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From: Zeev Vax <zeevvax@microsoft.com>
To: VAN CAENEGEM Tom <Tom.Van_Caenegem@alcatel-lucent.be>, Peilin YANG <yangpeilin@huawei.com>, "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Ali C. Begen (abegen)" <abegen@cisco.com>
Date: Tue, 31 Mar 2009 10:05:42 -0700
Thread-Topic: Revised text of the proposal for describing the functionality of Rate Adaptation
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Cc: Tom Taylor <tom.taylor@rogers.com>, "avt@ietf.org" <avt@ietf.org>
Subject: Re: [AVT] Revised text of the proposal for describing the functionality of Rate Adaptation
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<div class=3DSection1>

<p class=3DMsoPlainText><span style=3D'color:#984806'>Please see inline.<o:=
p></o:p></span></p>

<p class=3DMsoPlainText><span style=3D'color:#984806'><o:p>&nbsp;</o:p></sp=
an></p>

<p class=3DMsoPlainText><span style=3D'color:#984806'>Zeev<o:p></o:p></span=
></p>

<p class=3DMsoPlainText><a name=3D"_MailEndCompose"><o:p>&nbsp;</o:p></a></=
p>

<p class=3DMsoPlainText>-----Original Message-----<br>
From: VAN CAENEGEM Tom [mailto:Tom.Van_Caenegem@alcatel-lucent.be] <br>
Sent: Monday, March 30, 2009 7:28 AM<br>
To: Peilin YANG; Bill Ver Steeg (versteb); Ali C. Begen (abegen); Zeev Vax<=
br>
Cc: Roni Even; Tom Taylor; Ye-Kui Wang; avt@ietf.org<br>
Subject: RE: Revised text of the proposal for describing the functionality =
of
Rate Adaptation</p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Hi Peilin,<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Thanks for your comments. For those that concern
editorial stuff, I have no comments back. <o:p></o:p></p>

<p class=3DMsoPlainText>I have comments on three of your suggested modifica=
tion
items, as explained below.<o:p></o:p></p>

<p class=3DMsoPlainText>Kind regards<o:p></o:p></p>

<p class=3DMsoPlainText>Tom <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Item 1) RMS-I message<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>You wrote: <o:p></o:p></p>

<p class=3DMsoPlainText>&quot;In the RMS-I message, the present values of
Response include 0,1 and 2. The value of 0 indicates that the rapid synchro=
nization
request has been rejected. Nevertheless, in the RMS-I message there is not =
a
field to indicate a rejected reason. In my point of view, tt is easier for =
RR
to understand the rejected reason if there is a field of reason. What's mor=
e,
it is convenient to make statistics report. <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The following Figure is my proposal.&nbsp; (..) &qu=
ot;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>TOM : in any case the message formats will be revis=
ed to
describe parameters as much as possible and/or useful by means of(optional)=
 TLV
fields. I believe Bill will make a proposal soon.<o:p></o:p></p>

<p class=3DMsoPlainText>I think the &quot;cause of rejection&quot; may be s=
uch a
TLV field, when the RS indeed rejects the Burst Request. However, I think t=
hat
may well be a vendor-specific field (for which we need a &quot;type&quot;
classifier), as there are many reasons why a request is not honored, but to
list them would IMO be out of scope of this draft. So, the question is: do =
we
define a Type for a TLV extension that we call e.g. &quot;reason of
rejection&quot; or do we just stick to a Type that only means &quot; vendor
specific&quot;. I think in both cases, an RR and a RS of a different vendor
will not understand each-other, so we may well make it very generic and onl=
y
define a TLV &quot;type&quot; that means &quot; vendor specific&quot;<o:p><=
/o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><b><i><span style=3D'color:#984806'>[Zeev Vax] </sp=
an></i></b><span
style=3D'color:#984806'>We still need to discuss the exact balance between =
TLV
and fixed fields. <o:p></o:p></span></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Item 2) Section 6.2: Message flows<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp; 3.&nbsp; Updated Request: The RR MAY s=
end a
new RMS-R message with a different<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; valule of the =
field
Max Receive Bitrate. The RS MAY use this updated<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; bitrate for se=
nding
the burst, or refine the value and use the refined <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; value for send=
ing
the burst. In any of the cases, the RS MAY send an <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RMS-I to indic=
ate
the bitrate being used. In case the RS receives <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; multiple RMS-R
messages at the same time, the latest RMS-R message,<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; as indicated b=
y the
value of the MSN field in the RMS-R message, MUST <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; be used, and o=
ther
RMS-R messages SHOULD be discarded.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>TOM: I suggest to delete &quot; In case the RS rece=
ives <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; multiple RMS-R
messages at the same time, the latest RMS-R message,<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; as indicated b=
y the
value of the MSN field in the RMS-R message, MUST <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; be used, and o=
ther
RMS-R messages SHOULD be discarded.&quot; <o:p></o:p></p>

<p class=3DMsoPlainText><b><i><span style=3D'color:#984806'>[Zeev Vax] I ag=
ree with
Tom, and his comment is align with what we agreed in the last meeting. Any =
following
RMS-R message might be used by the RS, but it is only recommendation. </spa=
n></i></b><span
style=3D'color:#984806'><o:p></o:p></span></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>This is because an RS may not support the feature o=
f
flexible burst rate adaptation under control of the RR. Hence in that case =
its
implementation may be such that it classifies incoming RMS-R based on versi=
on
number and internal state. When the RS&nbsp; has already an active thread f=
or a
certain RR and certain SSM (a pending or active burst request) , the RS can
simply drop any new RMS-R (with a higher message sequence number) transmitt=
ed
by that specific RR for that specific ssm.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Item 3) Section on RMS-R message (section 7.1)<o:p>=
</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Message Sequence Num=
ber (8
bits) :&nbsp; During rapid synchronization,<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the RMS-R message(s)=
 may
be sent more than once.&nbsp; The first RMS-R<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; message SHALL have a=
n MSN
value of 0. If a new information is conveyed<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in a new RMS-R messa=
ge,
the MSN value SHALL be incremented by one.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>TOM: an RMS-R message may also be sent multiple tim=
es
with the same information for redundancy purposes. For that reason, I sugge=
st
we have instead of &quot; If a new information is conveyed in a new RMS-R
message, the MSN value SHALL be incremented by one &quot;, we are more spec=
ific
and align with the phrasing as used for the RMS-I message:<o:p></o:p></p>

<p class=3DMsoPlainText>&quot;This value SHALL NOT be changed if the same R=
MS-R
message is sent to the same RS multiple times for redundancy purposes. If a=
 new
information is conveyed in a new RMS-R message, the MSN value SHALL be
incremented by one&quot;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><span style=3D'color:#984806'>[Zeev Vax] Both sound=
 reasonable
to me<o:p></o:p></span></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>-----Original Message-----<o:p></o:p></p>

<p class=3DMsoPlainText>From: Peilin YANG [mailto:yangpeilin@huawei.com] <o=
:p></o:p></p>

<p class=3DMsoPlainText>Sent: maandag 30 maart 2009 13:20<o:p></o:p></p>

<p class=3DMsoPlainText>To: Bill Ver Steeg (versteb); Ali C. Begen (abegen)=
; VAN
CAENEGEM Tom; zeevvax@microsoft.com<o:p></o:p></p>

<p class=3DMsoPlainText>Cc: Roni Even; Tom Taylor; Ye-Kui Wang<o:p></o:p></=
p>

<p class=3DMsoPlainText>Subject: Revised text of the proposal for describin=
g the
functionality of Rate Adaptation<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Dear Bill, Ali, Tom and Zeev,<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>I added some revised text of the proposal for descr=
ibing
the functionality of Rate Adaptation for the Burst. Please see the attached=
 file.
The revised text mainly involves in a new step 3, part of step 1, &quot;7.1=
.
RMS Request&quot; and Figure 3 depicts an example of messaging flow.<o:p></=
o:p></p>

<p class=3DMsoPlainText>I am very glad to disucss it with you. In the attac=
hment
file it also includes a little revision. Please see if not suitable for
revison.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>In addition, I think, the following suggestions als=
o need
to be discussed.&nbsp; <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>1) IGMP <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>In the draft like the following text it only descri=
bes
the IGMP based on IPv4.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&quot;When a receiver wishes to join the same multi=
cast
session, instead of simply issuing&nbsp;&nbsp; an Internet Group Management
Protocol (IGMP) [RFC3376] Join message, it sends a request to the feedback
target address for the session asking for the Reference Information. &quot;=
<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Do you think it is necessary to consider MLD protoc=
ol based
on IPv6 in the document as follows? <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&quot;When a receiver wishes to join the same multi=
cast
session, instead of simply issuing&nbsp;&nbsp; an Internet Group Management
Protocol (IGMP) [RFC3376] or Multicast Listener Discovery (MLD) [RFC 3810] =
Join
message, it sends a request to the feedback target address for the session
asking for the Reference Information. &quot;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>2) RMS-I message<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>In the RMS-I message, the present values of Respons=
e
include 0,1 and 2. The value of 0 indicates that the rapid synchronization
request has been rejected. Nevertheless, in the RMS-I message there is not =
a
field to indicate a rejected reason. In my point of view, tt is easier for =
RR
to understand the rejected reason if there is a field of reason. What's mor=
e,
it is convenient to make statistics report. <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The following Figure is my proposal.&nbsp; <o:p></o=
:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
3<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 =
0 1 2
3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;
MSN&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; Response&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Reason&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp=
;
Rsvd.&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Extended RTP Seqnum of the Firs=
t
Burst Packet&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |<o:p></=
o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Earliest IGMP Join
Time&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;Rapid
Synchronization
Duration&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Max Burst
Bitrate&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
|<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<o:p></o:p=
></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Figure 5=
: FCI
field syntax for the RMS Information message<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>3) &quot;an/the RTP Receiver&quot; and &quot;a/the
Receiver&quot;<o:p></o:p></p>

<p class=3DMsoPlainText>In the document there are many &quot;an/the RTP
Receiver&quot; and &quot;a/the Receiver&quot;. Do you think it is necessary=
 to
keep these terms to be consistent throughout the document?&nbsp; I revised =
them
as &quot;an/the RTP Receiver&quot; in the revised proposal text. <o:p></o:p=
></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>4) &quot;the burst&quot;, &quot;a/the unicast
burst&quot;, &quot;an/the RTP burst&quot; and &quot;the RTP unicast burst&q=
uot;<o:p></o:p></p>

<p class=3DMsoPlainText>In the document there are many &quot;the burst&quot=
;,
&quot;a/the unicast burst&quot;, &quot;an/the RTP burst&quot; and &quot;the=
 RTP
unicast burst&quot;. Do you think it is necessary to keep these terms to be
consistent throughout the document?&nbsp; I revised them as &quot;a/the uni=
cast
burst&quot; in the revised proposal text.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>5) &quot;joining the multicast session&quot; and
&quot;joining the multicast group&quot;<o:p></o:p></p>

<p class=3DMsoPlainText>In the document there is only one &quot;joining the
multicast group&quot;. I modified &quot;joining the multicast group&quot; t=
o
&quot;joining the multicast session&quot;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>6) The description of &quot;Updated Responses&quot;=
 of
step 3 In the document there is a general description for &quot;Updated
Responses&quot; as follows. When I discussed with our development enginerrs=
, as
on-native english speakers they said it is not completely clear to how to
implement it.<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp; &quot;3.&nbsp; Updated Responses:&nbsp; RS m=
ay
send more than one RMS-I messages,<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; e.g., to updat=
e the
burst bitrate information when the bitrate is<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; adapted and/or=
 to
signal RR in real time to join the SSM session.&quot;<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>They said the original text from the draft -
draft-levin-avt-rtcp-burst-00 is easier to understand.<o:p></o:p></p>

<p class=3DMsoPlainText>So I suggest to add a little description example at=
 the
end of the sensence.<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>The following text is my proposal. The last sensenc=
e is
largely extracted from the draft - draft-levin-avt-rtcp-burst-00&nbsp; <o:p=
></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&nbsp; &quot;3.&nbsp; Updated Responses:&nbsp; RS m=
ay
send more than one RMS-I messages,<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; e.g., to updat=
e the
unicast burst bitrate information when the bitrate is<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; adapted and/or=
 to
signal RR in real time to join the SSM session; to notify <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RR to join the
multicast session when the unicast burst catches up with <o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the original
multicast session.&quot; <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>Regards,<o:p></o:p></p>

<p class=3DMsoPlainText>Peilin<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>***************************************************=
***************************************<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;This email and its attachments contain confid=
ential
information from HUAWEI, which is intended only for the person or entity wh=
ose
address is listed above. Any use of the information contained here in any w=
ay
(including, but not limited to, total or partial disclosure, reproduction, =
or
dissemination) by persons other than the intended recipient(s) is prohibite=
d.
If you receive this email in error, please notify the sender by phone or
email&nbsp; immediately and delete it!<o:p></o:p></p>

<p class=3DMsoPlainText>&nbsp;*********************************************=
********************************************<o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>----- <span style=3D'font-family:"MS Mincho"'>=D4=
=AD</span><span
style=3D'font-family:"Calibri","sans-serif"'>=D3=CA=BC=FE</span> -----<o:p>=
</o:p></p>

<p class=3DMsoPlainText><span style=3D'font-family:"Calibri","sans-serif"'>=
=B7=A2=BC=FE=C8=CB</span>:
&quot;Bill Ver Steeg (versteb)&quot; &lt;versteb@cisco.com&gt;<o:p></o:p></=
p>

<p class=3DMsoPlainText><span style=3D'font-family:"MS Mincho"'>=C8=D5=C6=
=DA</span>: <span
style=3D'font-family:"MS Mincho"'>=D0=C7=C6=DA=CE=E5</span>, <span style=3D=
'font-family:"MS Mincho"'>=C8=FD=D4=C2</span>
27<span style=3D'font-family:"MS Mincho"'>=C8=D5</span>, 2009 <span style=
=3D'font-family:
"MS Mincho"'>=C9=CF=CE=E7</span>3:43<o:p></o:p></p>

<p class=3DMsoPlainText><span style=3D'font-family:"MS Mincho"'>=D6=F7</spa=
n><span
style=3D'font-family:"Calibri","sans-serif"'>=CC=E2</span>: RE: Hi Bill and=
 all,<o:p></o:p></p>

<p class=3DMsoPlainText><span style=3D'font-family:"MS Mincho"'>=CA=D5=BC=
=FE=C8=CB</span>: Peilin
YANG &lt;yangpeilin@huawei.com&gt;, &quot;Ali C. Begen (abegen)&quot;
&lt;abegen@cisco.com&gt;, Tom.Van_Caenegem@alcatel-lucent.be,
zeevvax@microsoft.com, Ye-Kui Wang &lt;yekuiwang@huawei.com&gt;<o:p></o:p><=
/p>

<p class=3DMsoPlainText><span style=3D'font-family:"MS Mincho"'>=B3=AD=CB=
=CD</span>: Roni
Even &lt;ron.even.tlv@gmail.com&gt;, Tom Taylor &lt;tom.taylor@rogers.com&g=
t;,
&quot;Dave Oran (oran)&quot; &lt;oran@cisco.com&gt;, avt@ietf.org<o:p></o:p=
></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

<p class=3DMsoPlainText>&gt; Peilin-<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; I think we made good progress on the draft. We=
 also
have some other <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; folks looking at how the rate adaptation inter=
acts
with the rest of <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; the signaling. Hopefully we will be able to cl=
eanly
incorporate rate <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; adaptation into the protocol. If you have some
suggested text in that <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; area, that would be great.<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Bill VerSteeg<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; -----Original Message-----<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; From: Peilin YANG [mailto:yangpeilin@huawei.co=
m]<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Sent: Wednesday, March 25, 2009 8:07 PM<o:p></=
o:p></p>

<p class=3DMsoPlainText>&gt; To: Bill Ver Steeg (versteb); Ali C. Begen (ab=
egen);
<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Tom.Van_Caenegem@alcatel-lucent.be;
zeevvax@microsoft.com; Ye-Kui Wang<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Cc: Roni Even; Tom Taylor; Dave Oran (oran);
avt@ietf.org<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Subject: Hi Bill and all,<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Hi Bill and all,<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Thank you very much for concerning Huawei's dr=
aft.<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; I am sorry I haven't participated the SF meeti=
ng
because I haven't got <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; an American Visa this time.<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; YK told me an informal result of break session
yesterday.<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Now I am considering the revised text for desc=
ribing
the functionality <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; of Rate Adaptation for the Burst to the common
converging solution and <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; other possible contributing text.<o:p></o:p></=
p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Regards,<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; Peilin<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt;
**********************************************************************<o:p>=
</o:p></p>

<p class=3DMsoPlainText>&gt; **<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; ******************<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; This email and its attachments contain confide=
ntial
information from <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; HUAWEI, which is intended only for the person =
or
entity whose address <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; is listed above. Any use of the information
contained here in any way <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; (including, but not limited to, total or parti=
al
disclosure, <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; reproduction, or dissemination) by persons oth=
er
than the intended<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; recipient(s) is prohibited. If you receive thi=
s
email in error, please <o:p></o:p></p>

