestamp clock defined in RFC2250 does not have sufficient
resolution (about two ticks per packet), so this draft recommends a 27MHz
clock, as defined by MPEG-2.  In fact, RTP with MP2T does not provide much
value compared to UDP for single-program transport streams (at lower data
rates for which 90kHz is appropriate), but the RTP timestamp does add value
for multiple-program transport streams given sufficient resolution.

Why is so much resolution needed? For faster clock recovery, for calculation 
of piece-wise CBR bitrate interpolating the timestamp for every MPEG packet
multiplexed into the RTP-encapsulation transport stream, and in order to
regenerate the exact timing for re-insertion of each packet into a native
MPEG-2 transport stream. No change to the actual payload format in RFC2250 
is required, just a change in the RTP timestamp clock rate. For applications 
which use SDP, this can be accomplished by specifying a clock rate of 27000000 
in the a=rtpmap attribute.

The payload for text conversation (draft-ietf-avt-rtp-text-02.txt) was 
presented by Ian Rytina. The issues resolved since Oslo include:
	- What type of redundancy should be used (FEC or RED)? Agreed to leave 
	  it open, but recommend RED.
	- Why do we need this format when IRC and telnet chat are available? 
	  Compatibility with existing products, better fit with SIP and H.323.
	- MIME registration? Done
	- Clock frequency? Irrelevant for text, but 1kHz now specified.
	- Order of redundant data? Must be inserted in age order with one 
	  redundant block from each transmitted packet, in order to correctly 
	  decode all cases.
	- Avoid double specification of fill character for missing data
	- Transmission during silence periods - send empty primary in the 
	  redundancy format for the last packet before silence.
This payload is now ready for working group last call: the main issue is the 
ordering of redundant data, which is somewhat subtle and should be verified to 
be correct. Jonathan Rosenberg noticed that the MIME type is text/t140, but SDP 
doesn't have a text media (on the m= line). It may be necessary to include a 
section on SDP usage? Steve Casner noted that this should not be necessary 
because the RTP MIME registration specifies a mapping to SDP.

Allison Mankin presented some new work on an RTP payload format for HDTV,
motivated by the start of commercial HTDV broadcasts in the US and the
proliferation of formats. The bandwidth of HDTV streams ranges from 176Mbps
to 1.49Gbps uncompressed, and compresses down to around 20-40Mbps for
broadcast.  These media formats are under standardisation by the ATSC
(http://www.atsc.org) and DVB (http://www.dvb.org) but with major
differences in transmission technology (COFDM vs 8VSB) and payload (eg: DVB
uses MPEG audio, ATSC uses AC-3 Dolby). The payload for compressed HDTV is
mostly covered by RFC2250. The uncompressed format is related to BT.656
with the addition of 4.2.0 colour sub- sampling and a 1080 line mode
(versus 525/625 lines for BT.656). One question is whether the BT.656
payload format can just be extended for HDTV, but here again the RTP
timestamp resolution may be an issue.  A payload is also needed for AC3
audio. This is an early "heads up" on the work, which is ongoing, and
further drafts are expected in future.

The major issue discussed on the second day was the transport of MPEG-4
media.   Colin Perkins started this by summarising the design team meeting
held earlier in the week. There are three proposals for discussion:
	- draft-ietf-avt-rtp-mpeg4-02.txt
	- draft-guillemot-genrtp-01.txt
	- draft-jnb-mpeg4av-rtp-00.txt
It is also expected that there will be a fourth proposal for a
packetization based on MPEG FlexMux. A number of open issues 
were identified with these proposals:
	- Use of MPEG-4 systems vs elementary streams. 
		- MPEG-4 has a complete system model, but some applications just 
		  desire to use the codecs. We accept that we need to generate 
		  payload formats for both cases, even though this has the potential 
		  for interoperability problems later (compare issues with MPEG-2). 
		- We also noted that MPEG-4 encompasses codecs which may have an 
		  existing RTP payload format, but we can't preclude the use of 
		  MPEG-4 specific packetization if the system model is to
		  be maintained. In addition, for error resilience, it is
		  desirable to packetize in a media aware manner - this
		  does not imply a choice between systems or elementary
		  streams.
	- Multiplexing multiple streams.
		- Five multiplexing options were identified: GeRM, FlexMux, "PPP 
		  Mux" with CRTP (see later), don't multiplex, and don't multiplex, 
		  but compress headers. We noted that we may need a FlexMux format, 
		  but nothing else requires special consideration.
	- Grouping within a stream.
		- Why group? to amortize header overhead and to aid error resilience 
		  (for example duplicate and group picture headers with each packet, 
		  or group FEC with media data). 
		- There is disagreement over the importance of grouping, and the 
		  mechanism to be used: draft-guillemot-genrtp-01.txt has support 
		  for grouping access units (ie: ADUs) and for sub-access units (eg: 
		  unequal FEC), the other proposals perform no additional grouping 
		  assuming anything needed will be done in the encoder. 
		- Further experimental testing is needed to determine
		  whether additional error resilience is worth the extra complexity.
	- Fragmentation
		- It is necessary to fragment a codec bit-stream in a media aware 
		  manner to achieve error resilience
		- We believe that the choice of fragmentation is a matter for the 
		  encoder, and that MPEG-4 should be able to fragment accordingly (a 
		  payload format should be able to transport oversize fragments, but 
		  this need not be efficient).
	- Error protection
		- We considered two forms of error protection: within the payload 
		  and across packets.
		- draft-guillemot-genrtp-01.txt uses unequal FEC within the payload 
		  (using a "typed segment" abstraction to be generic - this 
		  abstraction doesn't exist in MPEG, although the information is 
		  available in an ES specific manner). The other proposals assume 
		  the MPEG bitstream is robust enough "as is". Further experimental 
		  testing is needed.
		- We may also apply any of the existing error protection mechanisms 
		  across packets (parity FEC, for example). Draft-guillemot-genrtp-
		  01.txt also duplicates these functions as part of its unequal FEC 
		  scheme.
		- Some MPEG-4 elementary streams MUST be reliably delivered (eg: 
		  control streams and streaming media such as Java class files). We 
		  reached no conclusion on how we do this - have focused on audio 
		  and video to date.
	- Timing model.
		- MPEG-4 and RTP have different timing models. If it is desired to 
		  synchronize MPEG-4 data with data using a native RTP packetization 
		  we must align the models (capture time vs composition time). We 
		  believe the text in draft-ietf-avt-rtp-mpeg4-02.txt describing 
		  this alignment is correct, but needs edits for clarity. This is an 
		  issue for all the proposals.
	- Byte alignment.
		- MPEG-4 systems has the potential to handle not byte aligned bit- 
		  streams (audio and video codecs do not but others may?), but the 
		  streams it produces are byte aligned.  The ES packetizations may 
		  have to similarly include the capability to add padding bits.

Following this summary, brief presentations were made on the individual proposals.
	- Andrea Basso presented draft-ietf-avt-rtp-mpeg4-02.txt, noting that 
	  text has been added describing the correspondance between the RTP and 
	  MPEG-4 timing models (this actually affects all the proposals). 
	  Implementations of this payload format exist by AT&T and Nokia, and 
	  interoperability results were presented to the recent MPEG meeting in   
	  Melbourne.
	- Paul Christ presented an overview of draft-guillemot-genrtp-01.txt 
	  and the motivations for this approach. The draft has not been changed 
	  since the Oslo meeting, but there is a plan to clean up the format and 
	  resubmit before Adelaide. An implmentation is underway - the mapping 
	  and de-mapping is done and under test, the unequal error protection 
	  is under development (see http://www-ks.rus.uni-stuttgart.de/PROJ/GP) 
	  for details. 
	- Yoshihiro Kikuchi presented draft-jnb-mpeg4av-rtp-00.txt. This is a 
	  new draft, which comprises a direct mapping of elementary streams to 
	  RTP, for applications like IP phone, mobile, and H.323. The MPEG-4 
	  systems model is not used, session/stream management is done via 
	  H.245 or SIP. A standards track payload format is needed, for ITU 
	  interaction, since an H.323 audio codepoint will be specified in 
	  February 2000. MPEG-4 video is mapped directly, audio is mapped via 
	  the LATM encapsulation. The draft also specifies back-channel signaling 
	  using RTCP.

The discussion of MPEG-4 was concluded by Colin Perkins, who outlined the 
chairs' view on future activity in this area. 
	- It was noted that the MPEG-4 audio and video codec elementary
	  streams can be packetized in RTP in a manner similar to other
	  codecs.  This path is well understood and makes sense to
	  standardize.  Standardization is also required for these payload
	  formats to be referenced by the ITU. Hence we propose to adopt
	  draft-jnb-mpeg4av-rtp-00.txt as an AVT work item, for eventual
	  submission on the standards track.
	- Multiplexed MPEG-4 media is to be treated in a similar manner to 
	  earlier bundled MPEG transport. We will therefore consider a FlexMux 
	  payload format, if one is submitted.
	- We do not believe we fully understand the issues involved in the 
	  transport of the complete MPEG-4 system over RTP. Such payload 
	  formats should therefore be submitted for publication as experimental 
	  RFCs, whilst we gain implementation experience. Accordingly, draft-
	  ietf-avt-rtp-mpeg4-02.txt and draft-guillemot-genrtp-01.txt will be 
	  progressed to experimental RFC status. 
The chairs asked for - and received - consensus that this is a viable approach. 

An a postscript to the MPEG-4 discussion, Dave Singer provided an overview
of draft-singer-rtp-qtfile-01.txt, the QuickTime file format and hint track
specification. This was presented for information and comments, it is not
intended to be published on the standards track.

The next subject was RTP multiplexing. The first presentation in this area was 
by Bruce Thompson on extensions to CRTP and RTP multiplexing using tunnels. The 
original work has was draft-wing-avt-tcrtp-00 which was submitted to the Oslo 
meeting. This was been broken into three parts: IP tunneling, PPP multiplexing, 
and enhancements to CRTP.

Three independent enhancements to CRTP are described in draft-koren-avt-crtp-
enhance-00.txt. The first is the ability to include a delta-timestamp in 
COMPRESSED_UDP packets to restore the state at the decompressor in the event 
of packet loss. This makes CRTP more robust to packet loss and also lets you 
refresh state and send it multiple times efficiently without the delay of 
error feedback.  Concern was expressed about the need to send the absolute 
value of IPv4 ID, which could be addressed by using the COMPRESSED_NON_TCP 
packet instead except that doesn't include the delta-T. The second extension 
is to add a NonRTP stream flag to the FULL_HEADER packet to notify the 
decompressor that the compressor will never send COMPRESSED_RTP packets for 
a particular context so that the decompressor does not have to maintain 
context for the RTP header.  The third extension is to define a reject packet 
to allow a decompressor to reject the use of a new compression context when 
it is out of resources.