<p class=3DMsoPlainText>&gt; notify the sender by phone or email&nbsp;
immediately and delete it!<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText>&gt;
**********************************************************************<o:p>=
</o:p></p>

<p class=3DMsoPlainText>&gt; **<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; *****************<o:p></o:p></p>

<p class=3DMsoPlainText>&gt; <o:p></o:p></p>

<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

--_000_5177B2A348E4B64EAD7D589A027892D96A29EAAC65NAEXMSGC111re_--

From yangpeilin@huawei.com  Wed Apr  1 06:34:12 2009
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Date: Wed, 01 Apr 2009 21:35:08 +0800
From: Peilin YANG <yangpeilin@huawei.com>
In-reply-to: <81B9B88E2F9D574AA5328A85547CDE91EC162E@xmb-rtp-21d.amer.cisco.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, zeevvax@microsoft.com, Tom.Van_Caenegem@alcatel-lucent.be, "Ali C. Begen (abegen)" <abegen@cisco.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fields
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Please see inline comments=2C

*************************************************************************=
*****************
 This email and its attachments contain confidential information from HUA=
WEI=2C which is intended only for the person or entity whose address is l=
isted above=2E Any use of the information contained here in any way (incl=
uding=2C but not limited to=2C total or partial disclosure=2C reproductio=
n=2C or dissemination) by persons other than the intended recipient(s) is=
 prohibited=2E If you receive this email in error=2C please notify the se=
nder by phone or email
 immediately and delete it!
 ************************************************************************=
*****************

----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40cisco=2E=
com=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=C8=FD=2C =CB=C4=D4=C2 1=C8=D5=2C 2009 =C9=CF=
=CE=E72=3A46
=D6=F7=CC=E2=3A =5BAVT=5D Rapid Acquisition of Multicast Streams (RAMS) -=
 Fixed and TLV fields
=CA=D5=BC=FE=C8=CB=3A avt=40ietf=2Eorg=2C zeevvax=40microsoft=2Ecom=2C To=
m=2EVan=5FCaenegem=40alcatel-lucent=2Ebe=2C =22Ali C=2E Begen (abegen)=22=
 =3Cabegen=40cisco=2Ecom=3E

=3E =

=3E One of the primary action items arising from the review of the Rapid
=3E Acquisition of Multicast Streams  draft is to identify which =

=3E fields are
=3E mandatory (and thus fixed format) and which fields are optional (and
=3E thus TLV format)=2E Here is my take on the matter=2E Once we reach
=3E consensus=2C I will update the draft=2E
=3E =

=3E I have also included a description of the requested =22motivating tex=
t=22
=3E for the fields that do not already have such text in the document=2E
=3E Please let me know if I am on the right track=2E
=3E =

=3E There is currently no version number in the messages=2E Ali and I do =
not
=3E think that this is a problem because we can add new TLVs as =

=3E needed=2E I
=3E want to be sure we intentionally decide on this point rather than bac=
k
=3E into it accidently=2E Any comments=3F
=3E =

=3E We have 3 messages - =

=3E =

=3E RAMS-Request
=3E Max Receive BitRate - mandatory
=3E Min Buffer Fill Requirement - optional ( Add wording - The =

=3E receiver may
=3E have knowledge of it=27s buffering requirements=2E These requirements=
 =

=3E may be
=3E application or device specific=2E For instance=2C the receiver may =

=3E need to
=3E have a certain amount of data in it=27s application buffer to handle
=3E interdependency within the data=2E If the RS knows the buffering abil=
ity
=3E of the receiver=2C it may more precisely tailor the burst to the clie=
nt=2E
=3E The methods used by the receiver to determine this value is =

=3E applicationspecific=2C and thus out of scope=2E This mechanism can be=
 =

=3E used by the
=3E receiver to signal the sender=2C if the receiver has such knowledge )=

=3E Max Buffer Fill Requirement - optional ( Add wording - The =

=3E receiver may
=3E have knowledge of it=27s buffering requirements=2E These requirements=
 =

=3E may be
=3E application or device specific=2E For instance=2C one receiver may =

=3E have more
=3E physical memory than another receiver=2C and thus can buffer more =

=3E data=2E If
=3E the RS knows the buffering ability of the receiver=2C it may more
=3E precisely tailor the burst to the client=2E The methods used by the
=3E receiver to determine this value is application specific=2C and thus =
out
=3E of scope=2E This mechanism can be used by the receiver to signal the
=3E sender=2C if the receiver has such knowledge )
=3E =

Peilin=3A Like Mr=2E Tom Taylor comment=2C these two optional parameters =
are quite clear to me=2E It is easier to understand and implement if ther=
e is an instance to explain it=2E =


=3E RAMS- Inform
=3E message Sequence Number - mandatory
=3E Response - mandatory=2C will include an enumerated list of response =

=3E codes- 0 is succeed=2C non zero is an error code=2E We will enumerate=
 =

=3E a list of
=3E error codes=2C and note that additional error codes are possible usin=
g
=3E vendor-specific TLV fields=2E The enumerated errors will include =5BN=
o
=3E server resources=2C No network resources=2C bad RMS-R request=2C WHAT=

=3E ELSE=3F=3F=3F=3F=5D =


Peilin=3A no buffered content=2E For example=2C when RR requests a channe=
l=2CRS hasn=27t buffered this channel=2CRS will reject this request=2E Th=
at is to say=2C RR is the first receiver to request to this channel becau=
se it is impossible for RS to buffer all channels=2E so I think =22no buf=
fered content=22 is a kind of error code=2E

Note - we have removed the =22join now semantic=2C as that is
=3E captured in the =22earliest join time=22 field=2E =

=3E Reserved - mandatory - must be all 0s
=3E Max Burst Bitrate - mandatory =

=3E Rapid Acquisition Duration - mandatory - 0 means =22join now=22 (Add
=3E wording to reflect that this is the server=27s current value=2C and =

=3E can be
=3E changed by subsequent RMS-I=2E Add wording to reflect that the RS can=
=27t
=3E consume resources forever and needs to deterministically stop the =

=3E flow=2EThe RS MUST terminate the flow at this time=2C even if does no=
t =

=3E get RMS-T
=3E from the RR=2E  ) =

=3E Extended RTP Sequence Number of First Burst Packet - mandatory
=3E Earliest IGMP Join Time - optional (we should consider making this
=3E mandatory in at least one RMS-I=2C as the client needs to eventually =

=3E join) =

=3E Rams -Terminate - =

=3E  Extended RTP seqnum of First Multicast packet - optional (intent =

=3E is to
=3E tell the RS the seqnum at which to stop the burst=2C but leave the =

=3E sessionopen=2E If the field is not present=2C it means =22stop now=22=
)
=3E =

=3E =

=3E =

=3E Bill VerSteeg
=3E 

From yangpeilin@huawei.com  Wed Apr  1 06:38:15 2009
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Date: Wed, 01 Apr 2009 21:38:40 +0800
From: Peilin YANG <yangpeilin@huawei.com>
In-reply-to: <81B9B88E2F9D574AA5328A85547CDE91EC1548@xmb-rtp-21d.amer.cisco.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Ali C. Begen (abegen)" <abegen@cisco.com>, Tom.Van_Caenegem@alcatel-lucent.be, zeevvax@microsoft.com
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Cc: avt@ietf.org
Subject: Re: [AVT] Revised text of the proposal for describing the functionality of Rate Adaptation
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Hi Bill=2C Ali=2C Tom and Zeev=2C

Thank you very much for reply and comments=2C and thank Bill for going th=
rough the document with my suggestion=2E
I also support Ali=27s advice that we should focus on issue at this time=2E=
Once we slove the major issues=2C we can make quick progress on the conve=
rging draft=2E

The most important issue is the proposal of =22fixed vs TLV fields=22 tha=
t Bill suggested to discuss in the mailing list=2E
Obviously both fixed vs TLV fields have their advantage and disadvantage=2E=
 For TLV fields=2C it is easier to allow vendor-specific extensions=2E Bu=
t I have no idea if TLV format will influence the maximum number of concu=
rrent of RS=2E It should be better if there is a small influence=2E

Another issue is the Sequence Number of RMS-R=2Cas Bill said=2Cit is not =
good that two message of the 3 have a seqnum type=2E I also think so=2E N=
evertheless=2C for RS perhaps it is more clear if there is a seqnum of RM=
S-R=2E I have no idea if there is another replaceable way for this=2E

Regards=2C

Peilin
*************************************************************************=
*****************
 This email and its attachments contain confidential information from HUA=
WEI=2C which is intended only for the person or entity whose address is l=
isted above=2E Any use of the information contained here in any way (incl=
uding=2C but not limited to=2C total or partial disclosure=2C reproductio=
n=2C or dissemination) by persons other than the intended recipient(s) is=
 prohibited=2E If you receive this email in error=2C please notify the se=
nder by phone or email
 immediately and delete it!
 ************************************************************************=
*****************

----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40cisco=2E=
com=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=C8=FD=2C =CB=C4=D4=C2 1=C8=D5=2C 2009 =C9=CF=
=CE=E71=3A10
=D6=F7=CC=E2=3A RE=3A Revised text of the proposal for describing the fun=
ctionality of Rate Adaptation
=CA=D5=BC=FE=C8=CB=3A Peilin YANG =3Cyangpeilin=40huawei=2Ecom=3E=2C =22A=
li C=2E Begen (abegen)=22 =3Cabegen=40cisco=2Ecom=3E=2C Tom=2EVan=5FCaene=
gem=40alcatel-lucent=2Ebe=2C zeevvax=40microsoft=2Ecom
=B3=AD=CB=CD=3A Roni Even =3Cron=2Eeven=2Etlv=40gmail=2Ecom=3E=2C Tom Tay=
lor =3Ctom=2Etaylor=40rogers=2Ecom=3E=2C Ye-Kui Wang =3Cyekuiwang=40huawe=
i=2Ecom=3E

=3E Peilin/Zeev
=3E =

=3E Thanks for the detailed suggestions=2E
=3E =

=3E I am still considering whether the RMS-I and RMS-T messages need a =

=3E sequence number=2E It sounds like we want a sequence number on the =

=3E RMS-I=2C but do not want one on the RMS-T=2E If we have a seqnum on 2=
 =

=3E of the 3 messages=2C I am inclined to put one on all of them=2E Does =

=3E anybody else have an opinion=3F
=3E =

=3E I should have a straw man proposal for fixed vs TLV fields in the =

=3E next few hours (worst case tomorrow)=2E We can discuss in detail via =

=3E email or on a call=2E
=3E =

=3E I am going through the document with Peilin=27s suggested word-
=3E smithing changes (as modified by Zeev and Tom)=2E I am incorporating =

=3E many of the suggested changes=2C with exceptions noted below-
=3E =

=3E 1- Regarding =22receiver=22 and =22RTP receiver=22 - I have changed m=
any =

=3E instances of =22receiver=22 to =22RTP receiver=22=2C but left some of=
 the =

=3E references as =22receiver=22 for readability=2E I tried to use =22RTP=
 =

=3E receiver=22 when referencing protocol specific actions=2C but use =

=3E =22receiver=22 when discussing a generic device=2E This is not large =

=3E point=2C so some areas could go either way without a loss of clarity=2E=

=3E =

=3E 2- Same comments regarding =22burst=22 and =22unicast burst=22=2E I d=
id =

=3E eliminate all instances of =22RTP burst=22
=3E =

=3E =

=3E =

=3E I should have a document out this evening or tomorrow afternoon =

=3E (depending on how many interrupts the =22real job=22 gives me today=2E=

=3E =

=3E Bvs
=3E =

=3E =

=3E -----Original Message-----
=3E From=3A Peilin YANG =5Bmailto=3Ayangpeilin=40huawei=2Ecom=5D =

=3E Sent=3A Monday=2C March 30=2C 2009 7=3A20 AM
=3E To=3A Bill Ver Steeg (versteb)=3B Ali C=2E Begen (abegen)=3B =

=3E Tom=2EVan=5FCaenegem=40alcatel-lucent=2Ebe=3B zeevvax=40microsoft=2Ec=
om
=3E Cc=3A Roni Even=3B Tom Taylor=3B Ye-Kui Wang
=3E Subject=3A Revised text of the proposal for describing the =

=3E functionality of Rate Adaptation
=3E =

=3E Dear Bill=2C Ali=2C Tom and Zeev=2C
=3E =

=3E I added some revised text of the proposal for describing the =

=3E functionality of Rate Adaptation for the Burst=2E Please see the =

=3E attached file=2E The revised text mainly involves in a new step 3=2C =

=3E part of step 1=2C =227=2E1=2E RMS Request=22 and Figure 3 depicts an =
example =

=3E of messaging flow=2E
=3E I am very glad to disucss it with you=2E In the attachment file it =

=3E also includes a little revision=2E Please see if not suitable for =

=3E revison=2E
=3E In addition=2C I think=2C the following suggestions also need to be =

=3E discussed=2E  =

=3E =

=3E 1) IGMP =

=3E =

=3E In the draft like the following text it only describes the IGMP =

=3E based on IPv4=2E
=3E =

=3E =22When a receiver wishes to join the same multicast session=2C =

=3E instead of simply issuing   an Internet Group Management Protocol =

=3E (IGMP) =5BRFC3376=5D Join message=2C it sends a request to the feedba=
ck =

=3E target address for the session asking for the Reference =

=3E Information=2E =22
=3E =

=3E Do you think it is necessary to consider MLD protocol based on =

=3E IPv6 in the document as follows=3F =

=3E =

=3E =22When a receiver wishes to join the same multicast session=2C =

=3E instead of simply issuing   an Internet Group Management Protocol =

=3E (IGMP) =5BRFC3376=5D or Multicast Listener Discovery (MLD) =5BRFC 381=
0=5D =

=3E Join message=2C it sends a request to the feedback target address =

=3E for the session asking for the Reference Information=2E =22
=3E =

=3E 2) RMS-I message
=3E =

=3E In the RMS-I message=2C the present values of Response include 0=2C1 =

=3E and 2=2E The value of 0 indicates that the rapid synchronization =

=3E request has been rejected=2E Nevertheless=2C in the RMS-I message =

=3E there is not a field to indicate a rejected reason=2E In my point of =

=3E view=2C tt is easier for RR to understand the rejected reason if =

=3E there is a field of reason=2E What=27s more=2C it is convenient to ma=
ke =

=3E statistics report=2E =

=3E =

=3E The following Figure is my proposal=2E  =

=3E =

=3E      0                   1                   2                   3
=3E      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 =

=3E 0 1
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E     =7C      MSN      =7C   Response    =7C      Reason   =7C     Rsv=
d=2E   =

=3E  =7C
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E     =7C        Extended RTP Seqnum of the First Burst Packet        =

=3E  =7C
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E     =7C                   Earliest IGMP Join Time                   =

=3E  =7C
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E     =7C               Rapid Synchronization Duration                =

=3E  =7C
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E     =7C                       Max Burst Bitrate                     =

=3E  =7C
=3E     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-
=3E +-+
=3E =

=3E        Figure 5=3A FCI field syntax for the RMS Information message
=3E =

=3E 3) =22an/the RTP Receiver=22 and =22a/the Receiver=22
=3E In the document there are many =22an/the RTP Receiver=22 and =22a/the=
 =

=3E Receiver=22=2E Do you think it is necessary to keep these terms to be=
 =

=3E consistent throughout the document=3F  I revised them as =22an/the RT=
P =

=3E Receiver=22 in the revised proposal text=2E =

=3E =

=3E 4) =22the burst=22=2C =22a/the unicast burst=22=2C =22an/the RTP burs=
t=22 and =22the =

=3E RTP unicast burst=22
=3E In the document there are many =22the burst=22=2C =22a/the unicast bu=
rst=22=2C =

=3E =22an/the RTP burst=22 and =22the RTP unicast burst=22=2E Do you thin=
k it is =

=3E necessary to keep these terms to be consistent throughout the =

=3E document=3F  I revised them as =22a/the unicast burst=22 in the revis=
ed =

=3E proposal text=2E
=3E =

=3E 5) =22joining the multicast session=22 and =22joining the multicast g=
roup=22
=3E In the document there is only one =22joining the multicast group=22=2E=
 I =

=3E modified =22joining the multicast group=22 to =22joining the multicas=
t =

=3E session=22
=3E 6) The description of =22Updated Responses=22 of step 3 In the =

=3E document there is a general description for =22Updated Responses=22 a=
s =

=3E follows=2E When I discussed with our development enginerrs=2C as on-
=3E native english speakers they said it is not completely clear to =

=3E how to implement it=2E
=3E =

=3E  =223=2E  Updated Responses=3A  RS may send more than one RMS-I messa=
ges=2C
=3E       e=2Eg=2E=2C to update the burst bitrate information when the =

=3E bitrate is
=3E       adapted and/or to signal RR in real time to join the SSM =

=3E session=2E=22
=3E They said the original text from the draft - draft-levin-avt-rtcp-
=3E burst-00 is easier to understand=2E
=3E So I suggest to add a little description example at the end of the =

=3E sensence=2E
=3E The following text is my proposal=2E The last sensence is largely =

=3E extracted from the draft - draft-levin-avt-rtcp-burst-00  =

=3E =

=3E  =223=2E  Updated Responses=3A  RS may send more than one RMS-I messa=
ges=2C
=3E       e=2Eg=2E=2C to update the unicast burst bitrate information whe=
n =

=3E the bitrate is
=3E       adapted and/or to signal RR in real time to join the SSM =

=3E session=3B to notify =

=3E       RR to join the multicast session when the unicast burst =

=3E catches up with =

=3E       the original multicast session=2E=22 =

=3E =

=3E Regards=2C
=3E Peilin
=3E =

=3E *********************************************************************=
*********************
=3E This email and its attachments contain confidential information =

=3E from HUAWEI=2C which is intended only for the person or entity whose =

=3E address is listed above=2E Any use of the information contained here =

=3E in any way (including=2C but not limited to=2C total or partial =

=3E disclosure=2C reproduction=2C or dissemination) by persons other than=
 =

=3E the intended recipient(s) is prohibited=2E If you receive this email =

=3E in error=2C please notify the sender by phone or email  immediately =

=3E and delete it!
=3E =

=3E *********************************************************************=
********************
=3E ----- =D4=AD=D3=CA=BC=FE -----
=3E =B7=A2=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40cis=
co=2Ecom=3E
=3E =C8=D5=C6=DA=3A =D0=C7=C6=DA=CE=E5=2C =C8=FD=D4=C2 27=C8=D5=2C 2009 =C9=
=CF=CE=E73=3A43
=3E =D6=F7=CC=E2=3A RE=3A Hi Bill and all=2C
=3E =CA=D5=BC=FE=C8=CB=3A Peilin YANG =3Cyangpeilin=40huawei=2Ecom=3E=2C =
=22Ali C=2E Begen (abegen)=22 =

=3E =3Cabegen=40cisco=2Ecom=3E=2C Tom=2EVan=5FCaenegem=40alcatel-lucent=2E=
be=2C =

=3E zeevvax=40microsoft=2Ecom=2C Ye-Kui Wang =3Cyekuiwang=40huawei=2Ecom=3E=

=3E =B3=AD=CB=CD=3A Roni Even =3Cron=2Eeven=2Etlv=40gmail=2Ecom=3E=2C Tom=
 Taylor =

=3E =3Ctom=2Etaylor=40rogers=2Ecom=3E=2C =22Dave Oran (oran)=22 =3Coran=40=
cisco=2Ecom=3E=2C =

=3E avt=40ietf=2Eorg
=3E =3E Peilin-
=3E =3E =

=3E =3E I think we made good progress on the draft=2E We also have some =

=3E other =

=3E =3E folks looking at how the rate adaptation interacts with the rest =

=3E of =

=3E =3E the signaling=2E Hopefully we will be able to cleanly incorporate=
 =

=3E rate =

=3E =3E adaptation into the protocol=2E If you have some suggested text i=
n =

=3E that =

=3E =3E area=2C that would be great=2E
=3E =3E =

=3E =3E Bill VerSteeg
=3E =3E =

=3E =3E -----Original Message-----
=3E =3E From=3A Peilin YANG =5Bmailto=3Ayangpeilin=40huawei=2Ecom=5D
=3E =3E Sent=3A Wednesday=2C March 25=2C 2009 8=3A07 PM
=3E =3E To=3A Bill Ver Steeg (versteb)=3B Ali C=2E Begen (abegen)=3B =

=3E =3E Tom=2EVan=5FCaenegem=40alcatel-lucent=2Ebe=3B zeevvax=40microsoft=
=2Ecom=3B Ye-
=3E Kui Wang
=3E =3E Cc=3A Roni Even=3B Tom Taylor=3B Dave Oran (oran)=3B avt=40ietf=2E=
org
=3E =3E Subject=3A Hi Bill and all=2C
=3E =3E =

=3E =3E Hi Bill and all=2C
=3E =3E =

=3E =3E Thank you very much for concerning Huawei=27s draft=2E
=3E =3E I am sorry I haven=27t participated the SF meeting because I =

=3E haven=27t got =

=3E =3E an American Visa this time=2E
=3E =3E =

=3E =3E YK told me an informal result of break session yesterday=2E
=3E =3E Now I am considering the revised text for describing the =

=3E functionality =

=3E =3E of Rate Adaptation for the Burst to the common converging =

=3E solution and =

=3E =3E other possible contributing text=2E
=3E =3E =

=3E =3E Regards=2C
=3E =3E =

=3E =3E Peilin
=3E =3E =

=3E =3E =

=3E *********************************************************************=
*=3E **
=3E =3E ******************
=3E =3E This email and its attachments contain confidential information =

=3E from =

=3E =3E HUAWEI=2C which is intended only for the person or entity whose =

=3E address =

=3E =3E is listed above=2E Any use of the information contained here in =

=3E any way =

=3E =3E (including=2C but not limited to=2C total or partial disclosure=2C=
 =

=3E =3E reproduction=2C or dissemination) by persons other than the inten=
ded
=3E =3E recipient(s) is prohibited=2E If you receive this email in error=2C=
 =

=3E please =

=3E =3E notify the sender by phone or email  immediately and delete it!
=3E =3E =

=3E =3E =

=3E *********************************************************************=
*=3E **
=3E =3E *****************
=3E =3E =

=3E 

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Thread-Topic: Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fields
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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Zeev Vax" <zeevvax@microsoft.com>, <Tom.Van_Caenegem@alcatel-lucent.be>,  "Ali C. Begen (abegen)" <abegen@cisco.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fields
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Sure - hopefully this evening. Tomorrow at the latest
=20
I have gotten some feedback on the included note. Once I get this text
updated and rolled into the document I will send it to you for your
updates.
=20
bvs
=20

________________________________

From: Zeev Vax [mailto:zeevvax@microsoft.com]=20
Sent: Wednesday, April 01, 2009 1:09 PM
To: Bill Ver Steeg (versteb); Tom.Van_Caenegem@alcatel-lucent.be; Ali C.
Begen (abegen)
Cc: avt@ietf.org
Subject: RE: Rapid Acquisition of Multicast Streams (RAMS) - Fixed and
TLV fields



Bill,

=20

Please send me the draft for review once you are done with your
editorial path, before we send it out for review I would like to add my
changes to the draft. =20

=20

Zeev

=20

From: Bill Ver Steeg (versteb) [mailto:versteb@cisco.com]=20
Sent: Tuesday, March 31, 2009 11:46 AM
To: avt@ietf.org; Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali C.
Begen (abegen)
Subject: Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV
fields

=20

=20

One of the primary action items arising from the review of the Rapid
Acquisition of Multicast Streams  draft is to identify which fields are
mandatory (and thus fixed format) and which fields are optional (and
thus TLV format). Here is my take on the matter. Once we reach
consensus, I will update the draft.

=20

I have also included a description of the requested "motivating text"
for the fields that do not already have such text in the document.
Please let me know if I am on the right track.

=20

There is currently no version number in the messages. Ali and I do not
think that this is a problem because we can add new TLVs as needed. I
want to be sure we intentionally decide on this point rather than back
into it accidently. Any comments?

=20

We have 3 messages -=20

=20

RAMS-Request

 Max Receive BitRate - mandatory

 Min Buffer Fill Requirement - optional ( Add wording - The receiver may
have knowledge of it's buffering requirements. These requirements may be
application or device specific. For instance, the receiver may need to
have a certain amount of data in it's application buffer to handle
interdependency within the data. If the RS knows the buffering ability
of the receiver, it may more precisely tailor the burst to the client.
The methods used by the receiver to determine this value is application
specific, and thus out of scope. This mechanism can be used by the
receiver to signal the sender, if the receiver has such knowledge )

 Max Buffer Fill Requirement - optional ( Add wording - The receiver may
have knowledge of it's buffering requirements. These requirements may be
application or device specific. For instance, one receiver may have more
physical memory than another receiver, and thus can buffer more data. If
the RS knows the buffering ability of the receiver, it may more
precisely tailor the burst to the client. The methods used by the
receiver to determine this value is application specific, and thus out
of scope. This mechanism can be used by the receiver to signal the
sender, if the receiver has such knowledge )

=20

RAMS- Inform

 message Sequence Number - mandatory

 Response - mandatory, will include an enumerated list of response codes
- 0 is succeed, non zero is an error code. We will enumerate a list of
error codes, and note that additional error codes are possible using
vendor-specific TLV fields. The enumerated errors will include [No
server resources, No network resources, bad RMS-R request, WHAT
ELSE????] Note - we have removed the "join now semantic, as that is
captured in the "earliest join time" field.=20

 Reserved - mandatory - must be all 0s

 Max Burst Bitrate - mandatory=20

 Rapid Acquisition Duration - mandatory - 0 means "join now" (Add
wording to reflect that this is the server's current value, and can be
changed by subsequent RMS-I. Add wording to reflect that the RS can't
consume resources forever and needs to deterministically stop the flow.
The RS MUST terminate the flow at this time, even if does not get RMS-T
from the RR.  )=20

 Extended RTP Sequence Number of First Burst Packet - mandatory

 Earliest IGMP Join Time - optional (we should consider making this
mandatory in at least one RMS-I, as the client needs to eventually join)

=20

Rams -Terminate -=20

  Extended RTP seqnum of First Multicast packet - optional (intent is to
tell the RS the seqnum at which to stop the burst, but leave the session
open. If the field is not present, it means "stop now")

=20

=20

=20

Bill VerSteeg


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
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color=3D#0000ff size=3D2>Sure - hopefully this evening. Tomorrow at the=20
latest</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I have gotten some feedback on the =
included&nbsp;note. Once=20
I get this text updated and rolled into the document I will send it to =
you for=20
your updates.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634423521-01042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Zeev Vax =
[mailto:zeevvax@microsoft.com]=20
<BR><B>Sent:</B> Wednesday, April 01, 2009 1:09 PM<BR><B>To:</B> Bill =
Ver Steeg=20
(versteb); Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen=20
(abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> RE: Rapid =
Acquisition of=20
Multicast Streams (RAMS) - Fixed and TLV fields<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoPlainText><SPAN style=3D"COLOR: =
#244061">Bill,<o:p></o:p></SPAN></P>
<P class=3DMsoPlainText><SPAN style=3D"COLOR: =
#244061"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoPlainText><SPAN style=3D"COLOR: #244061">Please send me =
the draft for=20
review once you are done with your editorial path, before we send it out =
for=20
review I would like to add my changes to the draft. =
&nbsp;<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #244061; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #244061; FONT-FAMILY: =
'Calibri','sans-serif'">Zeev<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><A name=3D_MailEndCompose><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></A></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
#b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: =
medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Bill Ver =
Steeg=20
(versteb) [mailto:versteb@cisco.com] <BR><B>Sent:</B> Tuesday, March 31, =
2009=20
11:46 AM<BR><B>To:</B> avt@ietf.org; Zeev Vax;=20
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen =
(abegen)<BR><B>Subject:</B>=20
Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV=20
fields<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">One of the =
primary=20
action items arising from the review of the Rapid Acquisition of =
Multicast=20
Streams&nbsp; draft is to identify which fields are mandatory (and thus =
fixed=20
format) and which fields are optional (and thus TLV format).&nbsp;Here =
is my=20
take on the matter. Once we reach consensus, I will update the=20
draft.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">I have also =
included=20
a description of the requested "motivating text" for the fields that do =
not=20
already have such text in the document. Please let me know if I am on =
the right=20
track.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">There is =
currently no=20
version number in the messages.&nbsp;Ali and I&nbsp;do not think that =
this is a=20
problem because we can&nbsp;add new TLVs as needed. I want to be sure we =

intentionally decide on this point rather than back into it accidently. =
Any=20
comments?</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">We have 3 =
messages -=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">RAMS-Request</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;Max =
Receive=20
BitRate - mandatory</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;Min =
Buffer Fill=20
Requirement&nbsp;- optional ( Add wording - The receiver may have =
knowledge of=20
it's buffering requirements. These requirements may be application or=20
device&nbsp;specific. For instance, the receiver may need to have a =
certain=20
amount of data in it's application buffer to handle interdependency =
within the=20
data. If the RS knows the buffering ability of the receiver, it may more =

precisely tailor the burst to the client. The&nbsp;methods used&nbsp;by =
the=20
receiver to determine this value is application specific, and thus out =
of scope.=20
This mechanism can be used by the receiver to signal the sender, if=20
the&nbsp;receiver has such knowledge&nbsp;)</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;Max =
Buffer Fill=20
Requirement - optional ( Add wording - The receiver may have knowledge =
of it's=20
buffering requirements. These requirements may be application or=20
device&nbsp;specific. For instance,&nbsp;one receiver may have more =
physical=20
memory than another receiver, and thus can buffer more data. If the RS =
knows the=20
buffering ability of the receiver, it may more precisely tailor the =
burst to the=20
client.&nbsp;The&nbsp;methods used&nbsp;by the receiver to determine =
this value=20
is application specific, and thus out of scope. This mechanism can be =
used by=20
the receiver to signal the sender, if the&nbsp;receiver has such=20
knowledge&nbsp;)</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">RAMS-=20
Inform</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;message=20
Sequence Number - mandatory</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;Response -=20
mandatory, will include an enumerated list of response codes - 0 is =
succeed, non=20
zero is an error code. We will enumerate a list of error codes, =
and&nbsp;note=20
that&nbsp;additional error codes are possible using vendor-specific TLV =
fields.=20
The enumerated errors will include [No server resources, No network =
resources,=20
bad RMS-R request, WHAT ELSE????] Note - we have removed the "join now =
semantic,=20
as that is captured in the "earliest join time"=20
field.&nbsp;</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;Reserved -=20
mandatory - must be all&nbsp;0s</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;Max =
Burst=20
Bitrate - mandatory </SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">&nbsp;Rapid =

Acquisition Duration - mandatory - 0 means "join now"&nbsp;(Add wording =
to=20
reflect that this is the server's current value, and can be =
changed&nbsp;by=20
subsequent RMS-I. Add wording to reflect that the RS can't consume =
resources=20
forever and needs to&nbsp;deterministically stop the flow. The&nbsp;RS =
MUST=20
terminate the flow at this time, even if does not get RMS-T from the RR. =

&nbsp;)</SPAN> <o:p></o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;Extended RTP=20
Sequence Number of First Burst Packet -=20
mandatory</SPAN><o:p></o:p></P></DIV></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Earliest =
IGMP Join=20
Time - optional (we should consider making this mandatory in at least =
one RMS-I,=20
as the client needs to eventually join)</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Rams =
-Terminate -=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;&nbsp;Extended=20
RTP seqnum of First Multicast packet - optional (intent is to tell the =
RS the=20
seqnum at which to stop the burst, but leave the session open. If the =
field is=20
not present, it means "stop now")</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Arial','sans-serif'">&nbsp;</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Bill=20
VerSteeg</SPAN><o:p></o:p></P></DIV></DIV></DIV></BODY></HTML>

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Thread-Topic: Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Zeev Vax" <zeevvax@microsoft.com>, <Tom.Van_Caenegem@alcatel-lucent.be>, "Ali C. Begen (abegen)" <abegen@cisco.com>
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Subject: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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=20

When we added the ability to send more than one RAMS-R message, we
opened up a few design points that we now need to consider.

We need to have some nomenclature for describing the "initial RMS-R" and
"subsequent RMS-R", so I will start with using those words. Better words
would be appreciated.

The first design point I would like to address is what the RS is
required to do when it receives an RAMS-R for a burst that is already in
flow (aka "subsequent RMS-R"). We stated that the least it could do was
"ignore it". Does this imply that there is no requirement to send an
RMS-I in response to RMS-R messages referencing an in-progress flow? If
so, we should add clarifying text in the document that cautions the RR
not to assume the RAMS-R was dropped. I briefly considered requiring the
RS to send a RAMS-I for each received RAMS-R. Unfortunately, this then
leads to another set of corner cases, and may result in a chatty
exchange of messages.

We also need to consider how to deterministically differentiate between
the initial RMS-R message and subsequent RMS-R messages, particularly in
the event of packet loss or (unlikely) re-ordering. This probably leads
us to including a monotonically increasing sequence number on the RMS-R.
Dropped RMS-R messages should not be a protocol problem, as the first
one received will trigger the burst. Subsequent RMS-R messages either
change the burst shape or they do not, and they do not trigger
additional messages. Similarly, if the initial RMS-I is dropped it will
probably trigger an additional RMS-R message. Subsequent RMS-I messages
will either be received by the RR or they will not, and they will not
trigger additional messages.

So, has anybody considered these design points in detail? I have started
to go down this path, but it gets complicated quickly.

In order to limit the complexity, I suggest that we state that the first
received RMS-R triggers a RMS-I. Further RMS-R messages may flow.
Further RMS-I messages may also flow, but there is no explicit
requirement to send an RMS-I in response to a subsequent RMS-R.