An overview of how this enhanced CRTP can be used for tunneling to provide a 
transparent and efficient multiplexing solution was also provided. RTP packets 
are encapsulated end-to-end for multiplexing, consisting of 
	- compression (CRTP, with enhancements for robustness to loss)
	- multiplexing (PPP multiplexing, draft-ietf-pppext-pppmux-00.txt, 
	  which is ongoing in the PPPEXT working group).
	- IP tunneling for PPP (L2TP, draft-ietf-pppext-l2tphc-03.txt, not
	  posted) which removes session ID, tunnel ID from L2TP, and the
	  UDP header for efficiency, using a negotiated IP protocol ID.
	- CRTP negotiation (RFC2509)
The application runs RTP with no knowledge of the tunnel/compression: CRTP
and muxing take place at layer 2, independent of the application. It was
noted that it is also possible to compress away the IP header on a per link
basis, if desired.  An informational document is to be produced, describing
how these pieces fit together to provide the complete solution.

Steve Casner noted that this approach to the entire muxing problem is clean
and moves the multiplexing down to the IP layer where it is more appropriate, 
keeping the RTP semantics tidy.  Input from the group was solicited on
the changes for CRTP.

The final presentation was by Colin Perkins, for Alexander Tulai who was unable 
to attend, on a dynamic Nx64 payload format (draft-tulai-avt-dynamic-nx64-00.txt), 
for multiplexing PCM coded audio streams (eg: transport of a T1 line).  The 
consensus of the group was that this draft is too narrowly focused, but a payload 
format for general circuit emulation may make sense.

The meeting concluded with a discussion of the working group charter. The main 
goal of the group has been the development of RTP; once this is done we have 
concluded most of our work. We also have steps in the charter for MPEG-4 and 
multiplexing, but we expect to complete these and our other remaining actions 
over the course of the next couple of meeting. This will leave only the 
continuing evolution of payload formats - these don't necessarily need AVT to 
meet everytime, since much of this work can be done on the mailing list.

There are a number of options for the future of the group: one possibility is 
that we may go into hibernation, meeting occasionally when there is sufficient 
work. It was suggested that we could separate the RTP protocol itself from the 
payload question: payloads don't necessarily have to be done in the RTP working 
group (although AVT - or its successor - may review such payload formats), and 
we could recharter AVT to do extensions of the core RTP protocol/profile only.
Another option would be to recharter to study new profiles for unicast RTP 
retransmission and/or congestion control. Input from the working group is 
solicited.




From rem-conf Sat Dec 11 17:19:22 1999 
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From rem-conf Mon Dec 13 00:54:45 1999 
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Date: Mon, 13 Dec 1999 00:35:08 -0800 (Pacific Standard Time)
From: Stephen Casner <casner@cisco.com>
To: Ram Kordale <kordale@relic.engr.sgi.com>
cc: rem-conf@es.net
Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
In-Reply-To: <9912092017.ZM15428@relic.engr.sgi.com>
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On Thu, 9 Dec 1999, Ram Kordale wrote:

> I forwarded my message to one of the authors of RFC 2250. He suggested that the
> solution to my example scenario is that P13 and P23 would have different
> timestamps. I disagreed because 1) they need not have different timestamps and
> 2) even if they are different there is no mechanism in RTP headers to
> distinguish them. I have not heard from him since!

Your 1) is not correct.  P13 and P23 must have different timestamps.
Even P12 and P13 must have different timestamps.  That is how the
multiple packets of one picture are distinguished from the multiple
packets of another picture.  This is a characteristic of most video
payload formats for RTP.

Your 2) I don't understand.  The timestamps are in the RTP header and
they are the mechanism to distinguish the two GOPs.  Granted, one must
be able to tell from the delta between timestamp values whether the
lost packets constitute a gap of one frame or a whole GOP.  If the
frame rate is constant and known, as it usually is for MPEG, then this
is easy.  If the frame rate is variable, you might not be able to
tell.
							-- Steve

> --- Forwarded mail from <kordale> ("Ram Kordale")
> 
> From: "Ram Kordale" <kordale>
> Date: Fri, 3 Dec 1999 15:26:06 -0800
> To: rem-conf@es.net
> Subject: MPEG-1 over RTP (RFC 2250) conflict?
> 
> Could someone please clarify the following "conflict" I see in the spec?
> 
> In the event of data loss, RFC 2250 has mechanisms to recover at the slice
> level(basically by discarding RTP packets until you find one with the
> BeginningOfSlice bit set to 1).
> 
> However, if you lose a few RTP packets, one is **really forced** to skip
> until the beginning of the next GOP!
> 
> This is an incredible amount of loss or I have misunderstood the spec. Could
> someone please clarify?
> 
> 
> Details:
> -------
> 
> Consider the following segment of MPEG-1 system stream.
> ...GOP_H P11 P12 P13 ... P1n GOP_H P21 P22 P23 ...
> 
> GOP_H refers to GOP Headers. Pij refers to picture j of the ith GOP.
> 
> For simplicity, assume that the stream and display order are the same. Thus,
> Pij will have a "Temporal Reference" of j. For simplicity, assume each
> picture is split into 5 RTP packets.
> 
> Suppose I receive RTP packets containing all of the first GOP header and
> P11 and I receive 1 RTP packet containing the first "piece" of P12.
> 
> Suppose the next RTP packet I received contains the "Temporal Reference"
> of 3, how do I know which GOP this picture belongs to. In other words, how
> do I know it belongs to P13 or P23 or in general Pi3.
> 
> If I cannot know, I will have to insert a broken GOP thus essentially
> losing data until the next GOP header is received.
> 
> Thanks for your help.
> Ram
> 
> --
> Ram Kordale
> Advanced Media Products
> 
> 
> ---End of forwarded mail from <kordale> ("Ram Kordale")




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for DTMF Digits, Telephony Tones and 
                          Telephony Signals
	Author(s)	: H. Schulzrinne, S. Petrack
	Filename	: draft-ietf-avt-tones-04.txt,.ps
	Pages		: 29
	Date		: 10-Dec-99
	
This memo describes how to carry dual-tone multifrequency (DTMF)
signaling, other tone signals and telephony events in RTP packets.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-tones-04.txt

Internet-Drafts are also available by anonymous FTP. Login with the username
"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-ietf-avt-tones-04.txt".

A list of Internet-Drafts directories can be found in
http://www.ietf.org/shadow.html 
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by e-mail.

Send a message to:
	mailserv@ietf.org.
In the body type:
	"FILE /internet-drafts/draft-ietf-avt-tones-04.txt".
	
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From rem-conf Mon Dec 13 07:53:57 1999 
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From: kordale@relic.engr.sgi.com (Ram Kordale)
Message-Id: <199912131550.HAA19081@relic.engr.sgi.com>
Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
To: casner@cisco.com (Stephen Casner)
Date: Mon, 13 Dec 1999 07:50:13 -0800 (PST)
Cc: rem-conf@es.net
In-Reply-To: <Pine.WNT.4.21.9912130002170.-378345@revelstoke.cisco.com> from "Stephen Casner" at Dec 13, 99 00:35:08 am
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> 
> On Thu, 9 Dec 1999, Ram Kordale wrote:
> 
> > I forwarded my message to one of the authors of RFC 2250. He suggested that the
> > solution to my example scenario is that P13 and P23 would have different
> > timestamps. I disagreed because 1) they need not have different timestamps and
> > 2) even if they are different there is no mechanism in RTP headers to
> > distinguish them. I have not heard from him since!
> 
> Your 1) is not correct.  P13 and P23 must have different timestamps.
> Even P12 and P13 must have different timestamps.  That is how the
> multiple packets of one picture are distinguished from the multiple
> packets of another picture.  This is a characteristic of most video
> payload formats for RTP.
> 

Thank you very much for the response.

I was interpreting the RTP timestamps for MPEG ES encapsulation to be the
PTS from the MPEG system stream and so I was asserting (1).


> Your 2) I don't understand.  The timestamps are in the RTP header and
> they are the mechanism to distinguish the two GOPs.  Granted, one must
> be able to tell from the delta between timestamp values whether the
> lost packets constitute a gap of one frame or a whole GOP.  If the
> frame rate is constant and known, as it usually is for MPEG, then this
> is easy.  If the frame rate is variable, you might not be able to
> tell.

This is exactly the case I am talking about. I am thinking of the
following solution to the problem. However, it breaks your assertion 
that P12 and P13 must have different RTP timestamps. I would really 
appreciate the group's input on the pros and cons. 

Posssible solution: Let all pictures between two I-pictures (including 
the first I-picture) be called a "mini-GOP"; then, the following apply:
1) All RTP packets for pictures in a mini-GOP must have the same RTP
timestamp.
2) RTP packets for pictures from different mini-GOPs must have different
RTP timestamps.

This gives the mechanism to recover from (2) and the ability to
reconstruct valid and useful GOP headers.