=20

Bill VerSteeg


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<DIV dir=3Dltr align=3Dleft><SPAN lang=3DEN>
<P><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</P>
<P>When we added the ability to send more than one RAMS-R message, we =
opened up=20
a few design points that we now need to consider.</P>
<P>We need to have some nomenclature for describing the "initial RMS-R" =
and=20
"subsequent RMS-R", so I will start with using those words. Better words =
would=20
be appreciated.</P>
<P>The first design point I would like to address is what the RS is =
required to=20
do when it receives an RAMS-R for a burst that is already in flow (aka=20
"subsequent RMS-R"). We stated that the least it could do was "ignore =
it". Does=20
this imply that there is no requirement to send an RMS-I in response to =
RMS-R=20
messages referencing an in-progress flow? If so, we should add =
clarifying text=20
in the document that cautions the RR not to assume the RAMS-R was =
dropped. I=20
briefly considered requiring the RS to send a RAMS-I for each received =
RAMS-R.=20
Unfortunately, this then leads to another set of corner cases, and may =
result in=20
a chatty exchange of messages.</P>
<P>We also need to consider how to deterministically differentiate =
between the=20
initial RMS-R message and subsequent RMS-R messages, particularly in the =
event=20
of packet loss or (unlikely) re-ordering. This probably leads us to =
including a=20
monotonically increasing sequence number on the RMS-R.<SPAN=20
class=3D639333823-01042009> Dropped RMS-R messages should not be a =
protocol=20
problem, as the first one received will trigger the burst. Subsequent =
RMS-R=20
messages either change the burst shape or they do not, and they do not =
trigger=20
additional messages. Similarly, if the initial RMS-I is dropped it will =
probably=20
trigger an additional RMS-R message. Subsequent RMS-I messages will =
either be=20
received by the RR or they will not, and they will not trigger =
additional=20
messages.</SPAN></P>
<P>So, has anybody considered these design points in detail? I have =
started to=20
go down this path, but it gets complicated quickly.</P>
<P>In order to limit the complexity, I suggest that we state that the=20
first&nbsp;<SPAN class=3D639333823-01042009>received </SPAN>RMS-R =
triggers a=20
RMS-I. Further RMS-R messages may flow. Further RMS-I messages may also =
flow,=20
but there is no explicit requirement to send an RMS-I in response to a=20
subsequent RMS-R.</P>
<P>&nbsp;</P>
<P>Bill VerSteeg</P></SPAN></DIV></BODY></HTML>

------_=_NextPart_001_01C9B324.E3051D80--

From yekuiwang@huawei.com  Wed Apr  1 17:52:17 2009
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Date: Wed, 01 Apr 2009 20:53:10 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
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To: "'Bill Ver Steeg (versteb)'" <versteb@cisco.com>, 'Zeev Vax' <zeevvax@microsoft.com>, Tom.Van_Caenegem@alcatel-lucent.be,  "'Ali C. Begen (abegen)'" <abegen@cisco.com>
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Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple	RAMS-R messages for the same unicast flow
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Sending of subsequent RAMS-I messages by the RS is not trigged by reception
of RAMS-R messages, but by that the server has decided to change information
contained in RAMS-I messages, including burst bitrate. Only in the case when
the RS decided to change the burst bitrate according to a subsequent RAMS-R
message received, the subsequent RAMS-R message indirectly triggers a RAMS-I
message. The process to me seems rather simple. 
 
BR, YK


  _____  

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Bill
Ver Steeg (versteb)
Sent: Wednesday, April 01, 2009 7:52 PM
To: Bill Ver Steeg (versteb); Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be;
Ali C. Begen (abegen)
Cc: avt@ietf.org
Subject: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple
RAMS-R messages for the same unicast flow


 

When we added the ability to send more than one RAMS-R message, we opened up
a few design points that we now need to consider.

We need to have some nomenclature for describing the "initial RMS-R" and
"subsequent RMS-R", so I will start with using those words. Better words
would be appreciated.

The first design point I would like to address is what the RS is required to
do when it receives an RAMS-R for a burst that is already in flow (aka
"subsequent RMS-R"). We stated that the least it could do was "ignore it".
Does this imply that there is no requirement to send an RMS-I in response to
RMS-R messages referencing an in-progress flow? If so, we should add
clarifying text in the document that cautions the RR not to assume the
RAMS-R was dropped. I briefly considered requiring the RS to send a RAMS-I
for each received RAMS-R. Unfortunately, this then leads to another set of
corner cases, and may result in a chatty exchange of messages.

We also need to consider how to deterministically differentiate between the
initial RMS-R message and subsequent RMS-R messages, particularly in the
event of packet loss or (unlikely) re-ordering. This probably leads us to
including a monotonically increasing sequence number on the RMS-R. Dropped
RMS-R messages should not be a protocol problem, as the first one received
will trigger the burst. Subsequent RMS-R messages either change the burst
shape or they do not, and they do not trigger additional messages.
Similarly, if the initial RMS-I is dropped it will probably trigger an
additional RMS-R message. Subsequent RMS-I messages will either be received
by the RR or they will not, and they will not trigger additional messages.

So, has anybody considered these design points in detail? I have started to
go down this path, but it gets complicated quickly.

In order to limit the complexity, I suggest that we state that the first
received RMS-R triggers a RMS-I. Further RMS-R messages may flow. Further
RMS-I messages may also flow, but there is no explicit requirement to send
an RMS-I in response to a subsequent RMS-R.

 

Bill VerSteeg


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<DIV dir=ltr align=left><SPAN class=031304500-02042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>Sending of subsequent RAMS-I messages by the RS 
is&nbsp;not&nbsp;trigged by reception of RAMS-R messages, but by that the server 
has decided to change information contained in RAMS-I messages, including burst 
bitrate. Only in the case when the RS decided to change the burst bitrate 
according to a subsequent&nbsp;RAMS-R message received, the 
subsequent&nbsp;RAMS-R message indirectly triggers a RAMS-I message. The process 
to me seems rather simple. </FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=031304500-02042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=031304500-02042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>BR, YK</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
  <HR tabIndex=-1>
  <FONT face=Tahoma size=2><B>From:</B> avt-bounces@ietf.org 
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Bill Ver Steeg 
  (versteb)<BR><B>Sent:</B> Wednesday, April 01, 2009 7:52 PM<BR><B>To:</B> Bill 
  Ver Steeg (versteb); Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali C. 
  Begen (abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> [AVT] Rapid 
  Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the 
  same unicast flow<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=ltr align=left><SPAN lang=EN>
  <P><FONT face=Arial color=#0000ff size=2></FONT>&nbsp;</P>
  <P>When we added the ability to send more than one RAMS-R message, we opened 
  up a few design points that we now need to consider.</P>
  <P>We need to have some nomenclature for describing the "initial RMS-R" and 
  "subsequent RMS-R", so I will start with using those words. Better words would 
  be appreciated.</P>
  <P>The first design point I would like to address is what the RS is required 
  to do when it receives an RAMS-R for a burst that is already in flow (aka 
  "subsequent RMS-R"). We stated that the least it could do was "ignore it". 
  Does this imply that there is no requirement to send an RMS-I in response to 
  RMS-R messages referencing an in-progress flow? If so, we should add 
  clarifying text in the document that cautions the RR not to assume the RAMS-R 
  was dropped. I briefly considered requiring the RS to send a RAMS-I for each 
  received RAMS-R. Unfortunately, this then leads to another set of corner 
  cases, and may result in a chatty exchange of messages.</P>
  <P>We also need to consider how to deterministically differentiate between the 
  initial RMS-R message and subsequent RMS-R messages, particularly in the event 
  of packet loss or (unlikely) re-ordering. This probably leads us to including 
  a monotonically increasing sequence number on the RMS-R.<SPAN 
  class=639333823-01042009> Dropped RMS-R messages should not be a protocol 
  problem, as the first one received will trigger the burst. Subsequent RMS-R 
  messages either change the burst shape or they do not, and they do not trigger 
  additional messages. Similarly, if the initial RMS-I is dropped it will 
  probably trigger an additional RMS-R message. Subsequent RMS-I messages will 
  either be received by the RR or they will not, and they will not trigger 
  additional messages.</SPAN></P>
  <P>So, has anybody considered these design points in detail? I have started to 
  go down this path, but it gets complicated quickly.</P>
  <P>In order to limit the complexity, I suggest that we state that the 
  first&nbsp;<SPAN class=639333823-01042009>received </SPAN>RMS-R triggers a 
  RMS-I. Further RMS-R messages may flow. Further RMS-I messages may also flow, 
  but there is no explicit requirement to send an RMS-I in response to a 
  subsequent RMS-R.</P>
  <P>&nbsp;</P>
  <P>Bill VerSteeg</P></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

--Boundary_(ID_jiY/hiJsyT5x6TNeI5lJIg)--

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Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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On Apr 1, 2009, at 7:52 PM, Bill Ver Steeg (versteb) wrote:

>
> When we added the ability to send more than one RAMS-R message, we  
> opened up a few design points that we now need to consider.
>
> We need to have some nomenclature for describing the "initial RMS-R"  
> and "subsequent RMS-R", so I will start with using those words.  
> Better words would be appreciated.
>
> The first design point I would like to address is what the RS is  
> required to do when it receives an RAMS-R for a burst that is  
> already in flow (aka "subsequent RMS-R"). We stated that the least  
> it could do was "ignore it". Does this imply that there is no  
> requirement to send an RMS-I in response to RMS-R messages  
> referencing an in-progress flow?
>
No. In order to have rational state machines the RS MUST reply to each  
RAMS-R with a RAMS-I. Whether there is a "Reference to the in-progress  
flow" is a secondary issue. The key point is that semantically the  
RAMS-I returned in response to a RAMS-R can be (from a state machine  
point of view) no different from the first, because either the RS may  
never have received the first RAMS-R, or conversely the RAMS-I sent in  
response to te earlier RAMS-R might have been lost.

This is how protocols utilizing idempotency semantics have to operate.
> If so, we should add clarifying text in the document that cautions  
> the RR not to assume the RAMS-R was dropped.
>
Well, "caution" is not the term I would use. Just show the state  
machines and the whole thins should be obvious (at least it would be  
to me).
> I briefly considered requiring the RS to send a RAMS-I for each  
> received RAMS-R. Unfortunately, this then leads to another set of  
> corner cases, and may result in a chatty exchange of messages.
>
Uh, I don't think so. The only time a message other than the two  
needed to initiate a burst is sent is if a message is lost. Worst case  
this is one extra message - if the sequence is RAMS-R->RAMS- 
I(lost)...RAMS-R(retry)->RAMS-I.
As long as none of the fields in the messages have themselves non- 
idempotent semantics (e.g. by being deltas rather than absolute  
values) I fail to see what corner cases arise.
> We also need to consider how to deterministically differentiate  
> between the initial RMS-R message and subsequent RMS-R messages,  
> particularly in the event of packet loss or (unlikely) re-ordering.
>
I would argue that we in fact don't need to do this as long as we wait  
long enough for retransmission to avoid delayed duplicates or messages  
crossing on the wire. If that is that is the case I would much prefer  
that we use transaction ids instead of or in addition to sequence  
numbers. But I think neither is in fact necessary

In low-probability case the RAMS-R messages permute, the client can  
always recover by sending another RAMS-R with the burst rate it wants.

What this does point out however, is that we should have either  
DIFFERENT fields for the values in the RAMS-I that are meant to echo/ 
confirm values requested by the RAMS-R, versus those set unilaterally  
by the server, or use a DIFFERENT message type for a unilateral RAMs- 
(something other than -I) generate by the RS that was not triggered by  
receipt of a RAMS-R.

Either would work.


> This probably leads us to including a monotonically increasing  
> sequence number on the RMS-R.
>
This begs the question of who throws away state when. For example, if  
I try to do a RAMS on multicast group 1, give up, try to do a RAMS on  
multicast group 2, give up, and then try to do a RAMS on group 1  
again. What sequence number does the last of these RAMS-R message  
have? Why would it be different than if I never had done the RAMS-R on  
flow 2? How can the RS possibly know which case obtains?

I think we're actually making the protocol much more complicated and  
with more error states by introducing sequencing to avoid the problems  
that would arise from permuted messages on a retransmission time-out.  
It's just so much easier to ensure all the protocol semantics are  
idempotent.
>  Dropped RMS-R messages should not be a protocol problem, as the  
> first one received will trigger the burst. Subsequent RMS-R messages  
> either change the burst shape or they do not, and they do not  
> trigger additional messages.
>
Really? Not if they have sequence numbers and the RS failed to forget  
the state of an earlier burst.
> Similarly, if the initial RMS-I is dropped it will probably trigger  
> an additional RMS-R message. Subsequent RMS-I messages will either  
> be received by the RR or they will not, and they will not trigger  
> additional messages.
>
> So, has anybody considered these design points in detail?
>
Yes. Not for RAMS, but for numerous other request/response protocols  
that operate idempotently.
> I have started to go down this path, but it gets complicated quickly.
>
So you're discovering :-)
> In order to limit the complexity, I suggest that we state that the  
> first received RMS-R triggers a RMS-I. Further RMS-R messages may  
> flow. Further RMS-I messages may also flow, but there is no explicit  
> requirement to send an RMS-I in response to a subsequent RMS-R.
>
No, this is a bug. It never allows the client to ascertain the state  
of the server.

DaveO.

>
> Bill VerSteeg
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Ye-Kui Wang" <yekuiwang@huawei.com>, "Zeev Vax" <zeevvax@microsoft.com>,  <Tom.Van_Caenegem@alcatel-lucent.be>, "Ali C. Begen (abegen)" <abegen@cisco.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for the same unicast flow
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Thanks YK-
=20
This is also where I ended up. Simple is good.
=20
I will be sure that the text is clear in this regard.
=20
bvs
=20

________________________________

From: Ye-Kui Wang [mailto:yekuiwang@huawei.com]=20
Sent: Wednesday, April 01, 2009 8:53 PM
To: Bill Ver Steeg (versteb); 'Zeev Vax';
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen (abegen)
Cc: avt@ietf.org
Subject: RE: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -
multipleRAMS-R messages for the same unicast flow


Sending of subsequent RAMS-I messages by the RS is not trigged by
reception of RAMS-R messages, but by that the server has decided to
change information contained in RAMS-I messages, including burst
bitrate. Only in the case when the RS decided to change the burst
bitrate according to a subsequent RAMS-R message received, the
subsequent RAMS-R message indirectly triggers a RAMS-I message. The
process to me seems rather simple.=20
=20
BR, YK


________________________________

	From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
Behalf Of Bill Ver Steeg (versteb)
	Sent: Wednesday, April 01, 2009 7:52 PM
	To: Bill Ver Steeg (versteb); Zeev Vax;
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen (abegen)
	Cc: avt@ietf.org
	Subject: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -
multiple RAMS-R messages for the same unicast flow
=09
=09
	=20

	When we added the ability to send more than one RAMS-R message,
we opened up a few design points that we now need to consider.

	We need to have some nomenclature for describing the "initial
RMS-R" and "subsequent RMS-R", so I will start with using those words.
Better words would be appreciated.

	The first design point I would like to address is what the RS is
required to do when it receives an RAMS-R for a burst that is already in
flow (aka "subsequent RMS-R"). We stated that the least it could do was
"ignore it". Does this imply that there is no requirement to send an
RMS-I in response to RMS-R messages referencing an in-progress flow? If
so, we should add clarifying text in the document that cautions the RR
not to assume the RAMS-R was dropped. I briefly considered requiring the
RS to send a RAMS-I for each received RAMS-R. Unfortunately, this then
leads to another set of corner cases, and may result in a chatty
exchange of messages.

	We also need to consider how to deterministically differentiate
between the initial RMS-R message and subsequent RMS-R messages,
particularly in the event of packet loss or (unlikely) re-ordering. This
probably leads us to including a monotonically increasing sequence
number on the RMS-R. Dropped RMS-R messages should not be a protocol
problem, as the first one received will trigger the burst. Subsequent
RMS-R messages either change the burst shape or they do not, and they do
not trigger additional messages. Similarly, if the initial RMS-I is
dropped it will probably trigger an additional RMS-R message. Subsequent
RMS-I messages will either be received by the RR or they will not, and
they will not trigger additional messages.

	So, has anybody considered these design points in detail? I have
started to go down this path, but it gets complicated quickly.

	In order to limit the complexity, I suggest that we state that
the first received RMS-R triggers a RMS-I. Further RMS-R messages may
flow. Further RMS-I messages may also flow, but there is no explicit
requirement to send an RMS-I in response to a subsequent RMS-R.

	=20

	Bill VerSteeg


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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thanks YK-</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>This is also where I ended up. Simple is=20
good.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I will be sure that the text is clear in this=20
regard.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Ye-Kui Wang =
[mailto:yekuiwang@huawei.com]=20
<BR><B>Sent:</B> Wednesday, April 01, 2009 8:53 PM<BR><B>To:</B> Bill =
Ver Steeg=20
(versteb); 'Zeev Vax'; Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen=20
(abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> RE: [AVT] Rapid=20
Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for =
the same=20
unicast flow<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>Sending of subsequent RAMS-I messages by the RS =

is&nbsp;not&nbsp;trigged by reception of RAMS-R messages, but by that =
the server=20
has decided to change information contained in RAMS-I messages, =
including burst=20
bitrate. Only in the case when the RS decided to change the burst =
bitrate=20
according to a subsequent&nbsp;RAMS-R message received, the=20
subsequent&nbsp;RAMS-R message indirectly triggers a RAMS-I message. The =
process=20
to me seems rather simple. </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>BR, YK</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Bill Ver Steeg=20
  (versteb)<BR><B>Sent:</B> Wednesday, April 01, 2009 7:52 =
PM<BR><B>To:</B> Bill=20
  Ver Steeg (versteb); Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali =
C.=20
  Begen (abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> [AVT] =
Rapid=20
  Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for =
the=20
  same unicast flow<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN lang=3DEN>
  <P><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</P>
  <P>When we added the ability to send more than one RAMS-R message, we =
opened=20
  up a few design points that we now need to consider.</P>
  <P>We need to have some nomenclature for describing the "initial =
RMS-R" and=20
  "subsequent RMS-R", so I will start with using those words. Better =
words would=20
  be appreciated.</P>
  <P>The first design point I would like to address is what the RS is =
required=20
  to do when it receives an RAMS-R for a burst that is already in flow =
(aka=20
  "subsequent RMS-R"). We stated that the least it could do was "ignore =
it".=20
  Does this imply that there is no requirement to send an RMS-I in =
response to=20
  RMS-R messages referencing an in-progress flow? If so, we should add=20
  clarifying text in the document that cautions the RR not to assume the =
RAMS-R=20
  was dropped. I briefly considered requiring the RS to send a RAMS-I =
for each=20
  received RAMS-R. Unfortunately, this then leads to another set of =
corner=20
  cases, and may result in a chatty exchange of messages.</P>
  <P>We also need to consider how to deterministically differentiate =
between the=20
  initial RMS-R message and subsequent RMS-R messages, particularly in =
the event=20
  of packet loss or (unlikely) re-ordering. This probably leads us to =
including=20
  a monotonically increasing sequence number on the RMS-R.<SPAN=20
  class=3D639333823-01042009> Dropped RMS-R messages should not be a =
protocol=20
  problem, as the first one received will trigger the burst. Subsequent =
RMS-R=20
  messages either change the burst shape or they do not, and they do not =
trigger=20
  additional messages. Similarly, if the initial RMS-I is dropped it =
will=20
  probably trigger an additional RMS-R message. Subsequent RMS-I =
messages will=20
  either be received by the RR or they will not, and they will not =
trigger=20
  additional messages.</SPAN></P>
  <P>So, has anybody considered these design points in detail? I have =
started to=20
  go down this path, but it gets complicated quickly.</P>
  <P>In order to limit the complexity, I suggest that we state that the=20
  first&nbsp;<SPAN class=3D639333823-01042009>received </SPAN>RMS-R =
triggers a=20
  RMS-I. Further RMS-R messages may flow. Further RMS-I messages may =
also flow,=20
  but there is no explicit requirement to send an RMS-I in response to a =

  subsequent RMS-R.</P>
  <P>&nbsp;</P>
  <P>Bill VerSteeg</P></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

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Thread-Topic: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -multipleRAMS-R messages for the same unicast flow
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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Ye-Kui Wang" <yekuiwang@huawei.com>, "Zeev Vax" <zeevvax@microsoft.com>, <Tom.Van_Caenegem@alcatel-lucent.be>, "Ali C. Begen (abegen)" <abegen@cisco.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -multipleRAMS-R messages for the same unicast flow
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Based on further responses, the jury is still out.
=20
I will mark this section of the text with an editor's note and we will
continue the discussion.
=20
bvs

________________________________

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Bill Ver Steeg (versteb)
Sent: Wednesday, April 01, 2009 9:01 PM
To: Ye-Kui Wang; Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali C.
Begen (abegen)
Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS)
-multipleRAMS-R messages for the same unicast flow


Thanks YK-
=20
This is also where I ended up. Simple is good.
=20
I will be sure that the text is clear in this regard.
=20
bvs
=20

________________________________

From: Ye-Kui Wang [mailto:yekuiwang@huawei.com]=20
Sent: Wednesday, April 01, 2009 8:53 PM
To: Bill Ver Steeg (versteb); 'Zeev Vax';
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen (abegen)
Cc: avt@ietf.org
Subject: RE: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -
multipleRAMS-R messages for the same unicast flow


Sending of subsequent RAMS-I messages by the RS is not trigged by
reception of RAMS-R messages, but by that the server has decided to
change information contained in RAMS-I messages, including burst
bitrate. Only in the case when the RS decided to change the burst
bitrate according to a subsequent RAMS-R message received, the
subsequent RAMS-R message indirectly triggers a RAMS-I message. The
process to me seems rather simple.=20
=20
BR, YK


________________________________

	From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
Behalf Of Bill Ver Steeg (versteb)
	Sent: Wednesday, April 01, 2009 7:52 PM
	To: Bill Ver Steeg (versteb); Zeev Vax;
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen (abegen)
	Cc: avt@ietf.org
	Subject: [AVT] Rapid Acquisition of Multicast Streams (RAMS) -
multiple RAMS-R messages for the same unicast flow
=09
=09
	=20

	When we added the ability to send more than one RAMS-R message,
we opened up a few design points that we now need to consider.

	We need to have some nomenclature for describing the "initial
RMS-R" and "subsequent RMS-R", so I will start with using those words.
Better words would be appreciated.

	The first design point I would like to address is what the RS is
required to do when it receives an RAMS-R for a burst that is already in
flow (aka "subsequent RMS-R"). We stated that the least it could do was
"ignore it". Does this imply that there is no requirement to send an
RMS-I in response to RMS-R messages referencing an in-progress flow? If
so, we should add clarifying text in the document that cautions the RR
not to assume the RAMS-R was dropped. I briefly considered requiring the
RS to send a RAMS-I for each received RAMS-R. Unfortunately, this then
leads to another set of corner cases, and may result in a chatty
exchange of messages.

	We also need to consider how to deterministically differentiate
between the initial RMS-R message and subsequent RMS-R messages,
particularly in the event of packet loss or (unlikely) re-ordering. This
probably leads us to including a monotonically increasing sequence
number on the RMS-R. Dropped RMS-R messages should not be a protocol
problem, as the first one received will trigger the burst. Subsequent
RMS-R messages either change the burst shape or they do not, and they do
not trigger additional messages. Similarly, if the initial RMS-I is
dropped it will probably trigger an additional RMS-R message. Subsequent
RMS-I messages will either be received by the RR or they will not, and
they will not trigger additional messages.

	So, has anybody considered these design points in detail? I have
started to go down this path, but it gets complicated quickly.

	In order to limit the complexity, I suggest that we state that
the first received RMS-R triggers a RMS-I. Further RMS-R messages may
flow. Further RMS-I messages may also flow, but there is no explicit
requirement to send an RMS-I in response to a subsequent RMS-R.

	=20

	Bill VerSteeg


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<BODY lang=3DEN-US vLink=3Dpurple link=3Dblue>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523180301-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Based on further responses, the jury is still=20
out.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523180301-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523180301-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I will mark this section of the text with an =
editor's note=20
and we will continue the discussion.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523180301-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D523180301-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
[mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Bill Ver Steeg=20
(versteb)<BR><B>Sent:</B> Wednesday, April 01, 2009 9:01 =
PM<BR><B>To:</B> Ye-Kui=20
Wang; Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen=20
(abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> Re: [AVT] Rapid=20
Acquisition of Multicast Streams (RAMS) -multipleRAMS-R messages for the =
same=20
unicast flow<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Thanks YK-</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>This is also where I ended up. Simple is=20
good.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I will be sure that the text is clear in this=20
regard.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D030545800-02042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Ye-Kui Wang =
[mailto:yekuiwang@huawei.com]=20
<BR><B>Sent:</B> Wednesday, April 01, 2009 8:53 PM<BR><B>To:</B> Bill =
Ver Steeg=20
(versteb); 'Zeev Vax'; Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen=20
(abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> RE: [AVT] Rapid=20
Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for =
the same=20
unicast flow<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>Sending of subsequent RAMS-I messages by the RS =

is&nbsp;not&nbsp;trigged by reception of RAMS-R messages, but by that =
the server=20
has decided to change information contained in RAMS-I messages, =
including burst=20
bitrate. Only in the case when the RS decided to change the burst =
bitrate=20
according to a subsequent&nbsp;RAMS-R message received, the=20
subsequent&nbsp;RAMS-R message indirectly triggers a RAMS-I message. The =
process=20
to me seems rather simple. </FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D031304500-02042009><FONT =
face=3D&#23435;&#20307;=20
color=3D#0000ff size=3D2>BR, YK</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Bill Ver Steeg=20
  (versteb)<BR><B>Sent:</B> Wednesday, April 01, 2009 7:52 =
PM<BR><B>To:</B> Bill=20
  Ver Steeg (versteb); Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali =
C.=20
  Begen (abegen)<BR><B>Cc:</B> avt@ietf.org<BR><B>Subject:</B> [AVT] =
Rapid=20
  Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for =
the=20
  same unicast flow<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV dir=3Dltr align=3Dleft><SPAN lang=3DEN>
  <P><FONT face=3DArial color=3D#0000ff size=3D2></FONT>&nbsp;</P>
  <P>When we added the ability to send more than one RAMS-R message, we =
opened=20
  up a few design points that we now need to consider.</P>
  <P>We need to have some nomenclature for describing the "initial =
RMS-R" and=20
  "subsequent RMS-R", so I will start with using those words. Better =
words would=20
  be appreciated.</P>
  <P>The first design point I would like to address is what the RS is =
required=20
  to do when it receives an RAMS-R for a burst that is already in flow =
(aka=20
  "subsequent RMS-R"). We stated that the least it could do was "ignore =
it".=20
  Does this imply that there is no requirement to send an RMS-I in =
response to=20
  RMS-R messages referencing an in-progress flow? If so, we should add=20
  clarifying text in the document that cautions the RR not to assume the =
RAMS-R=20
  was dropped. I briefly considered requiring the RS to send a RAMS-I =
for each=20
  received RAMS-R. Unfortunately, this then leads to another set of =
corner=20
  cases, and may result in a chatty exchange of messages.</P>
  <P>We also need to consider how to deterministically differentiate =
between the=20
  initial RMS-R message and subsequent RMS-R messages, particularly in =
the event=20
  of packet loss or (unlikely) re-ordering. This probably leads us to =
including=20
  a monotonically increasing sequence number on the RMS-R.<SPAN=20
  class=3D639333823-01042009> Dropped RMS-R messages should not be a =
protocol=20
  problem, as the first one received will trigger the burst. Subsequent =
RMS-R=20
  messages either change the burst shape or they do not, and they do not =
trigger=20
  additional messages. Similarly, if the initial RMS-I is dropped it =
will=20
  probably trigger an additional RMS-R message. Subsequent RMS-I =
messages will=20
  either be received by the RR or they will not, and they will not =
trigger=20
  additional messages.</SPAN></P>
  <P>So, has anybody considered these design points in detail? I have =
started to=20
  go down this path, but it gets complicated quickly.</P>
  <P>In order to limit the complexity, I suggest that we state that the=20
  first&nbsp;<SPAN class=3D639333823-01042009>received </SPAN>RMS-R =
triggers a=20
  RMS-I. Further RMS-R messages may flow. Further RMS-I messages may =
also flow,=20
  but there is no explicit requirement to send an RMS-I in response to a =

  subsequent RMS-R.</P>
  <P>&nbsp;</P>
  <P>Bill VerSteeg</P></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

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Date: Wed, 01 Apr 2009 22:33:15 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
In-reply-to: <99D207F9-2F80-48E8-95B0-DCE28D43B469@cisco.com>
To: 'David R Oran' <oran@cisco.com>, "'Bill Ver Steeg (versteb)'" <versteb@cisco.com>
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Cc: Tom.Van_Caenegem@alcatel-lucent.be, avt@ietf.org, 'Zeev Vax' <zeevvax@microsoft.com>
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for the same unicast flow
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I am leaving all but the last comment from Dave for Bill.
 
> > In order to limit the complexity, I suggest that we state that the  
> > first received RMS-R triggers a RMS-I. Further RMS-R messages may  
> > flow. Further RMS-I messages may also flow, but there is no 
> explicit  
> > requirement to send an RMS-I in response to a subsequent RMS-R.
> >
> No, this is a bug. It never allows the client to ascertain the state  
> of the server.

This relates to the discussion about loss of RMS-I message during Bill's
presentation in San Francisco. My understanding then and also now is as
follows. If during the burst, the burst bitrate is changed by the RS due to
whatever reason, and the RS has not sent an RMS-I or the sent RMS-I was
lost, it should not be a disater to the client as it knows the its buffer
status, and it can also know the change of the burst bitrate and even the
value of the changed burst bitrate if it wants. So why there is a problem
here?

BR, YK


> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of David R Oran
> Sent: Wednesday, April 01, 2009 9:02 PM
> To: Bill Ver Steeg (versteb)
> Cc: Tom.Van_Caenegem@alcatel-lucent.be; avt@ietf.org; Zeev Vax
> Subject: Re: [AVT] Rapid Acquisition of Multicast Streams 
> (RAMS) - multipleRAMS-R messages for the same unicast flow
> 
> 
> On Apr 1, 2009, at 7:52 PM, Bill Ver Steeg (versteb) wrote:
> 
> >
> > When we added the ability to send more than one RAMS-R message, we 
> > opened up a few design points that we now need to consider.
> >
> > We need to have some nomenclature for describing the 
> "initial RMS-R"  
> > and "subsequent RMS-R", so I will start with using those words.  
> > Better words would be appreciated.
> >
> > The first design point I would like to address is what the RS is 
> > required to do when it receives an RAMS-R for a burst that 
> is already 
> > in flow (aka "subsequent RMS-R"). We stated that the least 
> it could do 
> > was "ignore it". Does this imply that there is no 
> requirement to send 
> > an RMS-I in response to RMS-R messages referencing an in-progress 
> > flow?
> >
> No. In order to have rational state machines the RS MUST 
> reply to each RAMS-R with a RAMS-I. Whether there is a 
> "Reference to the in-progress flow" is a secondary issue. The 
> key point is that semantically the RAMS-I returned in 
> response to a RAMS-R can be (from a state machine point of 
> view) no different from the first, because either the RS may 
> never have received the first RAMS-R, or conversely the 
> RAMS-I sent in response to te earlier RAMS-R might have been lost.
> 
> This is how protocols utilizing idempotency semantics have to operate.
> > If so, we should add clarifying text in the document that 
> cautions the 
> > RR not to assume the RAMS-R was dropped.
> >
> Well, "caution" is not the term I would use. Just show the 
> state machines and the whole thins should be obvious (at 
> least it would be to me).
> > I briefly considered requiring the RS to send a RAMS-I for each 
> > received RAMS-R. Unfortunately, this then leads to another set of 
> > corner cases, and may result in a chatty exchange of messages.
> >
> Uh, I don't think so. The only time a message other than the 
> two needed to initiate a burst is sent is if a message is 
> lost. Worst case this is one extra message - if the sequence 
> is RAMS-R->RAMS- I(lost)...RAMS-R(retry)->RAMS-I.
> As long as none of the fields in the messages have themselves 
> non- idempotent semantics (e.g. by being deltas rather than absolute
> values) I fail to see what corner cases arise.
> > We also need to consider how to deterministically differentiate 
> > between the initial RMS-R message and subsequent RMS-R messages, 
> > particularly in the event of packet loss or (unlikely) re-ordering.
> >
> I would argue that we in fact don't need to do this as long 
> as we wait long enough for retransmission to avoid delayed 
> duplicates or messages crossing on the wire. If that is that 
> is the case I would much prefer that we use transaction ids 
> instead of or in addition to sequence numbers. But I think 
> neither is in fact necessary
> 
> In low-probability case the RAMS-R messages permute, the 
> client can always recover by sending another RAMS-R with the 
> burst rate it wants.
> 
> What this does point out however, is that we should have 
> either DIFFERENT fields for the values in the RAMS-I that are 
> meant to echo/ confirm values requested by the RAMS-R, versus 
> those set unilaterally by the server, or use a DIFFERENT 
> message type for a unilateral RAMs- (something other than -I) 
> generate by the RS that was not triggered by receipt of a RAMS-R.
> 
> Either would work.
> 
> 
> > This probably leads us to including a monotonically increasing 
> > sequence number on the RMS-R.
> >
> This begs the question of who throws away state when. For 
> example, if I try to do a RAMS on multicast group 1, give up, 
> try to do a RAMS on multicast group 2, give up, and then try 
> to do a RAMS on group 1 again. What sequence number does the 
> last of these RAMS-R message have? Why would it be different 
> than if I never had done the RAMS-R on flow 2? How can the RS 
> possibly know which case obtains?
> 
> I think we're actually making the protocol much more 
> complicated and with more error states by introducing 
> sequencing to avoid the problems that would arise from 
> permuted messages on a retransmission time-out.  
> It's just so much easier to ensure all the protocol semantics 
> are idempotent.
> >  Dropped RMS-R messages should not be a protocol problem, 
> as the first 
> > one received will trigger the burst. Subsequent RMS-R 
> messages either 
> > change the burst shape or they do not, and they do not trigger 
> > additional messages.
> >
> Really? Not if they have sequence numbers and the RS failed 
> to forget the state of an earlier burst.
> > Similarly, if the initial RMS-I is dropped it will probably 
> trigger an 
> > additional RMS-R message. Subsequent RMS-I messages will either be 
> > received by the RR or they will not, and they will not trigger 
> > additional messages.
> >
> > So, has anybody considered these design points in detail?
> >
> Yes. Not for RAMS, but for numerous other request/response 
> protocols that operate idempotently.
> > I have started to go down this path, but it gets 
> complicated quickly.
> >
> So you're discovering :-)
> > In order to limit the complexity, I suggest that we state that the  
> > first received RMS-R triggers a RMS-I. Further RMS-R messages may  
> > flow. Further RMS-I messages may also flow, but there is no 
> explicit  
> > requirement to send an RMS-I in response to a subsequent RMS-R.
> >
> No, this is a bug. It never allows the client to ascertain the state  
> of the server.
> 
> DaveO.
> 
> >
> > Bill VerSteeg
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 