Ram
> 							-- Steve
> 
> > --- Forwarded mail from <kordale> ("Ram Kordale")
> > 
> > From: "Ram Kordale" <kordale>
> > Date: Fri, 3 Dec 1999 15:26:06 -0800
> > To: rem-conf@es.net
> > Subject: MPEG-1 over RTP (RFC 2250) conflict?
> > 
> > Could someone please clarify the following "conflict" I see in the spec?
> > 
> > In the event of data loss, RFC 2250 has mechanisms to recover at the slice
> > level(basically by discarding RTP packets until you find one with the
> > BeginningOfSlice bit set to 1).
> > 
> > However, if you lose a few RTP packets, one is **really forced** to skip
> > until the beginning of the next GOP!
> > 
> > This is an incredible amount of loss or I have misunderstood the spec. Could
> > someone please clarify?
> > 
> > 
> > Details:
> > -------
> > 
> > Consider the following segment of MPEG-1 system stream.
> > ...GOP_H P11 P12 P13 ... P1n GOP_H P21 P22 P23 ...
> > 
> > GOP_H refers to GOP Headers. Pij refers to picture j of the ith GOP.
> > 
> > For simplicity, assume that the stream and display order are the same. Thus,
> > Pij will have a "Temporal Reference" of j. For simplicity, assume each
> > picture is split into 5 RTP packets.
> > 
> > Suppose I receive RTP packets containing all of the first GOP header and
> > P11 and I receive 1 RTP packet containing the first "piece" of P12.
> > 
> > Suppose the next RTP packet I received contains the "Temporal Reference"
> > of 3, how do I know which GOP this picture belongs to. In other words, how
> > do I know it belongs to P13 or P23 or in general Pi3.
> > 
> > If I cannot know, I will have to insert a broken GOP thus essentially
> > losing data until the next GOP header is received.
> > 
> > Thanks for your help.
> > Ram
> > 
> > --
> > Ram Kordale
> > Advanced Media Products
> > 
> > 
> > ---End of forwarded mail from <kordale> ("Ram Kordale")
> 


-- 
Ram Kordale
Advanced Media Products



From rem-conf Mon Dec 13 08:27:16 1999 
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Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
From: "M. Reha Civanlar" <civanlar@research.att.com>
To: "Ram Kordale" <kordale@relic.engr.sgi.com>,
	"Stephen Casner" <casner@cisco.com>
Cc: <rem-conf@es.net>
References: <199912131550.HAA19081@relic.engr.sgi.com>
Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
Date: Mon, 13 Dec 1999 11:25:05 -0500
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----- Original Message -----

> I was interpreting the RTP timestamps for MPEG ES
encapsulation to be the
> PTS from the MPEG system stream and so I was asserting
(1).
>
>
> > Your 2) I don't understand.  The timestamps are in the
RTP header and
> > they are the mechanism to distinguish the two GOPs.
Granted, one must
> > be able to tell from the delta between timestamp values
whether the
> > lost packets constitute a gap of one frame or a whole
GOP.  If the
> > frame rate is constant and known, as it usually is for
MPEG, then this
> > is easy.  If the frame rate is variable, you might not
be able to
> > tell.

The temporal reference field, TR, in the header, which
counts pictures and
resets at each GOP header, is to be used for this purpose.
Please, READ the appendix of RFC 2250.

------------------------------------------------------------
-----------
M. Reha Civanlar
Division Manager, AT&T Labs - Research
100 Schultz Drive, Red Bank, NJ 07701, U.S.A
civanlar@research.att.com
Phone: +1 732 345 3305 Fax:+1 732 345 3033
http://www.research.att.com/info/mrc








From rem-conf Mon Dec 13 10:21:38 1999 
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From: "Ram Kordale" <kordale@relic.engr.sgi.com>
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Date: Mon, 13 Dec 1999 10:18:17 -0800
In-Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
        "Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?" (Dec 13, 11:25am)
References: <199912131550.HAA19081@relic.engr.sgi.com> 
	<013801bf4586$9d3d07a0$4683cf87@research.att.com>
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        "Stephen Casner" <casner@cisco.com>
Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?
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I have read the appendix. Let me make my example simpler. Consider the
following stream:  GOPHdr1 P11 P12 GOPHdr2 P21 P22 GOPHdr3...

For simplicity, I am not distinguishing picture types and maintaining only one
temporal reference counter. Suppose:

1) Each of the pictures is fragmented into 3 RTP packets.
2) Client receives the first RTP packet of the first GOP and loses the next
two.
3) Client then receives first RTP packet corresponding to P22.

The scheme in your appendix will not help the client recognize that a GOP hdr
was lost because the client TR counter is at 2 and the TR field in the RTP
packet is also 2.

Have I assumed something wrongly here?

Thanks.
Ram

On Dec 13, 11:25am, M. Reha Civanlar wrote:
> Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
>
> > > Your 2) I don't understand.  The timestamps are in the
> RTP header and
> > > they are the mechanism to distinguish the two GOPs.
> Granted, one must
> > > be able to tell from the delta between timestamp values
> whether the
> > > lost packets constitute a gap of one frame or a whole
> GOP.  If the
> > > frame rate is constant and known, as it usually is for
> MPEG, then this
> > > is easy.  If the frame rate is variable, you might not
> be able to
> > > tell.
>
> The temporal reference field, TR, in the header, which
> counts pictures and
> resets at each GOP header, is to be used for this purpose.
> Please, READ the appendix of RFC 2250.
>


-- 
Ram Kordale
Advanced Media Products



From rem-conf Mon Dec 13 12:19:17 1999 
From rem-conf-request@es.net Mon Dec 13 12:19:16 1999
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From: "M. Reha Civanlar" <civanlar@research.att.com>
To: "Ram Kordale" <kordale@relic.engr.sgi.com>
Cc: <rem-conf@es.net>
References: <199912131550.HAA19081@relic.engr.sgi.com> <013801bf4586$9d3d07a0$4683cf87@research.att.com> <9912131018.ZM19287@relic.engr.sgi.com>
Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?
Date: Mon, 13 Dec 1999 15:15:21 -0500
Organization: AT&T Labs - Research
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Yes, if you allow for variable framerates AND large variations in the bitrates
AND you loose a large number of packets (covering one or more GOP's
entirely), you may not be able to tell whether you lost a GOP header or not
using the existing mechanisms. However, under these circumstances, I don't think
that you will be able to do much even if you do detect a GOP header loss.

-Reha

----- Original Message -----
From: Ram Kordale <kordale@relic.engr.sgi.com>
To: M. Reha Civanlar <civanlar@research.att.com>; Stephen Casner
<casner@cisco.com>
Cc: <kordale@relic.engr.sgi.com>; <rem-conf@es.net>
Sent: Monday, December 13, 1999 1:18 PM
Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?


> I have read the appendix. Let me make my example simpler. Consider the
> following stream:  GOPHdr1 P11 P12 GOPHdr2 P21 P22 GOPHdr3...
>
> For simplicity, I am not distinguishing picture types and maintaining only one
> temporal reference counter. Suppose:
>
> 1) Each of the pictures is fragmented into 3 RTP packets.
> 2) Client receives the first RTP packet of the first GOP and loses the next
> two.
> 3) Client then receives first RTP packet corresponding to P22.
>
> The scheme in your appendix will not help the client recognize that a GOP hdr
> was lost because the client TR counter is at 2 and the TR field in the RTP
> packet is also 2.
>
> Have I assumed something wrongly here?
>
> Thanks.
> Ram
>
> On Dec 13, 11:25am, M. Reha Civanlar wrote:
> > Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
> >
> > > > Your 2) I don't understand.  The timestamps are in the
> > RTP header and
> > > > they are the mechanism to distinguish the two GOPs.
> > Granted, one must
> > > > be able to tell from the delta between timestamp values
> > whether the
> > > > lost packets constitute a gap of one frame or a whole
> > GOP.  If the
> > > > frame rate is constant and known, as it usually is for
> > MPEG, then this
> > > > is easy.  If the frame rate is variable, you might not
> > be able to
> > > > tell.
> >
> > The temporal reference field, TR, in the header, which
> > counts pictures and
> > resets at each GOP header, is to be used for this purpose.
> > Please, READ the appendix of RFC 2250.
> >
>
>
> --
> Ram Kordale
> Advanced Media Products
>




From rem-conf Mon Dec 13 13:48:00 1999 
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In-Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
        "Re: MPEG-1 over RTP (RFC 2250) conflict?" (Dec 13,  3:15pm)
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I do not see what frame rate and bit rate have to do with this given that they
have no relationship to the number of pictures in a GOP. Given this, losing a
GOP is not abnormal.

However, you may be right in that we should *NOT* be looking for a GOP header
loss after all. We could either
(i) Just recover at the slice level and not care which GOP a picture belongs to
and possibly deteriorate viewing quality.
(ii) OR add provisions in the spec to determine an I picture loss (which is
more significant anyway).

My "mini-GOP" proposal addresses (ii).

I would appreciate comments on (i) vs (ii) and on the proposal for (ii).

Thanks.
Ram

On Dec 13,  3:15pm, M. Reha Civanlar wrote:
> Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?
> Yes, if you allow for variable framerates AND large variations in the
bitrates
> AND you loose a large number of packets (covering one or more GOP's
> entirely), you may not be able to tell whether you lost a GOP header or not
> using the existing mechanisms. However, under these circumstances, I don't
think
> that you will be able to do much even if you do detect a GOP header loss.
>
> -Reha
>
> ----- Original Message -----
> From: Ram Kordale <kordale@relic.engr.sgi.com>
> To: M. Reha Civanlar <civanlar@research.att.com>; Stephen Casner
> <casner@cisco.com>
> Cc: <kordale@relic.engr.sgi.com>; <rem-conf@es.net>
> Sent: Monday, December 13, 1999 1:18 PM
> Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?
>
>
> > I have read the appendix. Let me make my example simpler. Consider the
> > following stream:  GOPHdr1 P11 P12 GOPHdr2 P21 P22 GOPHdr3...
> >
> > For simplicity, I am not distinguishing picture types and maintaining only
one
> > temporal reference counter. Suppose:
> >
> > 1) Each of the pictures is fragmented into 3 RTP packets.
> > 2) Client receives the first RTP packet of the first GOP and loses the next
> > two.
> > 3) Client then receives first RTP packet corresponding to P22.
> >
> > The scheme in your appendix will not help the client recognize that a GOP
hdr
> > was lost because the client TR counter is at 2 and the TR field in the RTP
> > packet is also 2.
> >
> > Have I assumed something wrongly here?
> >
> > Thanks.
> > Ram
> >
> > On Dec 13, 11:25am, M. Reha Civanlar wrote:
> > > Subject: Re: (Fwd) MPEG-1 over RTP (RFC 2250) conflict?
> > >
> > > > > Your 2) I don't understand.  The timestamps are in the
> > > RTP header and
> > > > > they are the mechanism to distinguish the two GOPs.
> > > Granted, one must
> > > > > be able to tell from the delta between timestamp values
> > > whether the
> > > > > lost packets constitute a gap of one frame or a whole
> > > GOP.  If the
> > > > > frame rate is constant and known, as it usually is for
> > > MPEG, then this
> > > > > is easy.  If the frame rate is variable, you might not
> > > be able to
> > > > > tell.
> > >
> > > The temporal reference field, TR, in the header, which
> > > counts pictures and
> > > resets at each GOP header, is to be used for this purpose.
> > > Please, READ the appendix of RFC 2250.
> > >
> >
> >
> > --
> > Ram Kordale
> > Advanced Media Products
> >
>-- End of excerpt from M. Reha Civanlar



-- 
Ram Kordale
Advanced Media Products



From rem-conf Mon Dec 13 14:49:52 1999 
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Reply-To: "M. Reha Civanlar" <civanlar@research.att.com>
From: "M. Reha Civanlar" <civanlar@research.att.com>
To: "Ram Kordale" <kordale@relic.engr.sgi.com>
Cc: <rem-conf@es.net>
References: <199912131550.HAA19081@relic.engr.sgi.com> <013801bf4586$9d3d07a0$4683cf87@research.att.com> <9912131018.ZM19287@relic.engr.sgi.com> <020e01bf45a6$c82dddc0$4683cf87@research.att.com> <9912131345.ZM19535@relic.engr.sgi.com>
Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?
Date: Mon, 13 Dec 1999 17:47:05 -0500
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The bitrate and the framerate get into the picture
because, for conventional rates and applications,
you can use the sequence number and the timestamp
to determine the possibility of a GOP header loss
most of the time even under large packet losses
that cover one or more GOPs.