From oran@cisco.com  Wed Apr  1 19:37:57 2009
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Cc: Tom.Van_Caenegem@alcatel-lucent.be, avt@ietf.org, "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>, 'Zeev Vax' <zeevvax@microsoft.com>
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for the same unicast flow
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On Apr 1, 2009, at 10:33 PM, Ye-Kui Wang wrote:

>
> I am leaving all but the last comment from Dave for Bill.
>
>>> In order to limit the complexity, I suggest that we state that the
>>> first received RMS-R triggers a RMS-I. Further RMS-R messages may
>>> flow. Further RMS-I messages may also flow, but there is no
>> explicit
>>> requirement to send an RMS-I in response to a subsequent RMS-R.
>>>
>> No, this is a bug. It never allows the client to ascertain the state
>> of the server.
>
> This relates to the discussion about loss of RMS-I message during  
> Bill's
> presentation in San Francisco. My understanding then and also now is  
> as
> follows. If during the burst, the burst bitrate is changed by the RS  
> due to
> whatever reason, and the RS has not sent an RMS-I or the sent RMS-I  
> was
> lost, it should not be a disater to the client as it knows the its  
> buffer
> status, and it can also know the change of the burst bitrate and  
> even the
> value of the changed burst bitrate if it wants. So why there is a  
> problem
> here?
>
The client doesn't know if the burst is going to start or not, or if  
the RS aborted the burst, or finished it. The client can't tell  
whether the RS failed to send an RAMS-I because it lost the RAMS-R or  
because it thought it was redundant - in a client/server protocol each  
request needs to be replied to with a response for the state machines  
to clean up correctly.



> BR, YK
>
>
>> -----Original Message-----
>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
>> Behalf Of David R Oran
>> Sent: Wednesday, April 01, 2009 9:02 PM
>> To: Bill Ver Steeg (versteb)
>> Cc: Tom.Van_Caenegem@alcatel-lucent.be; avt@ietf.org; Zeev Vax
>> Subject: Re: [AVT] Rapid Acquisition of Multicast Streams
>> (RAMS) - multipleRAMS-R messages for the same unicast flow
>>
>>
>> On Apr 1, 2009, at 7:52 PM, Bill Ver Steeg (versteb) wrote:
>>
>>>
>>> When we added the ability to send more than one RAMS-R message, we
>>> opened up a few design points that we now need to consider.
>>>
>>> We need to have some nomenclature for describing the
>> "initial RMS-R"
>>> and "subsequent RMS-R", so I will start with using those words.
>>> Better words would be appreciated.
>>>
>>> The first design point I would like to address is what the RS is
>>> required to do when it receives an RAMS-R for a burst that
>> is already
>>> in flow (aka "subsequent RMS-R"). We stated that the least
>> it could do
>>> was "ignore it". Does this imply that there is no
>> requirement to send
>>> an RMS-I in response to RMS-R messages referencing an in-progress
>>> flow?
>>>
>> No. In order to have rational state machines the RS MUST
>> reply to each RAMS-R with a RAMS-I. Whether there is a
>> "Reference to the in-progress flow" is a secondary issue. The
>> key point is that semantically the RAMS-I returned in
>> response to a RAMS-R can be (from a state machine point of
>> view) no different from the first, because either the RS may
>> never have received the first RAMS-R, or conversely the
>> RAMS-I sent in response to te earlier RAMS-R might have been lost.
>>
>> This is how protocols utilizing idempotency semantics have to  
>> operate.
>>> If so, we should add clarifying text in the document that
>> cautions the
>>> RR not to assume the RAMS-R was dropped.
>>>
>> Well, "caution" is not the term I would use. Just show the
>> state machines and the whole thins should be obvious (at
>> least it would be to me).
>>> I briefly considered requiring the RS to send a RAMS-I for each
>>> received RAMS-R. Unfortunately, this then leads to another set of
>>> corner cases, and may result in a chatty exchange of messages.
>>>
>> Uh, I don't think so. The only time a message other than the
>> two needed to initiate a burst is sent is if a message is
>> lost. Worst case this is one extra message - if the sequence
>> is RAMS-R->RAMS- I(lost)...RAMS-R(retry)->RAMS-I.
>> As long as none of the fields in the messages have themselves
>> non- idempotent semantics (e.g. by being deltas rather than absolute
>> values) I fail to see what corner cases arise.
>>> We also need to consider how to deterministically differentiate
>>> between the initial RMS-R message and subsequent RMS-R messages,
>>> particularly in the event of packet loss or (unlikely) re-ordering.
>>>
>> I would argue that we in fact don't need to do this as long
>> as we wait long enough for retransmission to avoid delayed
>> duplicates or messages crossing on the wire. If that is that
>> is the case I would much prefer that we use transaction ids
>> instead of or in addition to sequence numbers. But I think
>> neither is in fact necessary
>>
>> In low-probability case the RAMS-R messages permute, the
>> client can always recover by sending another RAMS-R with the
>> burst rate it wants.
>>
>> What this does point out however, is that we should have
>> either DIFFERENT fields for the values in the RAMS-I that are
>> meant to echo/ confirm values requested by the RAMS-R, versus
>> those set unilaterally by the server, or use a DIFFERENT
>> message type for a unilateral RAMs- (something other than -I)
>> generate by the RS that was not triggered by receipt of a RAMS-R.
>>
>> Either would work.
>>
>>
>>> This probably leads us to including a monotonically increasing
>>> sequence number on the RMS-R.
>>>
>> This begs the question of who throws away state when. For
>> example, if I try to do a RAMS on multicast group 1, give up,
>> try to do a RAMS on multicast group 2, give up, and then try
>> to do a RAMS on group 1 again. What sequence number does the
>> last of these RAMS-R message have? Why would it be different
>> than if I never had done the RAMS-R on flow 2? How can the RS
>> possibly know which case obtains?
>>
>> I think we're actually making the protocol much more
>> complicated and with more error states by introducing
>> sequencing to avoid the problems that would arise from
>> permuted messages on a retransmission time-out.
>> It's just so much easier to ensure all the protocol semantics
>> are idempotent.
>>> Dropped RMS-R messages should not be a protocol problem,
>> as the first
>>> one received will trigger the burst. Subsequent RMS-R
>> messages either
>>> change the burst shape or they do not, and they do not trigger
>>> additional messages.
>>>
>> Really? Not if they have sequence numbers and the RS failed
>> to forget the state of an earlier burst.
>>> Similarly, if the initial RMS-I is dropped it will probably
>> trigger an
>>> additional RMS-R message. Subsequent RMS-I messages will either be
>>> received by the RR or they will not, and they will not trigger
>>> additional messages.
>>>
>>> So, has anybody considered these design points in detail?
>>>
>> Yes. Not for RAMS, but for numerous other request/response
>> protocols that operate idempotently.
>>> I have started to go down this path, but it gets
>> complicated quickly.
>>>
>> So you're discovering :-)
>>> In order to limit the complexity, I suggest that we state that the
>>> first received RMS-R triggers a RMS-I. Further RMS-R messages may
>>> flow. Further RMS-I messages may also flow, but there is no
>> explicit
>>> requirement to send an RMS-I in response to a subsequent RMS-R.
>>>
>> No, this is a bug. It never allows the client to ascertain the state
>> of the server.
>>
>> DaveO.
>>
>>>
>>> Bill VerSteeg
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>>
>
>


From yekuiwang@huawei.com  Wed Apr  1 19:56:30 2009
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Date: Wed, 01 Apr 2009 22:57:23 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
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To: 'David R Oran' <oran@cisco.com>
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Cc: Tom.Van_Caenegem@alcatel-lucent.be, avt@ietf.org, "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>, 'Zeev Vax' <zeevvax@microsoft.com>
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multipleRAMS-R messages for the same unicast flow
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> > This relates to the discussion about loss of RMS-I message during 
> > Bill's presentation in San Francisco. My understanding then 
> and also 
> > now is as follows. If during the burst, the burst bitrate 
> is changed 
> > by the RS due to whatever reason, and the RS has not sent 
> an RMS-I or 
> > the sent RMS-I was lost, it should not be a disater to the 
> client as 
> > it knows the its buffer status, and it can also know the 
> change of the 
> > burst bitrate and even the value of the changed burst bitrate if it 
> > wants. So why there is a problem here?
> >
> The client doesn't know if the burst is going to start or not,

The client knows that the burst started if burst packets are coming. Of
course the first RMS-I message is still important.

> or if the RS aborted the burst, or finished it. The 

Isn't that the RS only aborts the burst after receiving RMS-T or BYE
message? And no draft has mentioned using a message from RS to tell that it
will abort or finish the burst anyway. 

> client can't tell whether the RS failed to send an RAMS-I 
> because it lost the RAMS-R or because it thought it was 
> redundant - in a client/server protocol each request needs to 
> be replied to with a response for the state machines to clean 
> up correctly.

Agreed. 

BR, YK



From yangpeilin@huawei.com  Thu Apr  2 01:41:57 2009
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Date: Thu, 02 Apr 2009 16:39:42 +0800
From: Peilin YANG <yangpeilin@huawei.com>
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To: David R Oran <oran@cisco.com>, "Bill Ver Steeg (versteb)" <versteb@cisco.com>, Tom.Van_Caenegem@alcatel-lucent.be, Zeev Vax <zeevvax@microsoft.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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The RS shall reply to each RAMS-R fom the viewpoint of state machine=2ENe=
vertheless=2C =

subsequent RAMS-I messages is not trigged by reception of RAMS-R messages=
=2C =

so I also think that adding a DIFFERENT field for the values in the RAMS-=
I should be a good choice=2E =

In the meantime this can avoid having a seqnum on 2 of the 3 messages=2E

Regards=2C

Peilin

*************************************************************************=
*****************
 This email and its attachments contain confidential information from HUA=
WEI=2C which is intended only for the person or entity whose address is l=
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----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A David R Oran =3Coran=40cisco=2Ecom=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=CB=C4=2C =CB=C4=D4=C2 2=C8=D5=2C 2009 =C9=CF=
=CE=E79=3A02
=D6=F7=CC=E2=3A Re=3A =5BAVT=5D Rapid Acquisition of Multicast Streams (R=
AMS) - multiple RAMS-R messages for the same unicast flow
=CA=D5=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40cisco=2E=
com=3E
=B3=AD=CB=CD=3A Tom=2EVan=5FCaenegem=40alcatel-lucent=2Ebe=2C avt=40ietf=2E=
org=2C Zeev Vax =3Czeevvax=40microsoft=2Ecom=3E

=3E =

=3E On Apr 1=2C 2009=2C at 7=3A52 PM=2C Bill Ver Steeg (versteb) wrote=3A=

=3E =

=3E =3E
=3E =3E When we added the ability to send more than one RAMS-R message=2C=
 =

=3E we  =

=3E =3E opened up a few design points that we now need to consider=2E
=3E =3E
=3E =3E We need to have some nomenclature for describing the =22initial =

=3E RMS-R=22  =

=3E =3E and =22subsequent RMS-R=22=2C so I will start with using those wo=
rds=2E  =

=3E =3E Better words would be appreciated=2E
=3E =3E
=3E =3E The first design point I would like to address is what the RS is =

=3E =

=3E =3E required to do when it receives an RAMS-R for a burst that is  =

=3E =3E already in flow (aka =22subsequent RMS-R=22)=2E We stated that th=
e =

=3E least  =

=3E =3E it could do was =22ignore it=22=2E Does this imply that there is =
no  =

=3E =3E requirement to send an RMS-I in response to RMS-R messages  =

=3E =3E referencing an in-progress flow=3F
=3E =3E
=3E No=2E In order to have rational state machines the RS MUST reply to =

=3E each  =

=3E RAMS-R with a RAMS-I=2E Whether there is a =22Reference to the in-
=3E progress  =

=3E flow=22 is a secondary issue=2E The key point is that semantically th=
e =

=3E =

=3E RAMS-I returned in response to a RAMS-R can be (from a state =

=3E machine  =

=3E point of view) no different from the first=2C because either the RS =

=3E may  =

=3E never have received the first RAMS-R=2C or conversely the RAMS-I =

=3E sent in  =

=3E response to te earlier RAMS-R might have been lost=2E
=3E =

=3E This is how protocols utilizing idempotency semantics have to operate=
=2E
=3E =3E If so=2C we should add clarifying text in the document that =

=3E cautions  =

=3E =3E the RR not to assume the RAMS-R was dropped=2E
=3E =3E
=3E Well=2C =22caution=22 is not the term I would use=2E Just show the st=
ate  =

=3E machines and the whole thins should be obvious (at least it would =

=3E be  =

=3E to me)=2E
=3E =3E I briefly considered requiring the RS to send a RAMS-I for each  =

=3E =3E received RAMS-R=2E Unfortunately=2C this then leads to another se=
t =

=3E of  =

=3E =3E corner cases=2C and may result in a chatty exchange of messages=2E=

=3E =3E
=3E Uh=2C I don=27t think so=2E The only time a message other than the tw=
o  =

=3E needed to initiate a burst is sent is if a message is lost=2E Worst =

=3E case  =

=3E this is one extra message - if the sequence is RAMS-R-=3ERAMS- =

=3E I(lost)=2E=2E=2ERAMS-R(retry)-=3ERAMS-I=2E
=3E As long as none of the fields in the messages have themselves non- =

=3E idempotent semantics (e=2Eg=2E by being deltas rather than absolute  =

=3E values) I fail to see what corner cases arise=2E
=3E =3E We also need to consider how to deterministically differentiate  =

=3E =3E between the initial RMS-R message and subsequent RMS-R messages=2C=
 =

=3E =

=3E =3E particularly in the event of packet loss or (unlikely) re-orderin=
g=2E
=3E =3E
=3E I would argue that we in fact don=27t need to do this as long as we =

=3E wait  =

=3E long enough for retransmission to avoid delayed duplicates or =

=3E messages  =

=3E crossing on the wire=2E If that is that is the case I would much =

=3E prefer  =

=3E that we use transaction ids instead of or in addition to sequence  =

=3E numbers=2E But I think neither is in fact necessary
=3E =

=3E In low-probability case the RAMS-R messages permute=2C the client =

=3E can  =

=3E always recover by sending another RAMS-R with the burst rate it wants=
=2E
=3E =

=3E What this does point out however=2C is that we should have either  =

=3E DIFFERENT fields for the values in the RAMS-I that are meant to =

=3E echo/ =

=3E confirm values requested by the RAMS-R=2C versus those set =

=3E unilaterally  =

=3E by the server=2C or use a DIFFERENT message type for a unilateral =

=3E RAMs- =

=3E (something other than -I) generate by the RS that was not =

=3E triggered by  =

=3E receipt of a RAMS-R=2E
=3E =

=3E Either would work=2E
=3E =

=3E =

=3E =3E This probably leads us to including a monotonically increasing  =

=3E =3E sequence number on the RMS-R=2E
=3E =3E
=3E This begs the question of who throws away state when=2E For example=2C=
 =

=3E if  =

=3E I try to do a RAMS on multicast group 1=2C give up=2C try to do a RAM=
S =

=3E on  =

=3E multicast group 2=2C give up=2C and then try to do a RAMS on group 1 =
 =

=3E again=2E What sequence number does the last of these RAMS-R message  =

=3E have=3F Why would it be different than if I never had done the RAMS-
=3E R on  =

=3E flow 2=3F How can the RS possibly know which case obtains=3F
=3E =

=3E I think we=27re actually making the protocol much more complicated =

=3E and  =

=3E with more error states by introducing sequencing to avoid the =

=3E problems  =

=3E that would arise from permuted messages on a retransmission time-
=3E out=2E  =

=3E It=27s just so much easier to ensure all the protocol semantics are  =

=3E idempotent=2E
=3E =3E  Dropped RMS-R messages should not be a protocol problem=2C as th=
e =

=3E =

=3E =3E first one received will trigger the burst=2E Subsequent RMS-R =

=3E messages  =

=3E =3E either change the burst shape or they do not=2C and they do not  =

=3E =3E trigger additional messages=2E
=3E =3E
=3E Really=3F Not if they have sequence numbers and the RS failed to =

=3E forget  =

=3E the state of an earlier burst=2E
=3E =3E Similarly=2C if the initial RMS-I is dropped it will probably =

=3E trigger  =

=3E =3E an additional RMS-R message=2E Subsequent RMS-I messages will =

=3E either  =

=3E =3E be received by the RR or they will not=2C and they will not =

=3E trigger  =

=3E =3E additional messages=2E
=3E =3E
=3E =3E So=2C has anybody considered these design points in detail=3F
=3E =3E
=3E Yes=2E Not for RAMS=2C but for numerous other request/response =

=3E protocols  =

=3E that operate idempotently=2E
=3E =3E I have started to go down this path=2C but it gets complicated =

=3E quickly=2E=3E
=3E So you=27re discovering =3A-)
=3E =3E In order to limit the complexity=2C I suggest that we state that =

=3E the  =

=3E =3E first received RMS-R triggers a RMS-I=2E Further RMS-R messages =

=3E may  =

=3E =3E flow=2E Further RMS-I messages may also flow=2C but there is no =

=3E explicit  =

=3E =3E requirement to send an RMS-I in response to a subsequent RMS-R=2E=

=3E =3E
=3E No=2C this is a bug=2E It never allows the client to ascertain the =

=3E state  =

=3E of the server=2E
=3E =

=3E DaveO=2E
=3E =

=3E =3E
=3E =3E Bill VerSteeg
=3E =3E
=3E =3E =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=

=3E =3E Audio/Video Transport Working Group
=3E =3E avt=40ietf=2Eorg
=3E =3E https=3A//www=2Eietf=2Eorg/mailman/listinfo/avt
=3E =

=3E =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F
=3E Audio/Video Transport Working Group
=3E avt=40ietf=2Eorg
=3E https=3A//www=2Eietf=2Eorg/mailman/listinfo/avt
=3E 

From tendyntu@huawei.com  Thu Apr  2 04:52:33 2009
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Date: Thu, 02 Apr 2009 19:53:19 +0800
From: Zou ZiXuan <tendyntu@huawei.com>
In-reply-to: <fa3efb73a7b76.a7b76fa3efb73@huawei.com>
To: 'Peilin YANG' <yangpeilin@huawei.com>, 'David R Oran' <oran@cisco.com>, "'Bill Ver Steeg (versteb)'" <versteb@cisco.com>, Tom.Van_Caenegem@alcatel-lucent.be, 'Zeev Vax' <zeevvax@microsoft.com>
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Cc: avt@ietf.org
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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> The RS shall reply to each RAMS-R fom the viewpoint of state
> machine.Nevertheless,
> subsequent RAMS-I messages is not trigged by reception of RAMS-R
> messages,
[ZiXuan] If subsequent RAM-I message is not triggered by reception of
RAMS-R, then why do we need a DIFFERENT field in the RAMS-I message? I
guess you meant those RAMS-I messages generated unilaterally by RS, is
my understanding correct?
> so I also think that adding a DIFFERENT field for the values in the
RAMS-I
> should be a good choice.
> In the meantime this can avoid having a seqnum on 2 of the 3 messages.
>=20
> Regards,
>=20
> Peilin
>=20
>
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************************************************************************
************
> *****
>=20
> ----- =D4=AD=D3=CA=BC=FE -----
> =B7=A2=BC=FE=C8=CB: David R Oran <oran@cisco.com>
> =C8=D5=C6=DA: =D0=C7=C6=DA=CB=C4, =CB=C4=D4=C2 2=C8=D5, 2009 =
=C9=CF=CE=E79:02
> =D6=F7=CC=E2: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) =
-
multiple
> RAMS-R messages for the same unicast flow
> =CA=D5=BC=FE=C8=CB: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
> =B3=AD=CB=CD: Tom.Van_Caenegem@alcatel-lucent.be, avt@ietf.org, Zeev =
Vax
> <zeevvax@microsoft.com>
>=20
> >
> > On Apr 1, 2009, at 7:52 PM, Bill Ver Steeg (versteb) wrote:
> >
> > >
> > > When we added the ability to send more than one RAMS-R message,
> > we
> > > opened up a few design points that we now need to consider.
> > >
> > > We need to have some nomenclature for describing the "initial
> > RMS-R"
> > > and "subsequent RMS-R", so I will start with using those words.
> > > Better words would be appreciated.
> > >
> > > The first design point I would like to address is what the RS is
> >
> > > required to do when it receives an RAMS-R for a burst that is
> > > already in flow (aka "subsequent RMS-R"). We stated that the
> > least
> > > it could do was "ignore it". Does this imply that there is no
> > > requirement to send an RMS-I in response to RMS-R messages
> > > referencing an in-progress flow?
> > >
> > No. In order to have rational state machines the RS MUST reply to
> > each
> > RAMS-R with a RAMS-I. Whether there is a "Reference to the in-
> > progress
> > flow" is a secondary issue. The key point is that semantically the
> >
> > RAMS-I returned in response to a RAMS-R can be (from a state
> > machine
> > point of view) no different from the first, because either the RS
> > may
> > never have received the first RAMS-R, or conversely the RAMS-I
> > sent in
> > response to te earlier RAMS-R might have been lost.
> >
> > This is how protocols utilizing idempotency semantics have to
operate.
> > > If so, we should add clarifying text in the document that
> > cautions
> > > the RR not to assume the RAMS-R was dropped.
> > >
> > Well, "caution" is not the term I would use. Just show the state
> > machines and the whole thins should be obvious (at least it would
> > be
> > to me).
> > > I briefly considered requiring the RS to send a RAMS-I for each
> > > received RAMS-R. Unfortunately, this then leads to another set
> > of
> > > corner cases, and may result in a chatty exchange of messages.
> > >
> > Uh, I don't think so. The only time a message other than the two
> > needed to initiate a burst is sent is if a message is lost. Worst
> > case
> > this is one extra message - if the sequence is RAMS-R->RAMS-
> > I(lost)...RAMS-R(retry)->RAMS-I.
> > As long as none of the fields in the messages have themselves non-
> > idempotent semantics (e.g. by being deltas rather than absolute
> > values) I fail to see what corner cases arise.
> > > We also need to consider how to deterministically differentiate
> > > between the initial RMS-R message and subsequent RMS-R messages,
> >
> > > particularly in the event of packet loss or (unlikely)
re-ordering.
> > >
> > I would argue that we in fact don't need to do this as long as we
> > wait
> > long enough for retransmission to avoid delayed duplicates or
> > messages
> > crossing on the wire. If that is that is the case I would much
> > prefer
> > that we use transaction ids instead of or in addition to sequence
> > numbers. But I think neither is in fact necessary
> >
> > In low-probability case the RAMS-R messages permute, the client
> > can
> > always recover by sending another RAMS-R with the burst rate it
wants.
> >
> > What this does point out however, is that we should have either
> > DIFFERENT fields for the values in the RAMS-I that are meant to
> > echo/
> > confirm values requested by the RAMS-R, versus those set
> > unilaterally
> > by the server, or use a DIFFERENT message type for a unilateral
> > RAMs-
> > (something other than -I) generate by the RS that was not
> > triggered by
> > receipt of a RAMS-R.
> >
> > Either would work.
> >
> >
> > > This probably leads us to including a monotonically increasing
> > > sequence number on the RMS-R.
> > >
> > This begs the question of who throws away state when. For example,
> > if
> > I try to do a RAMS on multicast group 1, give up, try to do a RAMS
> > on
> > multicast group 2, give up, and then try to do a RAMS on group 1
> > again. What sequence number does the last of these RAMS-R message
> > have? Why would it be different than if I never had done the RAMS-
> > R on
> > flow 2? How can the RS possibly know which case obtains?
> >
> > I think we're actually making the protocol much more complicated
> > and
> > with more error states by introducing sequencing to avoid the
> > problems
> > that would arise from permuted messages on a retransmission time-
> > out.
> > It's just so much easier to ensure all the protocol semantics are
> > idempotent.
> > >  Dropped RMS-R messages should not be a protocol problem, as the
> >
> > > first one received will trigger the burst. Subsequent RMS-R
> > messages
> > > either change the burst shape or they do not, and they do not
> > > trigger additional messages.
> > >
> > Really? Not if they have sequence numbers and the RS failed to
> > forget
> > the state of an earlier burst.
> > > Similarly, if the initial RMS-I is dropped it will probably
> > trigger
> > > an additional RMS-R message. Subsequent RMS-I messages will
> > either
> > > be received by the RR or they will not, and they will not
> > trigger
> > > additional messages.
> > >
> > > So, has anybody considered these design points in detail?
> > >
> > Yes. Not for RAMS, but for numerous other request/response
> > protocols
> > that operate idempotently.
> > > I have started to go down this path, but it gets complicated
> > quickly.>
> > So you're discovering :-)
> > > In order to limit the complexity, I suggest that we state that
> > the
> > > first received RMS-R triggers a RMS-I. Further RMS-R messages
> > may
> > > flow. Further RMS-I messages may also flow, but there is no
> > explicit
> > > requirement to send an RMS-I in response to a subsequent RMS-R.
> > >
> > No, this is a bug. It never allows the client to ascertain the
> > state
> > of the server.
> >
> > DaveO.
> >
> > >
> > > Bill VerSteeg
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www.ietf.org/mailman/listinfo/avt
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> >
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From yangpeilin@huawei.com  Thu Apr  2 05:03:37 2009
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From: Peilin YANG <yangpeilin@huawei.com>
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Cc: Tom.Van_Caenegem@alcatel-lucent.be, 'David R Oran' <oran@cisco.com>, avt@ietf.org, "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>, 'Zeev Vax' <zeevvax@microsoft.com>
Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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Yes=2C your understanding is corrrect=2E That is my meaning=2E

Peilin

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 This email and its attachments contain confidential information from HUA=
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uding=2C but not limited to=2C total or partial disclosure=2C reproductio=
n=2C or dissemination) by persons other than the intended recipient(s) is=
 prohibited=2E If you receive this email in error=2C please notify the se=
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----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A Zou ZiXuan =3Ctendyntu=40huawei=2Ecom=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=CB=C4=2C =CB=C4=D4=C2 2=C8=D5=2C 2009 =CF=C2=
=CE=E77=3A55
=D6=F7=CC=E2=3A RE=3A =5BAVT=5D Rapid Acquisition of Multicast Streams (R=
AMS) - multiple RAMS-R messages for the same unicast flow
=CA=D5=BC=FE=C8=CB=3A =27Peilin YANG=27 =3Cyangpeilin=40huawei=2Ecom=3E=2C=
 =27David R Oran=27 =3Coran=40cisco=2Ecom=3E=2C =22=27Bill Ver Steeg (ver=
steb)=27=22 =3Cversteb=40cisco=2Ecom=3E=2C Tom=2EVan=5FCaenegem=40alcatel=
-lucent=2Ebe=2C =27Zeev Vax=27 =3Czeevvax=40microsoft=2Ecom=3E
=B3=AD=CB=CD=3A avt=40ietf=2Eorg

=3E =3E The RS shall reply to each RAMS-R fom the viewpoint of state
=3E =3E machine=2ENevertheless=2C
=3E =3E subsequent RAMS-I messages is not trigged by reception of RAMS-R
=3E =3E messages=2C
=3E =5BZiXuan=5D If subsequent RAM-I message is not triggered by receptio=
n of
=3E RAMS-R=2C then why do we need a DIFFERENT field in the RAMS-I =

=3E message=3F I
=3E guess you meant those RAMS-I messages generated unilaterally by =

=3E RS=2C is
=3E my understanding correct=3F
=3E =3E so I also think that adding a DIFFERENT field for the values in t=
he
=3E RAMS-I
=3E =3E should be a good choice=2E
=3E =3E In the meantime this can avoid having a seqnum on 2 of the 3 =

=3E messages=2E=3E =

=3E =3E Regards=2C
=3E =3E =

=3E =3E Peilin
=3E =3E =

=3E =3E
=3E *********************************************************************=
***
=3E ************
=3E =3E ******
=3E =3E  This email and its attachments contain confidential information =

=3E from=3E HUAWEI=2C which is intended only for the person or entity =

=3E whose address
=3E is
=3E =3E listed above=2E Any use of the information contained here in any =
way
=3E (including=2C
=3E =3E but not limited to=2C total or partial disclosure=2C reproduction=
=2C or
=3E dissemination)
=3E =3E by persons other than the intended recipient(s) is prohibited=2E =

=3E If you
=3E receive
=3E =3E this email in error=2C please notify the sender by phone or email=

=3E =3E  immediately and delete it!
=3E =3E =

=3E =3E
=3E *********************************************************************=
***
=3E ************
=3E =3E *****
=3E =3E =

=3E =3E ----- =D4=AD=D3=CA=BC=FE -----
=3E =3E =B7=A2=BC=FE=C8=CB=3A David R Oran =3Coran=40cisco=2Ecom=3E
=3E =3E =C8=D5=C6=DA=3A =D0=C7=C6=DA=CB=C4=2C =CB=C4=D4=C2 2=C8=D5=2C 200=
9 =C9=CF=CE=E79=3A02
=3E =3E =D6=F7=CC=E2=3A Re=3A =5BAVT=5D Rapid Acquisition of Multicast St=
reams (RAMS) -
=3E multiple
=3E =3E RAMS-R messages for the same unicast flow
=3E =3E =CA=D5=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40=
cisco=2Ecom=3E
=3E =3E =B3=AD=CB=CD=3A Tom=2EVan=5FCaenegem=40alcatel-lucent=2Ebe=2C avt=
=40ietf=2Eorg=2C Zeev Vax
=3E =3E =3Czeevvax=40microsoft=2Ecom=3E
=3E =3E =

=3E =3E =3E
=3E =3E =3E On Apr 1=2C 2009=2C at 7=3A52 PM=2C Bill Ver Steeg (versteb) =
wrote=3A
=3E =3E =3E
=3E =3E =3E =3E
=3E =3E =3E =3E When we added the ability to send more than one RAMS-R me=
ssage=2C
=3E =3E =3E we
=3E =3E =3E =3E opened up a few design points that we now need to conside=
r=2E
=3E =3E =3E =3E
=3E =3E =3E =3E We need to have some nomenclature for describing the =22i=
nitial
=3E =3E =3E RMS-R=22
=3E =3E =3E =3E and =22subsequent RMS-R=22=2C so I will start with using =
those words=2E
=3E =3E =3E =3E Better words would be appreciated=2E
=3E =3E =3E =3E
=3E =3E =3E =3E The first design point I would like to address is what th=
e =

=3E RS is
=3E =3E =3E
=3E =3E =3E =3E required to do when it receives an RAMS-R for a burst tha=
t is
=3E =3E =3E =3E already in flow (aka =22subsequent RMS-R=22)=2E We stated=
 that the
=3E =3E =3E least
=3E =3E =3E =3E it could do was =22ignore it=22=2E Does this imply that t=
here is no
=3E =3E =3E =3E requirement to send an RMS-I in response to RMS-R message=
s
=3E =3E =3E =3E referencing an in-progress flow=3F
=3E =3E =3E =3E
=3E =3E =3E No=2E In order to have rational state machines the RS MUST re=
ply to
=3E =3E =3E each
=3E =3E =3E RAMS-R with a RAMS-I=2E Whether there is a =22Reference to th=
e in-
=3E =3E =3E progress
=3E =3E =3E flow=22 is a secondary issue=2E The key point is that semanti=
cally the
=3E =3E =3E
=3E =3E =3E RAMS-I returned in response to a RAMS-R can be (from a state
=3E =3E =3E machine
=3E =3E =3E point of view) no different from the first=2C because either =
the RS
=3E =3E =3E may
=3E =3E =3E never have received the first RAMS-R=2C or conversely the RAM=
S-I
=3E =3E =3E sent in
=3E =3E =3E response to te earlier RAMS-R might have been lost=2E
=3E =3E =3E
=3E =3E =3E This is how protocols utilizing idempotency semantics have to=

=3E operate=2E
=3E =3E =3E =3E If so=2C we should add clarifying text in the document th=
at
=3E =3E =3E cautions
=3E =3E =3E =3E the RR not to assume the RAMS-R was dropped=2E
=3E =3E =3E =3E
=3E =3E =3E Well=2C =22caution=22 is not the term I would use=2E Just sho=
w the state
=3E =3E =3E machines and the whole thins should be obvious (at least it w=
ould
=3E =3E =3E be
=3E =3E =3E to me)=2E
=3E =3E =3E =3E I briefly considered requiring the RS to send a RAMS-I fo=
r each
=3E =3E =3E =3E received RAMS-R=2E Unfortunately=2C this then leads to an=
other set
=3E =3E =3E of
=3E =3E =3E =3E corner cases=2C and may result in a chatty exchange of me=
ssages=2E
=3E =3E =3E =3E
=3E =3E =3E Uh=2C I don=27t think so=2E The only time a message other tha=
n the two
=3E =3E =3E needed to initiate a burst is sent is if a message is lost=2E=
 Worst
=3E =3E =3E case
=3E =3E =3E this is one extra message - if the sequence is RAMS-R-=3ERAMS=
-
=3E =3E =3E I(lost)=2E=2E=2ERAMS-R(retry)-=3ERAMS-I=2E
=3E =3E =3E As long as none of the fields in the messages have themselves=
 =