The problem with your proposal is that it decreases the
accuracy of the timestamps which are used for many
important functions including jitter computation and
media synchronization.

As I mentioned before, if the extent of the packet loss
is such that an entire GOP is lost, knowing that a
GOP header is lost may not be useful for any practical
purpose.

-Reha


----- Original Message -----
From: Ram Kordale <kordale@relic.engr.sgi.com>
To: M. Reha Civanlar <civanlar@research.att.com>
Cc: <rem-conf@es.net>
Sent: Monday, December 13, 1999 4:45 PM
Subject: Re: MPEG-1 over RTP (RFC 2250) conflict?


> I do not see what frame rate and bit rate have to do with this given that they
> have no relationship to the number of pictures in a GOP. Given this, losing a
> GOP is not abnormal.
>
> However, you may be right in that we should *NOT* be looking for a GOP header
> loss after all. We could either
> (i) Just recover at the slice level and not care which GOP a picture belongs
to
> and possibly deteriorate viewing quality.
> (ii) OR add provisions in the spec to determine an I picture loss (which is
> more significant anyway).
>
> My "mini-GOP" proposal addresses (ii).
>
> I would appreciate comments on (i) vs (ii) and on the proposal for (ii).
>
> Thanks.
> Ram





From rem-conf Tue Dec 14 04:06:54 1999 
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From: "Tony Berthaud" <tony.berthaud@thomcast.thomson-csf.com>
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Hello,

Is anybody has information about the MPEG-2 TS over IP ?

Is there a RFC or a DVB/ATSC application notes about this subject ?

Best regards.

Tony




From rem-conf Tue Dec 14 07:22:56 1999 
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Folks,

This is an announcement of the mailing list to be
used for issues concerning robust header compression as
outlined by the robhc BOF held in Washington.

Please take a look at the tentative WG charter

	http://www.cdt.luth.se/~micke/robhc-charter.txt

The mailing list is robhc@cdt.luth.se
Subscribe by sending an email to:     majordomo@cdt.luth.se
where the body of the email contains the single line

subscribe robhc@cdt.luth.se

Let's talk about whether additional TCP header compression 
is needed and specifically what TCP headers we should aim 
to compress.

Seems to me that as a first approximation, compression of 
SACK options, TIMESTAMP options, and T/TCP headers
might be worthwhile over low-bandwidth and high-latency links.

Cheers,

Mikael Degermark



From rem-conf Tue Dec 14 08:26:08 1999 
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this belongs on rem-conf, which handles RTP questions.

-Jonathan R.
-- 
Jonathan D. Rosenberg                       200 Executive Drive
Chief Scientist                             Suite 120 
dynamicsoft                                 West Orange, NJ 07052
jdrosen@dynamicsoft.com                     FAX:   (732) 741-4778
http://www.cs.columbia.edu/~jdrosen         PHONE: (732) 741-7244
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Hi,
I would like to know if it is mandatory to implement Mixers ans
Translators at the terminal.

Regards
Anil



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From rem-conf Tue Dec 14 15:50:21 1999 
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From: mike599@youpy.com
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From rem-conf Thu Dec 16 08:33:35 1999 
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Subject: Reminder: WG last call in progess
Date: Thu, 16 Dec 1999 16:00:18 +0000
Message-ID: <1481.945360018@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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A quick reminder that we are in the midst of a working group last call for
the following drafts:

	Civanlar and Cash, "RTP Payload Format for Real-Time Pointers",
	draft-ietf-avt-pointer-00.txt 

	Hellstrom, "RTP Payload for Text Conversation",
	draft-ietf-avt-rtp-text-02.txt

If no comments requiring substantive changes are received by 22nd December,
these drafts will be submitted for publication as standards track RFCs.

Colin



From rem-conf Thu Dec 16 09:43:02 1999 
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Date: Thu, 16 Dec 1999 09:26:02 -0800
From: Koichi Yano <yano@cs.berkeley.edu>
To: rem-conf@es.net
Subject: RTCP Retransmission
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We are working on refining the RTCP retransmission draft.
I have questions about how to negotiate the type of
retransmission--whether or not retransmissions will be
used, and if so what kind of retransmission
algorithms/behavior will you use.

In the IETF meeting, it was suggested that a new payload type should be
defined in case of deployment of RTCP retransmission.
Is there another possible mechanism for negotiation about
RTCP deployment?

Is it necessary to define a new payload type for each
existing type?  In other words, do we need to
define new dynamic payload types like H261-rx,
H263-rx, DAT12-rx, etc?  When we want to add
retransmission on exisiting static payload type, could
it (a new payload type) become a problem for being
deployed on original implementation?

Koichi Yano




From rem-conf Fri Dec 17 01:30:50 1999 
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Date: Fri, 17 Dec 1999 09:22:28 +0000
From: Nikki Cranley <94426082@eeng.dcu.ie>
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Hi,

I am looking at RTCP-Rx also with respect to MPEG-4. In MPEG-4 the BIFS and
IOD are very important for establishing the transmission and replay of the
MPEG-4 file. I am thinking of using ways to implement the RTCP-Rx to ensure
that these files are received.
I think that there should be new payload types defined to reflect different
media types etc. The intended purpose of RTCP-Rx is selective
retransmisison  and so there need to be selection criteria otherwise its
like TCP.

Regards,
Nikki

Koichi Yano wrote:

> We are working on refining the RTCP retransmission draft.
> I have questions about how to negotiate the type of
> retransmission--whether or not retransmissions will be
> used, and if so what kind of retransmission
> algorithms/behavior will you use.
>
> In the IETF meeting, it was suggested that a new payload type should be
> defined in case of deployment of RTCP retransmission.
> Is there another possible mechanism for negotiation about
> RTCP deployment?
>
> Is it necessary to define a new payload type for each
> existing type?  In other words, do we need to
> define new dynamic payload types like H261-rx,
> H263-rx, DAT12-rx, etc?  When we want to add
> retransmission on exisiting static payload type, could
> it (a new payload type) become a problem for being
> deployed on original implementation?
>
> Koichi Yano




From rem-conf Fri Dec 17 08:35:49 1999 
From rem-conf-request@es.net Fri Dec 17 08:35:49 1999
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Subject: Re: RTCP Retransmission
In-reply-to: Your message of "Fri, 17 Dec 1999 09:22:28 GMT." <385A00D4.CD5CEEAE@eeng.dcu.ie>
Date: Fri, 17 Dec 1999 16:30:22 +0000
Message-ID: <1201.945448222@cs.ucl.ac.uk>
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Nikki Cranley writes:
>I am looking at RTCP-Rx also with respect to MPEG-4. In MPEG-4 the BIFS and
>IOD are very important for establishing the transmission and replay of the
>MPEG-4 file. I am thinking of using ways to implement the RTCP-Rx to ensure
>that these files are received.
>I think that there should be new payload types defined to reflect different
>media types etc. The intended purpose of RTCP-Rx is selective
>retransmisison  and so there need to be selection criteria otherwise its
>like TCP.

For unicast, I wonder if TCP is not the correct means of transporting this
information, and would be interested in arguments why not.  For multicast, 
I remind you of RFC2357 and the work in RMT, which might be appropriate. 

Cheers,
Colin



From rem-conf Fri Dec 17 08:35:52 1999 
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cc: rem-conf@es.net
Subject: Re: RTCP Retransmission
In-reply-to: Your message of "Thu, 16 Dec 1999 09:26:02 PST." <385920AA14.04F5YANO@pop.cs.berkeley.edu>
Date: Fri, 17 Dec 1999 16:26:02 +0000
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From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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--> Koichi Yano writes:
>We are working on refining the RTCP retransmission draft.
>I have questions about how to negotiate the type of
>retransmission--whether or not retransmissions will be
>used, and if so what kind of retransmission
>algorithms/behavior will you use.
>
>In the IETF meeting, it was suggested that a new payload type should be
>defined in case of deployment of RTCP retransmission.
>Is there another possible mechanism for negotiation about
>RTCP deployment?

You could define a new RTP profile, which uses the existing set of payload
formats but adds the retransmission scheme in RTCP (and, perhaps, also adds
congestion control).

That would be signalled in SDP, much like:
	m=audio 49000 RTP/rx-profile 0
with the "0" representing the payload type (PCMU audio, in this case), and
the "RTP/rx-profile" indicating the new profile. 

Colin



From rem-conf Fri Dec 17 08:44:11 1999 
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From: Nikki Cranley <94426082@eeng.dcu.ie>
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Hi Colin -

The problem with TCP is that for real-time or near real time transmission its
not feasible to use TCP as there may be lost packets/delays etc.and
retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
used.
But in the case of MPEG-4 - there are some important packets that probably
shouldn't be lost - imagine if the packet with all the decoding information got
lost - well that would render everything else useless. I am looking at this only
>from the perspective of unicast anyway - plus if we were to extend this to VR -
then RTCP-Rx could be used to dictate user interaction as there is no scope for
this in RTP/RTCP as far as I know. It would only be needed from client to
server.
I'm only new to this but thats what came to mind. I was thinking along the lines
of say incorporating media specific functionality into the RTCP-Rx.

Rgds,
Nikki

Colin Perkins wrote:

> --> Nikki Cranley writes:
> >I am looking at RTCP-Rx also with respect to MPEG-4. In MPEG-4 the BIFS and
> >IOD are very important for establishing the transmission and replay of the
> >MPEG-4 file. I am thinking of using ways to implement the RTCP-Rx to ensure
> >that these files are received.
> >I think that there should be new payload types defined to reflect different
> >media types etc. The intended purpose of RTCP-Rx is selective
> >retransmisison  and so there need to be selection criteria otherwise its
> >like TCP.
>
> For unicast, I wonder if TCP is not the correct means of transporting this
> information, and would be interested in arguments why not.  For multicast,
> I remind you of RFC2357 and the work in RMT, which might be appropriate.
>
> Cheers,
> Colin




From rem-conf Fri Dec 17 08:53:05 1999 
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Subject: Re: RTCP Retransmission
In-reply-to: Your message of "Fri, 17 Dec 1999 16:41:53 GMT." <385A67D1.9FBCAE69@eeng.dcu.ie>
Date: Fri, 17 Dec 1999 16:52:05 +0000
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Hi,

--> Nikki Cranley writes:
>The problem with TCP is that for real-time or near real time transmission its
>not feasible to use TCP as there may be lost packets/delays etc.and
>retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
>used.
>But in the case of MPEG-4 - there are some important packets that probably
>shouldn't be lost - imagine if the packet with all the decoding information got
>lost - well that would render everything else useless. I am looking at this only
>from the perspective of unicast anyway - plus if we were to extend this to VR -
>then RTCP-Rx could be used to dictate user interaction as there is no scope for
>this in RTP/RTCP as far as I know. It would only be needed from client to
>server.
>I'm only new to this but thats what came to mind. I was thinking along the lines
>of say incorporating media specific functionality into the RTCP-Rx.

I was envisaging that RTCP-Rx would be used for streams which need to be
resilient to loss, but real-time and not 100% reliable. That fits for some
types of MPEG-4 content, but other types MUST be delivered with 100%
reliability, and I believe TCP is a better fit for these.

Colin



From rem-conf Fri Dec 17 10:34:05 1999 
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Subject: Re: RTCP Retransmission
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>
>For unicast, I wonder if TCP is not the correct means of transporting this
>information, and would be interested in arguments why not.

My (personal) interest in using RTCP-based re-transmission is to 
cover the case where you want the receiver clock to run, even if data 
never arrives, but you want the greatly improved (but not perfect) 
reliability.  You can't get both guaranteed delivery and on-time 
behavior.

If you are willing to wait indefinitely, I can promise to deliver 
everything (or at least you cannot prove that I won't).  For that 
case, TCP is mostly fine;  the only wrinkle is knowing when you are 
starving because the sender sent nothing, and when you are starving 
because the network is denying you data.

If you are not willing to wait indefinitely, TCP attempting for ever 
to re-transmit packets you no longer care about, and refusing to give 
you later packets that have already arrived, is unhelpful.  This is 
the space I'd like to see a RTCP-based algorithm fit into.

So I don't see it as a multicast/unicast thing, though of course an 
RTCP-based approach might lend itself to multicast.  However, you 
then have to assess questions like "should every recipient on a 
multicast signal loss of a packet, or should they watch each other?", 
"should I re-multicast this packet to everyone or re-unicast it to 
the complainers?", and so on.  Step 2, in my opinion.

David Singer
Apple Computer/QuickTime



From rem-conf Fri Dec 17 11:29:48 1999 
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Colin Perkins wrote:
> 
> Hi,
> 
> --> Nikki Cranley writes:
> >The problem with TCP is that for real-time or near real time transmission its
> >not feasible to use TCP as there may be lost packets/delays etc.and
> >retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
> >used.
> >But in the case of MPEG-4 - there are some important packets that probably
> >shouldn't be lost - imagine if the packet with all the decoding Hi,

Colin Perkins wrote:
> I was envisaging that RTCP-Rx would be used for streams which need to be
> resilient to loss, but real-time and not 100% reliable. 

Basically, I agree with you.

But we can prepare the flag for strong request, which means "send
retransmission surely" wihtout thinking playout time or something.
I think it is a good idea.

By using a receiver's timer and tuning the frequency of re-request of
retransmission, reliability could be tuned.

Because the framework for retransmission request could be extensible to
ACK (not NACK), TCP-like 100% relible protocol could be implemented on
RTCP-rx.

I think just deployment of TCP is the easiest way for 100% reliability,
though.

Koichi



From rem-conf Fri Dec 17 12:22:08 1999 
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Subject: Re: RTCP Retransmission
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I agree that TCP is usable and sufficent for transferring the "vital" part
of the data in a real-time streaming application.

We actually used TCP successfully in transmitting high priority information with
MPEG2 ES transport. RFC2448 outlines our general approach and observations.

I believe, RTCP-Rx needs to be used when there is time for retransmission, but
the data to be retransmitted is loss resilient (i.e. would be useful if received
on-time but, not vital).

-----------------------------------------------------------------------
M. Reha Civanlar


----- Original Message -----
From: Colin Perkins <C.Perkins@cs.ucl.ac.uk>
To: <94426082@eeng.dcu.ie>
Cc: Koichi Yano <yano@cs.berkeley.edu>; <rem-conf@es.net>
Sent: Friday, December 17, 1999 11:52 AM
Subject: Re: RTCP Retransmission


> Hi,
>
> --> Nikki Cranley writes:
> >The problem with TCP is that for real-time or near real time transmission its
> >not feasible to use TCP as there may be lost packets/delays etc.and
> >retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
> >used.
> >But in the case of MPEG-4 - there are some important packets that probably
> >shouldn't be lost - imagine if the packet with all the decoding information
got
> >lost - well that would render everything else useless. I am looking at this
only
> >from the perspective of unicast anyway - plus if we were to extend this to
VR -
> >then RTCP-Rx could be used to dictate user interaction as there is no scope
for
> >this in RTP/RTCP as far as I know. It would only be needed from client to
> >server.
> >I'm only new to this but thats what came to mind. I was thinking along the
lines
> >of say incorporating media specific functionality into the RTCP-Rx.
>
> I was envisaging that RTCP-Rx would be used for streams which need to be
> resilient to loss, but real-time and not 100% reliable. That fits for some
> types of MPEG-4 content, but other types MUST be delivered with 100%
> reliability, and I believe TCP is a better fit for these.
>
> Colin
>
>




From rem-conf Fri Dec 17 12:37:33 1999 
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------- Blind-Carbon-Copy

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CALL FOR PAPERS:

Special Issue of Computer Communications on

      Integrating Multicast into the Internet
      ---------------------------------------

CFP URL:      http://www.cs.ucsb.edu/~almeroth/COMCOM:cfp.html
Journal URL:  http://www.elsevier.com/inca/publications/store/5/2/5/4/4/0/


TOPICS:

The amount of bandwidth available in the Internet is increasing
dramatically, both in the backbone networks, as well as in the last mile
(broadband access). One consequence is that the delivery of high-quality
multimedia data will become feasible, and multimedia data, including audio
and video, will become the dominant traffic. As more users gain access to
broadband services, new applications and services will become possible. The
result will be a growing demand for large-scale multimedia applications and
services.

Multicasting as a pure networking solution to the transport of media is
recognized as offering economies-of-scale for large-scale applications.
Multicast is also believed to enable applications which provide service to
thousands, even millions of customers. While there has been significant
research work on multicast, efforts to successfully deploy a production
service have lagged. Reasons range from frequently changing protocols,
management and engineering problems, legacy hardware limitations, traffic
conflicts with unicast, etc.. This special issue focuses on research issues
dealing with the challenges of deploying multicast in the network. Specific
topics include, but are not limited to:

    * Multicast Congestion Control
    * Multicast Overlay Networks
    * Next-Generation Multicast Routing Algorithms
    * Multicast Media Transport
    * Security for Multicast-Based Services                      
    * Multicast Traffic Engineering and Management
    * Multicast Pricing and Quality of Service

GUEST EDITORS:

Kevin Almeroth                    Jorg Nonnenmacher
Department of Computer Science    Network Research Dept, Lucent Technologies
University of California          600-700 Mountain Avenue
Santa Barbara, CA 93106-5110      Murray Hill, NJ 07974-0636
Phone: +1(805) 893-2777           Phone: +1(908) 582-1707
Fax: +1(805) 893-8553             Fax: +1(908) 582-6632
Email: almeroth@cs.ucsb.edu       Email: nonnen@research.bell-labs.com


IMPORTANT DATES:

    Paper Submission Deadline:    February 1, 2000
    Feedback to Authors:          May 1, 2000            
    Final Manuscripts Due:        June 1, 2000
    Publication Date:             Fall 2000


SUBMISSION INSTRUCTIONS:

Authors are requested to send e-mail to one or both of the Guest Editors
(almeroth@cs.ucsb.edu, nonnen@research.bell-labs.com) giving a URL from
where a postscript or PDF file can be downloaded. Successfully received
papers will be acknowledged.


AUTHOR INFORMATION:

Submissions made to the special issue should not have appeared in, or been
submitted to other archival publications. All papers will be subjected to
the journal's usual refereeing process.




------- End of Blind-Carbon-Copy



From rem-conf Fri Dec 17 14:23:42 1999 
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To: rem-conf@es.net
From: alagu@entera.com (Alagu Periyannan)
Subject: Re: RTCP Retransmission
Cc: yano@cs.berkeley.edu,Colin Perkins <C.Perkins@cs.ucl.ac.uk>
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At 11:27 AM 12/17/99 -0800, Koichi Yano wrote:
>Colin Perkins wrote:
>> I was envisaging that RTCP-Rx would be used for streams which need to be
>> resilient to loss, but real-time and not 100% reliable. 
>
>Basically, I agree with you.
>

I think there is some merit in introducing the notion that certain RTP
packets are indispensible, i.e. 100% reliable. When such packets are lost
the receiver stops the clock and waits.

This RTP feature should be used only in non-interactive (i.e. streaming)
applications. It should not be used for video key frames and other such
error resilient packets.

We can keep saying use TCP for that, why bother enhancing RTP? If I use
TCP, I have to build a hybrid protocol that uses RTP/UDP and TCP to achieve
my goal. Why not make it simpler and offer the functionality in RTP?

This issue has been the cause of the growth of proprietary streaming
protocols by a number of vendors. Most of them have invented their own
protocol and do not use RTP because of this missing functionality. Even
some vendors such as Apple who are using RTP are implementing "repeat"
packets to get around this which is very inefficient.

Let's not preclude a "100% reliability" feature in RTP-Rx.


--

Alagu Periyannan                     alagu@entera.com
Entera, Inc.                         +1 510 770 5225




From rem-conf Fri Dec 17 14:45:42 1999 
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----- Original Message ----- 
From: Alagu Periyannan <alagu@entera.com>

> I think there is some merit in introducing the notion that certain RTP
> packets are indispensible, i.e. 100% reliable. When such packets are lost
> the receiver stops the clock and waits.
> 
> This RTP feature should be used only in non-interactive (i.e. streaming)
> applications. It should not be used for video key frames and other such
> error resilient packets.
> 

I think, what the end point will do when a vital packet is lost is an
implementation issue. Sending a packet with 100% reliability, on the
other hand, is a protocol design issue. TCP does this for unicast and
there is a WG for reliable multicast.

> We can keep saying use TCP for that, why bother enhancing RTP? If I use
> TCP, I have to build a hybrid protocol that uses RTP/UDP and TCP to achieve
> my goal. Why not make it simpler and offer the functionality in RTP?

The problem is that this won't make RTP simpler.

> 
> This issue has been the cause of the growth of proprietary streaming
> protocols by a number of vendors. Most of them have invented their own
> protocol and do not use RTP because of this missing functionality. Even
> some vendors such as Apple who are using RTP are implementing "repeat"
> packets to get around this which is very inefficient.

How can they achieve 100% reliability with "repeat" packets anyway?

-----------------------------------------------------------------------
M. Reha Civanlar







From rem-conf Fri Dec 17 16:29:19 1999 
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From: Matt Podolsky <podolsky@CS.Berkeley.EDU>
To: 94426082@eeng.dcu.ie
CC: C.Perkins@cs.ucl.ac.uk, yano@cs.berkeley.edu, rem-conf@es.net,
        singer@apple.com
In-reply-to: <385A67D1.9FBCAE69@eeng.dcu.ie> (message from Nikki Cranley on
	Fri, 17 Dec 1999 16:41:53 +0000)
Subject: Re: RTCP Retransmission
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> > --> Nikki Cranley writes:
> > >The problem with TCP is that for real-time or near real time 
> transmission its
> > >not feasible to use TCP as there may be lost packets/delays etc.and
> > >retransmission of these will hinder the real-time ability - so 
> RTP/UDP/IP is
> > >used.
> > >But in the case of MPEG-4 - there are some important packets that probably
> > >shouldn't be lost - imagine if the packet with all the decoding Hi,
>
>Colin Perkins wrote:
> > I was envisaging that RTCP-Rx would be used for streams which need to be
> > resilient to loss, but real-time and not 100% reliable.

At 11:27 AM 12/17/99 -0800, Koichi Yano wrote:
>Basically, I agree with you.

At 10:28 AM 12/17/99 -0800, Dave Singer wrote:
>My (personal) interest in using RTCP-based re-transmission is to cover the 
>case where you want the receiver clock to run, even if data never arrives, 
>but you want the greatly improved (but not perfect) reliability.  You 
>can't get both guaranteed delivery and on-time behavior.

I too agree with Colin and Dave's statements.  Our original vision of 
RTCP-Rx was a protocol that was generic enough that it could be applied to 
any RTP stream to improve its reliability, without making any 
guarantees.  We didn't want to have to write (or require others to write) a 
new specification or subtype for every payload out there.

However, as Koichi points out, we could add hooks to better enable a higher 
degree of reliability, such as something like a "retransmit this now, i 
don't care if you've sent it already or you think it won't arrive in time 
to be useful" flag on the receiver requests.  If one chose to one could 
then either write an MPEG-4-specific retransmission algorithm (as specified 
by the RXP subtype) or just write a receiver RTCP-Rx implementation that 
implements slightly different behavior if the payload is MPEG-4.  The 
MPEG-4 algorithm could then require critical information to be 
re-transmitted an unlimited number of times (with proper RTT spacing, of 
course) until it got to the receiver; if it wants to delay playback until 
it gets the critical decoding info perhaps it could couple this with a RTSP 
signal to pause the stream at the sender so the receiver's playback buffer 
doesn't get overrun.

In any case we wanted to avoid going into the specifics of implementing 
variable levels of reliability in the draft.  However, I'm not against 
providing hooks to enable these features, provided they don't cause things 
to become overly complex. And I think a "definitely retransmit" flag like 
the one I've described above would have additional value, specifically in 
distinguishing duplicate retransmission requests.  A receiver might repeat 
a retransmission request multiple times to protect against NACK loss, 
without wanting the sender to send more than one retransmission for the 
whole block of requests (this would be natural if it sent one NACK for each 
lost packet, since the BLP field allows up to 16 neighboring packets to be 
reported as lost).  At other times, the receiver might also repeat a 
retransmission request because the first retransmission was lost (or at 
least it thinks it was) and it has enough time before playout for a second 
attempt.  One way to distinguish between these two cases is to have the 
sender keep track of the time between a retransmission of a packet and a 
new request; if it's more than RTT+delta, it assumes the first 
retransmission failed and retransmits.  But as an alternative the receiver 
could instead simply set a timer when it makes the very  first request 
(possibly repeating it in subsequent NACKs); if the timer expires it sends 
a new request marked as "retransmit this now."  The sender in this case 
would just only send additional retransmissions after the first one if
the "retransmit now" flag was set (note that this means the re-requests can't
themselves be repeated for NACK-loss resilience, but that might be overkill 
anyway).

Still another idea would be to have a payload-specific algorithm (like for 
MPEG-4) in which the sender and receiver both know which pieces parts of 
the stream need full reliability; these would be ACKed instead of NACKed 
(by defining an ACK RXP subtype) and the sender would be responsible for 
reliability, a la TCP.

Matt

-- 

Dad always thought laughter was the best medicine, which I guess is why several
of us died of tuberculosis.

-- "Deep Thoughts" by Jack Handey



From rem-conf Fri Dec 17 17:33:32 1999 
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Subject: Re: RTCP Retransmission
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The only reason I might want to define a 100%-reliable mode in RTP is 
to get around the starvation puzzle I mentioned in a previous 
message:  I'm not sure if there is any way to tell if a TCP end-point 
is not giving you data because (a) it is waiting for a 
re-transmission which doesn't seem to be able to get through or (b) 
it can't get any response out of the source or (c) it is perfectly 
happy and the source isn't sending anything.

The problem with the clock-stop algorithm is that it basically says 
"stop the clock at the time for which you didn't get a packet that 
was sent";  of course, the time to stop the clock is in the packet 
you didn't get.  You can fix this, I guess, by stopping at the 
previous packet's time, or by indicating the next packet time in each 
packet (which also helps solve the starvation problem).
David Singer
Apple Computer/QuickTime



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From rem-conf Mon Dec 20 10:59:34 1999 
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Cc: Koichi Yano <yano@cs.berkeley.edu>, rem-conf@es.net
Subject: RE: RTCP Retransmission
Date: Mon, 20 Dec 1999 10:44:41 -0800
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Early experiments using TCP for streaming at Microsoft several years back seemed to indicate that TCP was an acceptable streaming transport over the Internet if (and only if) the TCP's sliding window was kept very small (e.g., 2 or 3 or 4).

> ----------
> From: 	Nikki Cranley[SMTP:94426082@eeng.dcu.ie]
> Reply To: 	94426082@eeng.dcu.ie
> Sent: 	Friday, December 17, 1999 8:41 AM
> To: 	Colin Perkins
> Cc: 	Koichi Yano; rem-conf@es.net
> Subject: 	Re: RTCP Retransmission
> 
> Hi Colin -
> 
> The problem with TCP is that for real-time or near real time transmission its
> not feasible to use TCP as there may be lost packets/delays etc.and
> retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
> used.
> But in the case of MPEG-4 - there are some important packets that probably
> shouldn't be lost - imagine if the packet with all the decoding information got
> lost - well that would render everything else useless. I am looking at this only
> from the perspective of unicast anyway - plus if we were to extend this to VR -
> then RTCP-Rx could be used to dictate user interaction as there is no scope for
> this in RTP/RTCP as far as I know. It would only be needed from client to
> server.
> I'm only new to this but thats what came to mind. I was thinking along the lines
> of say incorporating media specific functionality into the RTCP-Rx.
> 
> Rgds,
> Nikki
> 
> Colin Perkins wrote:
> 
> > --> Nikki Cranley writes:
> > >I am looking at RTCP-Rx also with respect to MPEG-4. In MPEG-4 the BIFS and
> > >IOD are very important for establishing the transmission and replay of the
> > >MPEG-4 file. I am thinking of using ways to implement the RTCP-Rx to ensure
> > >that these files are received.
> > >I think that there should be new payload types defined to reflect different
> > >media types etc. The intended purpose of RTCP-Rx is selective
> > >retransmisison  and so there need to be selection criteria otherwise its
> > >like TCP.
> >
> > For unicast, I wonder if TCP is not the correct means of transporting this
> > information, and would be interested in arguments why not.  For multicast,
> > I remind you of RFC2357 and the work in RMT, which might be appropriate.
> >
> > Cheers,
> > Colin
> 
> 



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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Audio/Video Transport Working Group of the IETF.

	Title		: RTP Payload for DTMF Digits, Telephony Tones and 
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	Author(s)	: H. Schulzrinne, S. Petrack
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	Date		: 21-Dec-99
	
This memo describes how to carry dual-tone multifrequency (DTMF)
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From rem-conf Thu Dec 23 06:28:56 1999 
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To: yano@cs.berkeley.edu
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Subject: Re: RTCP Retransmission
In-Reply-To: Message from Koichi Yano <yano@cs.berkeley.edu> of "Fri, 17 Dec 1999 11:27:34 PST." <385A8EA6.4F0B46D0@cs.berkeley.edu>
Date: Thu, 23 Dec 1999 14:11:56 +0000
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--> Koichi Yano writes:
>But we can prepare the flag for strong request, which means "send
>retransmission surely" wihtout thinking playout time or something.
>I think it is a good idea.

But not forgetting congestion control, right?

Colin



From rem-conf Thu Dec 23 06:28:56 1999 
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To: alagu@entera.com (Alagu Periyannan)
cc: rem-conf@es.net, yano@cs.berkeley.edu
Subject: Re: RTCP Retransmission
In-Reply-To: Message from alagu@entera.com (Alagu Periyannan) of "Fri, 17 Dec 1999 14:28:18 PST." <3.0.6.32.19991217142818.01530650@mailserver.entera.com>
Date: Thu, 23 Dec 1999 13:53:05 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Alagu Periyannan writes:
>At 11:27 AM 12/17/99 -0800, Koichi Yano wrote:
>>Colin Perkins wrote:
>>> I was envisaging that RTCP-Rx would be used for streams which need to be
>>> resilient to loss, but real-time and not 100% reliable. 
>>
>>Basically, I agree with you.
>
>I think there is some merit in introducing the notion that certain RTP
>packets are indispensible, i.e. 100% reliable. When such packets are lost
>the receiver stops the clock and waits.
>
>This RTP feature should be used only in non-interactive (i.e. streaming)
>applications. It should not be used for video key frames and other such
>error resilient packets.
>
>We can keep saying use TCP for that, why bother enhancing RTP? If I use
>TCP, I have to build a hybrid protocol that uses RTP/UDP and TCP to achieve
>my goal. Why not make it simpler and offer the functionality in RTP?
>
>This issue has been the cause of the growth of proprietary streaming
>protocols by a number of vendors. Most of them have invented their own
>protocol and do not use RTP because of this missing functionality. Even
>some vendors such as Apple who are using RTP are implementing "repeat"
>packets to get around this which is very inefficient.
>
>Let's not preclude a "100% reliability" feature in RTP-Rx.

I think "100% reliable" is a very loaded phrase. I accept that we may need
to allow for streaming media, where some packets have to be delivered with
very high probability, but with some time bound. This is where I believe
RTCP-Rx fits in.

However, I don't believe we want to use RTCP-Rx for streams where the
entire stream is to be delivered 100% reliably. That's not streaming
anymore, and is the case where TCP is used.

I think this means I agree with you - but we should be careful when we say
we want 100% reliable transmission using RTCP-Rx.

Colin



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In-Reply-To: Message from Nikki Cranley <94426082@eeng.dcu.ie> of "Fri, 17 Dec 1999 16:41:53 GMT." <385A67D1.9FBCAE69@eeng.dcu.ie>
Date: Thu, 23 Dec 1999 14:10:30 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Nikki Cranley writes:
>The problem with TCP is that for real-time or near real time transmission its
>not feasible to use TCP as there may be lost packets/delays etc.and
>retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
>used.

Of course - I think we all agree that TCP is not appropriate for streaming. 

>But in the case of MPEG-4 - there are some important packets that probably
>shouldn't be lost - imagine if the packet with all the decoding information got
>lost - well that would render everything else useless. I am looking at this only
>from the perspective of unicast anyway - plus if we were to extend this to VR -
>then RTCP-Rx could be used to dictate user interaction as there is no scope for
>this in RTP/RTCP as far as I know. It would only be needed from client to
>server.

This is where the problem gets interesting - how to transport a mixed media
presentation? MPEG-4 is one example of this - but there are others - where
some parts of the presentation must be delivered reliably (there might be
some HTML and a Java applet, for example) and other parts can be streamed. 
My opinion is that this requires several protocols: it's not appropriate to
try to extend RTP so it can be used for all case.

There's a spectrum of reliability:

	reliable				unreliable
	not-timely				real-time
	+------------------------------------------------+
	TCP                         ~~~~~~~~~~~~~~~~~~ RTP

I envisage RTCP-Rx to be useful in the region labelled '~', for media which
is mostly real-time, but with a need for enhanced reliability subject to
some time bounds. I don't see it as appropriate for fully reliable streams.

Sorry for keeping on about this - but I think it's important to scope the
problem we're trying to solve with RTCP-Rx, before we get into the details
of the protocol. 

Colin



From rem-conf Thu Dec 23 10:49:03 1999 
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From: Thomas Marshall Eubanks <tme@21rst-century.com>
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Organization: Multicast Technologies LLC
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Subject: Re: RTCP Retransmission
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Colin Perkins wrote:

> --> Nikki Cranley writes:
> >The problem with TCP is that for real-time or near real time transmission its
> >not feasible to use TCP as there may be lost packets/delays etc.and
> >retransmission of these will hinder the real-time ability - so RTP/UDP/IP is
> >used.
>
> Of course - I think we all agree that TCP is not appropriate for streaming.
>
> >But in the case of MPEG-4 - there are some important packets that probably
> >shouldn't be lost - imagine if the packet with all the decoding information got
> >lost - well that would render everything else useless. I am looking at this only
> >from the perspective of unicast anyway - plus if we were to extend this to VR -
> >then RTCP-Rx could be used to dictate user interaction as there is no scope for
> >this in RTP/RTCP as far as I know. It would only be needed from client to
> >server.
>
> This is where the problem gets interesting - how to transport a mixed media
> presentation? MPEG-4 is one example of this - but there are others - where
> some parts of the presentation must be delivered reliably (there might be
> some HTML and a Java applet, for example) and other parts can be streamed.
> My opinion is that this requires several protocols: it's not appropriate to
> try to extend RTP so it can be used for all case.
>

Hello;

     Any real time system has to acknowledge the possibility of
drop-outs and data loss. The goal has to be to set specs on what is
acceptable and to engineer things to meet that. If you are saying that certain data
cannot, no matter what, be lost, then you are not talking about real time transport,

because at some times preventing that will require retransmission.  This is
really a function of reliable multicast transport (RMT - rmt@lbl.gov).
Otherwise this is coming close to re-inventing a (not yet fully invented) wheel.

If you are really operating in real-time, and you say that some bits are more
important than others,
then what you are really saying is that some bits need more redundancy than others.
For example, if there was low rate text (say html, which does not gracefully suffer
dropouts) combined with some high rate audio or video stream (which does), the low
rate
text could be sent in some highly redundant fashion, so that it could be
reconstructed
if pieces were lost.

The question I would have is, should this be allowed for at the protocol level,
or should this be viewed as an application level problem that the protocol should
avoid ?
I would argue that the most sensible way of doing this in a protocol would be to
allow
for sub-streams, which could have different levels of redundancy.



Regards

Marshall Eubanks

T.M. Eubanks
Multicast Technologies, LLC
e-mail : tme@21rst-century.com



>
> There's a spectrum of reliability:
>
>         reliable                                unreliable
>         not-timely                              real-time
>         +------------------------------------------------+
>         TCP                         ~~~~~~~~~~~~~~~~~~ RTP
>
> I envisage RTCP-Rx to be useful in the region labelled '~', for media which
> is mostly real-time, but with a need for enhanced reliability subject to
> some time bounds. I don't see it as appropriate for fully reliable streams.
>
> Sorry for keeping on about this - but I think it's important to scope the
> problem we're trying to solve with RTCP-Rx, before we get into the details
> of the protocol.
>
> Colin







From rem-conf Thu Dec 23 12:39:22 1999 
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To: tme@21rst-century.com
cc: 94426082@eeng.dcu.ie, Koichi Yano <yano@cs.berkeley.edu>, rem-conf@es.net
Subject: Re: RTCP Retransmission
In-Reply-To: Message from Thomas Marshall Eubanks <tme@21rst-century.com> of "Thu, 23 Dec 1999 13:46:58 EST." <38626E07.5D84B2E@21rst-century.com>
Date: Thu, 23 Dec 1999 20:21:14 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Thomas Marshall Eubanks writes:
>     Any real time system has to acknowledge the possibility of
>drop-outs and data loss. The goal has to be to set specs on what is
>acceptable and to engineer things to meet that. If you are saying that
>certain data cannot, no matter what, be lost, then you are not talking
>about real time transport, because at some times preventing that will
>require retransmission.  This is really a function of reliable multicast
>transport (RMT - rmt@lbl.gov).  Otherwise this is coming close to
>re-inventing a (not yet fully invented) wheel.
>
>If you are really operating in real-time, and you say that some bits are
>more important than others, then what you are really saying is that some
>bits need more redundancy than others.  For example, if there was low rate
>text (say html, which does not gracefully suffer dropouts) combined with
>some high rate audio or video stream (which does), the low rate text could
>be sent in some highly redundant fashion, so that it could be
>reconstructed if pieces were lost.
>
>The question I would have is, should this be allowed for at the protocol
>level, or should this be viewed as an application level problem that the
>protocol should avoid ?  I would argue that the most sensible way of doing
>this in a protocol would be to allow for sub-streams, which could have
>different levels of redundancy.

We have traditionally sent each media type as a separate RTP stream, so it
is simple to provide different levels of protection for each. This is one
of the main reasons why multiplexing media streams together in RTP is not
always a good idea.

Colin



From rem-conf Thu Dec 23 17:53:49 1999 
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Date: Thu, 23 Dec 1999 17:57:11 -0800
To: "M. Reha Civanlar" <civanlar@research.att.com>,<rem-conf@es.net>
From: alagu@entera.com (Alagu Periyannan)
Subject: Re: RTCP Retransmission
In-Reply-To: <016201bf48df$d7626540$4683cf87@research.att.com>
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 <3.0.6.32.19991217142818.01530650@mailserver.entera.com>
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At 05:41 PM 12/17/99 -0500, M. Reha Civanlar wrote:
>From: Alagu Periyannan <alagu@entera.com>
>> We can keep saying use TCP for that, why bother enhancing RTP? If I use
>> TCP, I have to build a hybrid protocol that uses RTP/UDP and TCP to achieve
>> my goal. Why not make it simpler and offer the functionality in RTP?
>
>The problem is that this won't make RTP simpler.

Correct. But if we take the pain and introduce this functionality in RTP-Rx
it will provide a standards-based way to deliver streams in which a few of
the packets require 100% reliability. Currently the only way to do this is
inventing my own protocol.

>
>> 
>> This issue has been the cause of the growth of proprietary streaming
>> protocols by a number of vendors. Most of them have invented their own
>> protocol and do not use RTP because of this missing functionality. Even
>> some vendors such as Apple who are using RTP are implementing "repeat"
>> packets to get around this which is very inefficient.
>
>How can they achieve 100% reliability with "repeat" packets anyway?
>

That's right you can't achieve 100% reliability with this method. I was
trying to illustrate how vendors are trying solve the reliability issue
within the current RTP framework.


--

Alagu Periyannan                     alagu@entera.com
Entera, Inc.                         +1 510 770 5225




From rem-conf Thu Dec 23 18:13:07 1999 
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Date: Thu, 23 Dec 1999 18:17:49 -0800
To: Colin Perkins <c.perkins@cs.ucl.ac.uk>
From: alagu@entera.com (Alagu Periyannan)
Subject: Re: RTCP Retransmission
Cc: rem-conf@es.net,yano@cs.berkeley.edu
In-Reply-To: <199912231353.NAA12171@cperkins-d.cs.ucl.ac.uk>
References: <Message from alagu@entera.com (Alagu Periyannan) of "Fri, 17 Dec 1999 14:28:18 PST." <3.0.6.32.19991217142818.01530650@mailserver.entera.com>
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At 01:53 PM 12/23/99 +0000, Colin Perkins wrote:
>--> Alagu Periyannan writes:
>>Let's not preclude a "100% reliability" feature in RTP-Rx.
>
>I think "100% reliable" is a very loaded phrase. I accept that we may need
>to allow for streaming media, where some packets have to be delivered with
>very high probability, but with some time bound. This is where I believe
>RTCP-Rx fits in.
>
>However, I don't believe we want to use RTCP-Rx for streams where the
>entire stream is to be delivered 100% reliably. That's not streaming
>anymore, and is the case where TCP is used.
>
>I think this means I agree with you - but we should be careful when we say
>we want 100% reliable transmission using RTCP-Rx.
>

Actually the fundamental debate here is whether RTP is solely a "realtime"
protocol or is it a "time-based streaming" protocol.

A "realtime" protocol in my definition is a protocol that delivers
timestamped packets in a time bounded fashion. Since it needs to deliver
packets in a time bounded fashion it can theoretically never achieve 100%
reliability.

A "time-based streaming" protocol in my definition is a protocol that
delivers timestamped packets period. It must cover the case of time bounded
delivery with no reliability guarantees and the case of 100% reliability
with no time bound.

A time-based streaming protocol's functionality is a superset of a realtime
protocol's functionality. I would like to see an IETF standard for a
time-based streaming protocol. The question is do we build that off RTP
(and its popularity and success) or do we just invent yet another RTXYZP
protocol?


--

Alagu Periyannan                     alagu@entera.com
Entera, Inc.                         +1 510 770 5225




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From rem-conf Mon Dec 27 06:11:54 1999 
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To: alagu@entera.com (Alagu Periyannan)
cc: rem-conf@es.net, yano@cs.berkeley.edu
Subject: Re: RTCP Retransmission
In-Reply-To: Message from alagu@entera.com (Alagu Periyannan) of "Thu, 23 Dec 1999 18:17:49 PST." <3.0.6.32.19991223181749.00c81590@mailserver.entera.com>
Date: Mon, 27 Dec 1999 13:56:36 +0000
From: Colin Perkins <c.perkins@cs.ucl.ac.uk>
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--> Alagu Periyannan writes:
>Actually the fundamental debate here is whether RTP is solely a "realtime"
>protocol or is it a "time-based streaming" protocol.
>
>A "realtime" protocol in my definition is a protocol that delivers
>timestamped packets in a time bounded fashion. Since it needs to deliver
>packets in a time bounded fashion it can theoretically never achieve 100%
>reliability.
>
>A "time-based streaming" protocol in my definition is a protocol that
>delivers timestamped packets period. It must cover the case of time bounded
>delivery with no reliability guarantees and the case of 100% reliability
>with no time bound.

We have RTP over UDP for the first case, and the new profile draft specifies a 
TCP encapsulation for RTP which should deal with the case of "100% reliability
with no time bound"? 

This leaves the case where we have a time bound, but desire enhanced reliability 
compared to UDP. We have some options here already (eg: redundant audio and
parity FEC), and the RTCP retransmission draft offers another alternative.

Colin



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Subject: Re: Adv: Find things out about anybody - Right now
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From rem-conf Tue Dec 28 11:32:50 1999 
From rem-conf-request@es.net Tue Dec 28 11:32:49 1999
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Date: Tue, 28 Dec 1999 11:31:31 -0800
To: Colin Perkins <c.perkins@cs.ucl.ac.uk>
From: alagu@entera.com (Alagu Periyannan)
Subject: Re: RTCP Retransmission
Cc: rem-conf@es.net,yano@cs.berkeley.edu
In-Reply-To: <199912271356.NAA05568@cperkins-d.cs.ucl.ac.uk>
References: <Message from alagu@entera.com (Alagu Periyannan) of "Thu, 23 Dec 1999 18:17:49 PST." <3.0.6.32.19991223181749.00c81590@mailserver.entera.com>
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At 01:56 PM 12/27/99 +0000, Colin Perkins wrote:
>--> Alagu Periyannan writes:
>>
>>A "time-based streaming" protocol in my definition is a protocol that
>>delivers timestamped packets period. It must cover the case of time bounded
>>delivery with no reliability guarantees and the case of 100% reliability
>>with no time bound.
>
>We have RTP over UDP for the first case, and the new profile draft
specifies a 
>TCP encapsulation for RTP which should deal with the case of "100%
reliability
>with no time bound"? 
>
>This leaves the case where we have a time bound, but desire enhanced
reliability 
>compared to UDP. We have some options here already (eg: redundant audio and
>parity FEC), and the RTCP retransmission draft offers another alternative.
>

I was not aware of the RTP over TCP draft. (A pointer is appreciated.)

Nevertheless, in many cases one may need to mix the various cases I
mentioned in one RTP session. For example if we are streaming an animation
over RTP, we may want to send the "actor" information in a reliable fashion
and the motion vectors in a non-reliable fashion.

This is not possible if the "100% reliability with no time bound" case is
covered in a separate profile that does not include features of the
standard A/V profile.


--

Alagu Periyannan                     alagu@entera.com
Entera, Inc.                         +1 510 770 5225




From rem-conf Wed Dec 29 16:45:47 1999 
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Date: Wed, 29 Dec 1999 16:30:34 -0800
From: "Lawrence A. Rowe" <Rowe@bmrc.berkeley.edu>
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Organization: U.C. Berkeley
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To: rem-conf <rem-conf@es.net>, bmrc-list@gumby.CS.Berkeley.EDU,
        298-list@gumby.CS.Berkeley.EDU, mm-group@gumby.CS.Berkeley.EDU,
        eecs-instructors@eecs.Berkeley.EDU
Subject: Distance Learning Experiment in Spring 2000
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Hi -

UC Berkeley now has in place the technology required to allow us 
to accept remote students into select classes.  We want to do an 
experiment during the spring 2000 semester in which we admit a limited 
number of remote students into our classes using concurrent 
enrollment through University Extension and Real Networks webcasting 
technology for distance/asynchronous learning.

The purpose of this message is to solicit your participation in this 
experiment by enrolling in one of the following courses:

   EECS 152 "Computer Architecture and Engineering" (5 units)
     Instructor: Professor Robert Broderson
     Meeting time: M/W/F 9:00-10:00 AM
   EECS 240 "Analog Integrated Circuit Design and Analysis" (3 units)
     Instructor: Professor Bernhard E. Boser
     Meeting time: T/Th 11:00-12:30 PM

EECS 152 is an undergraduate class and EECS 240 is a graduate class.
Detailed descriptions of these courses are given below.  

Qualified students will be admitted through concurrent enrollment 
at UC Berkeley Extension.  Lectures will be webcast live and 
stored for on-demand replay using Real Networks technology.  See the 
following web pages for examples:

   http://media2.bmrc.berkeley.edu/bibs/schedule.cfm
  
http://media2.bmrc.berkeley.edu/projects/alps/classes/cs298-5/19990825/

Although this technology operates at lower bit-rates, it is 
strongly suggested that you have access to a broadband Internet 
connection with speeds higher than 250 Kbs.  As a student, you 
can view the live broadcast or watch lectures at a later time 
on-demand.  You will be expected to do all the assignments and 
take the same examinations as students enrolled on campus.  You 
will have a discussion section and TA assigned.  Course assignments 
will be distributed and turned in through the web.  You will 
have contact with the instructors and other students using email, 
a class newsgroup, and using traditional phone conferencing.  We 
will also be experimenting with a remote question link during the
lectures 
and a chat tool which will allow real time questions and answers.  
Special arrangements will be made for taking the examinations at 
or near your work location.

A limited number of students will be admitted to these classes.
Individuals interested in taking this courses and participating 
in this experiment should fill-in and submit the application form 
given below.  Students selected for the experiment will enroll 
through concurrent enrollment, which will cost $4,000 for EECS 152 
and $2,400 for EECS 240 (i.e., $800/unit), and will be given 
the appropriate access information required to take the course. 

If you have further questions please contact us.
        Larry Rowe
---
Professor Lawrence A. Rowe          Internet:  Rowe@BMRC.Berkeley.EDU
Computer Science Division - EECS       Phone: 510-642-5117
University of California, Berkeley       Fax: 510-642-5615
Berkeley, CA 94720-1776            URL: http://bmrc.berkeley.edu/~larry

== Course Descriptions =======================

EECS 152 "Computer Architecture and Engineering" (5 Units)
Instructor: Professor Robert Broderson

Course Format: Three hours of lecture and two hours of 
discussion per week and one large design project.

Prerequisites: EECS 150 or equivalent.

Description: Instruction set design, Register Transfer. 
Computer design project requiring about 100 hours. 
Data-path design.  Controller design. Memory system. 
Addressing. Microprogramming. Computer arithmetic. 
Survey of real computers and microprocessors.

----------------------------------------------

EECS 240 "Analog Integrated Circuit Design and Analysis" (3 units)
Instructor: Professor Bernhard E. Boser

Course Format: Three hours of lecture per week.

Prerequisites: EECS 140 or equivalent.

Description: Analysis and optimized design of monolithic 
operational amplifiers and wide-band amplifiers; methods of 
achieving wide-band amplification, gain-bandwidth considerations; 
analysis of noise in integrated circuits and low noise design.  
Precision passive elements, analog switches, amplifiers and 
comparators, voltage reference in NMOS and CMOS circuits. 
Serial successive-approximation, and parallel analog-to-digital 
converters. Switched-capacitor and CCD filters.  Applications to 
codecs and modems.

==============================================

== Application Form ==========================

Name:
Email Address:
Postal Address:
Educational Background:
Course Enrolling In:  EECS 152  or  EECS 240
Desktop Computer and OS:
Internet Connection/Speed:

Why do you want to take this course?

Please include a copy of your resume.

Send form to Rowe@CS.Berkeley.EDU
===============================================



From rem-conf Thu Dec 30 16:57:05 1999 
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From: Jori <jori@lumumba.luc.ac.be>
To: rem-conf@es.net
Subject: RTP library
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Hi there !

Just to say that I've written a RTP library. If anyone is interested, you
can download it from http://lumumba.luc.ac.be/jori/rtp/rtp.html
The library is written in C++ and should work on pretty much any unix-like
platform and a ms-windows platform.

Bye,
Jori