=3E non-
=3E =3E =3E idempotent semantics (e=2Eg=2E by being deltas rather than ab=
solute
=3E =3E =3E values) I fail to see what corner cases arise=2E
=3E =3E =3E =3E We also need to consider how to deterministically differe=
ntiate
=3E =3E =3E =3E between the initial RMS-R message and subsequent RMS-R me=
ssages=2C
=3E =3E =3E
=3E =3E =3E =3E particularly in the event of packet loss or (unlikely)
=3E re-ordering=2E
=3E =3E =3E =3E
=3E =3E =3E I would argue that we in fact don=27t need to do this as long=
 as we
=3E =3E =3E wait
=3E =3E =3E long enough for retransmission to avoid delayed duplicates or=

=3E =3E =3E messages
=3E =3E =3E crossing on the wire=2E If that is that is the case I would m=
uch
=3E =3E =3E prefer
=3E =3E =3E that we use transaction ids instead of or in addition to sequ=
ence
=3E =3E =3E numbers=2E But I think neither is in fact necessary
=3E =3E =3E
=3E =3E =3E In low-probability case the RAMS-R messages permute=2C the cl=
ient
=3E =3E =3E can
=3E =3E =3E always recover by sending another RAMS-R with the burst rate =
it
=3E wants=2E
=3E =3E =3E
=3E =3E =3E What this does point out however=2C is that we should have ei=
ther
=3E =3E =3E DIFFERENT fields for the values in the RAMS-I that are meant =
to
=3E =3E =3E echo/
=3E =3E =3E confirm values requested by the RAMS-R=2C versus those set
=3E =3E =3E unilaterally
=3E =3E =3E by the server=2C or use a DIFFERENT message type for a unilat=
eral
=3E =3E =3E RAMs-
=3E =3E =3E (something other than -I) generate by the RS that was not
=3E =3E =3E triggered by
=3E =3E =3E receipt of a RAMS-R=2E
=3E =3E =3E
=3E =3E =3E Either would work=2E
=3E =3E =3E
=3E =3E =3E
=3E =3E =3E =3E This probably leads us to including a monotonically incre=
asing
=3E =3E =3E =3E sequence number on the RMS-R=2E
=3E =3E =3E =3E
=3E =3E =3E This begs the question of who throws away state when=2E For e=
xample=2C
=3E =3E =3E if
=3E =3E =3E I try to do a RAMS on multicast group 1=2C give up=2C try to =
do a RAMS
=3E =3E =3E on
=3E =3E =3E multicast group 2=2C give up=2C and then try to do a RAMS on =
group 1
=3E =3E =3E again=2E What sequence number does the last of these RAMS-R m=
essage
=3E =3E =3E have=3F Why would it be different than if I never had done th=
e =

=3E RAMS-
=3E =3E =3E R on
=3E =3E =3E flow 2=3F How can the RS possibly know which case obtains=3F
=3E =3E =3E
=3E =3E =3E I think we=27re actually making the protocol much more compli=
cated
=3E =3E =3E and
=3E =3E =3E with more error states by introducing sequencing to avoid the=

=3E =3E =3E problems
=3E =3E =3E that would arise from permuted messages on a retransmission =

=3E time-
=3E =3E =3E out=2E
=3E =3E =3E It=27s just so much easier to ensure all the protocol semanti=
cs are
=3E =3E =3E idempotent=2E
=3E =3E =3E =3E  Dropped RMS-R messages should not be a protocol problem=2C=
 as the
=3E =3E =3E
=3E =3E =3E =3E first one received will trigger the burst=2E Subsequent R=
MS-R
=3E =3E =3E messages
=3E =3E =3E =3E either change the burst shape or they do not=2C and they =
do not
=3E =3E =3E =3E trigger additional messages=2E
=3E =3E =3E =3E
=3E =3E =3E Really=3F Not if they have sequence numbers and the RS failed=
 to
=3E =3E =3E forget
=3E =3E =3E the state of an earlier burst=2E
=3E =3E =3E =3E Similarly=2C if the initial RMS-I is dropped it will prob=
ably
=3E =3E =3E trigger
=3E =3E =3E =3E an additional RMS-R message=2E Subsequent RMS-I messages =
will
=3E =3E =3E either
=3E =3E =3E =3E be received by the RR or they will not=2C and they will n=
ot
=3E =3E =3E trigger
=3E =3E =3E =3E additional messages=2E
=3E =3E =3E =3E
=3E =3E =3E =3E So=2C has anybody considered these design points in detai=
l=3F
=3E =3E =3E =3E
=3E =3E =3E Yes=2E Not for RAMS=2C but for numerous other request/respons=
e
=3E =3E =3E protocols
=3E =3E =3E that operate idempotently=2E
=3E =3E =3E =3E I have started to go down this path=2C but it gets compli=
cated
=3E =3E =3E quickly=2E=3E
=3E =3E =3E So you=27re discovering =3A-)
=3E =3E =3E =3E In order to limit the complexity=2C I suggest that we sta=
te that
=3E =3E =3E the
=3E =3E =3E =3E first received RMS-R triggers a RMS-I=2E Further RMS-R me=
ssages
=3E =3E =3E may
=3E =3E =3E =3E flow=2E Further RMS-I messages may also flow=2C but there=
 is no
=3E =3E =3E explicit
=3E =3E =3E =3E requirement to send an RMS-I in response to a subsequent =
RMS-R=2E
=3E =3E =3E =3E
=3E =3E =3E No=2C this is a bug=2E It never allows the client to ascertai=
n the
=3E =3E =3E state
=3E =3E =3E of the server=2E
=3E =3E =3E
=3E =3E =3E DaveO=2E
=3E =3E =3E
=3E =3E =3E =3E
=3E =3E =3E =3E Bill VerSteeg
=3E =3E =3E =3E
=3E =3E =3E =3E =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F
=3E =3E =3E =3E Audio/Video Transport Working Group
=3E =3E =3E =3E avt=40ietf=2Eorg
=3E =3E =3E =3E https=3A//www=2Eietf=2Eorg/mailman/listinfo/avt
=3E =3E =3E
=3E =3E =3E =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F
=3E =3E =3E Audio/Video Transport Working Group
=3E =3E =3E avt=40ietf=2Eorg
=3E =3E =3E https=3A//www=2Eietf=2Eorg/mailman/listinfo/avt
=3E =3E =3E
=3E =3E =5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=
=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=5F=

=3E =3E Audio/Video Transport Working Group
=3E =3E avt=40ietf=2Eorg
=3E =3E https=3A//www=2Eietf=2Eorg/mailman/listinfo/avt
=3E =

=3E 

From Thomas.Schierl@hhi.fraunhofer.de  Thu Apr  2 05:24:05 2009
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Subject: Re: [AVT] Fwd: I-D Action:draft-perkins-avt-rapid-rtp-sync-03.txt
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Hi Michael,<br>
<br>
What I can see from the ATSC M/H Transport spec is that there are
requirements on encoder and decoder clock precisions, which for sure
give a complete different environment than assumed by RFC3550 with
respect to the clock precision. If we could assure such clocks for all
devices which may potentially encode and stream layered media via multi
session transmission, the problem would have been solved anyway. But
that is not the case, as I tried to show by pointing to some of the
relevant discussions
on the mailing list.<br>
<br>
About your second point: RTP does not require NTP to be used as
wallclock. In RTP, there is much more freedom in selecting the source
for this clock. RTP uses
the NTP timestamp format just for indicating the wallclock in the RTCP
sender reports. Furthermore draft-perkins-avt-rapid-rtp-sync-03 does
only require to use the same wallclock as used for the SRs also for the
wallclock timestamps in the
header extensions.<br>
<br>
Another interesting point is that the ATSC Transport spec is not
mentioning SVC while mentioning AVC and AAC. If looking at the ATSC
Video Systems spec. (<a class="moz-txt-link-freetext"
 href="http://www.atsc.org/standards/cs_documents/a153-2009-03-13/S4-136r7-A153-Part-7-Video-System.pdf">S4-136r7-A153-Part-7-Video-System</a>),
there is a solution
which requires RTP timestamp synchronization for "SVC Transport in Two
RTP Sessions". This for sure solves the clock skew problem, but as
discussed in AVT, this is not a general solution
and has different implications if applied to all use cases of RTP multi
session transmission as, e.g. conversational services. Therefore we now
have a solution which seems to be agreeable by the group, which solves
the
problem and is more in line with RFC 3550, i.e. using the wallclock for
the
data alignment between the session.<br>
<br>
Best regards,<br>
Thomas<br>
<br>
<br>
Michael A Dolan wrote:
<blockquote cite="mid:0MKp8S-1Lof8c1LTC-000fjs@mrelay.perfora.net"
 type="cite">
  <meta http-equiv="Context-Type" content="text/html; charset=us-ascii">
Hi Thomas-<br>
  <br>
Thanks.&nbsp; Yes, I am familiar with the discussion.<br>
  <br>
I'm not necessarily pushing the ATSC solution, but offering some points
to consider, which apply equally to the problem at hand here, I
think.<br>
  <br>
Also, it is important to note some more things about the ATSC
text:<br>
  <br>
1. The language in the specification very carefully recommends the
clock
precision without requiring it.&nbsp; It is expected to work over routers
with jitter and delays and with media components from outside the
broadcast facility. The difference is that the client is expected to be
able to perform tight sync when the servers are able to provide the
recommended timing model (encoding one hop away, etc).&nbsp; It
gracefully degrades to "Internet" timing model
quality.<br>
  <br>
2. The clock, although "NTP" syntax, is not time of day (e.g.
reference_ID= "GPS"),
but "PPS", thus allowing the components to be encoded and
decoded without concern for end-to-end delivery delay.&nbsp; This only
works when NTP is explicitly signaled as part of the encoding and both
components share a clock (same or closely coupled encoders). So it will
not work in the general case.&nbsp; But there are many systems where it
will and perhaps consideration should be given to permitting
non-"wall clock" timebases.<br>
  <br>
The NTP hdr ext here is an encoding optimization that indeed has some
other good properties.&nbsp; Although the draft ties the NTP timestamp
back to the RTCP SR records, the timebase is not explicitly defined
(which is also true of RTCP SR).&nbsp; Some reference_ID value must be
assumed and
there is enough delta between, for example, GPS and UTC, timebases to
cause things to break if the entire system is not implicitly assumed to
be only one timebase. For example, without an explicit reference_ID,
how
could one use other timebases defined in NTPv4?&nbsp; This is a broader
issue than the hdr ext syntax, but since we're discussing NTP, I
thought
I'd bring it up.<br>
  <br>
I think the above and the other points below are relevant here and
not inconsistent with the RTCP SR methods. The fact that the ATSC
design
*can* be more precise is not alone cause to discard the techniques
developed there.<br>
  <br>
Regards,<br>
  <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Mike<br>
  <br>
At 09:34 AM 3/31/2009 +0200, <a class="moz-txt-link-abbreviated" href="mailto:Thomas.Schierl@hhi.fraunhofer.de">Thomas.Schierl@hhi.fraunhofer.de</a> wrote:<br>
  <blockquote type="cite" cite="">Dear Michael,<br>
    <br>
Thanks for the comments and the reference to the ATSC M/H.<br>
    <br>
I guess, what we are looking for in AVT is a solution which also works
for encoders having non-perfect clocks. There was a long discussion on
the list, here are two pointers:<br>
    <br>
    <a moz-do-not-send="true"
 href="http://www.ietf.org/mail-archive/web/avt/current/msg09144.html">
http://www.ietf.org/mail-archive/web/avt/current/msg09144.html</a><br>
    <a moz-do-not-send="true"
 href="http://www.ietf.org/mail-archive/web/avt/current/msg10491.html">
http://www.ietf.org/mail-archive/web/avt/current/msg10491.html</a><br>
    <br>
ATSC surely can assume to have a perfect clock in the NTP streams,
which
will not be the case on PC systems that may be used as an encoder for
multi session transmission of a layered codec. This is one major
problem
we intended to solve with the header extension while keeping the whole
approach as close as possible in line with the existing RTCP SR
method.<br>
    <br>
    <br>
Best regards,<br>
Thomas<br>
    <br>
    <br>
    <br>
______________________________________________<br>
Thomas Schierl, Dipl.-Ing.<br>
Fraunhofer HHI<br>
Senior Research Engineer<br>
Image Processing Dept. / Multimedia Transport<br>
Einsteinufer 37, 10587 Berlin, Germany<br>
PHONE: +49 30 31002 227 , FAX: +49 30 392 7200<br>
WEB: <a moz-do-not-send="true" href="http://ip.hhi.de/">http://ip.hhi.de</a>
,
SKYPE: thomas.schierl<br>
EMAIL: <a class="moz-txt-link-abbreviated" href="mailto:thomas.schierl@hhi.fraunhofer.de">thomas.schierl@hhi.fraunhofer.de</a><br>
    <br>
    <br>
    <br>
Michael A Dolan wrote:<br>
    <blockquote type="cite" cite="">Those interested in this topic
may also find this interesting:<br>
      <br>
      <a moz-do-not-send="true"
 href="http://www.atsc.org/standards/cs_documents/a153-2009-03-13/S4-132r13-A153-Part-3-Transport.pdf">
http://www.atsc.org/standards/cs_documents/a153-2009-03-13/S4-132r13-A153-Part-3-Transport.pdf</a>
      <br>
      <br>
Section 10.3.<br>
      <br>
Some differences are:<br>
      <br>
1. There is one NTP packet stream per service, rather than the encoding
optimization in the hdr ext;<br>
2. RTCP SR records are emitted "near" video random access
points, rather than on a prescribed time schedule; and<br>
3. The above works in multicast-only, UDLR networks (e.g. television
broadcast), rather than requiring a 2-way link.<br>
      <br>
#2 is interesting since in order to begin a *synchronized* decode
(video
+ audio properly correlated), a client must have acquired all of:<br>
a. NTP<br>
b. RTCP SR for each component<br>
c. video reference frame and decoding metadata (i.e. a random access
point)<br>
d. audio frame<br>
      <br>
So, the relative timing of the RTCP SR records to the RTP stream
content
is perhaps more important than the emission period.&nbsp; Having the RTCP
SR records arrive faster than the video reference frames doesn't help
of
course.&nbsp; And even if they are "fast enough", having them
arrive out of sync with the reference frames delays the service
acquisition by half the period on average.<br>
      <br>
Bandwidth utilization (how many bits to devote to RTCP and NTP) can
arguably be more efficiently managed by the server in most cases, not
individual client requests for faster RTCP SR packets. And, the client
would have no knowledge of the intended (video) reference frame periods
being used by the server, so it would not, in general, know what to ask
for.<br>
      <br>
In practice, won't a server that wants to enable faster acquisition and
sync do things like the above?&nbsp; I'm concerned that detailed requests
from clients beyond "please use more bandwidth and enable fast
sync" won't be as valuable as they seem since in the general case
the client doesn't know what to ask for exactly.<br>
      <br>
FYI, the above ATSC document is in a "candidate" state where
public comments are solicited.&nbsp; Independent of this thread, if
anyone has comments on it, I could convey them back to the ATSC.<br>
      <br>
Regards,<br>
      <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Mike<br>
      <br>
At 07:43 PM 3/9/2009 +0000, Colin Perkins wrote:<br>
      <blockquote type="cite" cite="">The changes in this version are
primarily editorial; we're aware that<br>
not all the open issues have been addressed yet. Any further
comments<br>
are welcomed.<br>
        <br>
Chairs: we'd like this to be considered as an AVT working group<br>
draft, as indicated privately.<br>
        <br>
Cheers,<br>
Colin<br>
        <br>
        <br>
        <br>
        <br>
Begin forwarded message:<br>
        <blockquote type="cite" cite="">From:
<a class="moz-txt-link-abbreviated" href="mailto:Internet-Drafts@ietf.org">Internet-Drafts@ietf.org</a><br>
Date: 9 March 2009 17:15:02 GMT<br>
To: <a class="moz-txt-link-abbreviated" href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br>
Subject: I-D Action:draft-perkins-avt-rapid-rtp-sync-03.txt<br>
Reply-To: <a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
          <br>
A New Internet-Draft is available from the on-line Internet-Drafts<br>
directories.<br>
          <br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Rapid
Synchronisation of RTP Flows<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : C. Perkins, T.
Schierl<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
draft-perkins-avt-rapid-rtp-sync-03.txt<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
17<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; :
2009-03-09<br>
          <br>
This memo outlines how RTP multimedia sessions are synchronised, and<br>
discusses how rapidly such synchronisation can occur.&nbsp; We show
that<br>
most RTP sessions can be synchronised immediately, but that the use<br>
of video switching multipoint conference units (MCUs) or large
source<br>
specific multicast (SSM) groups can greatly increase the initial<br>
synchronisation delay.&nbsp; This increase in delay can be unacceptable
to<br>
some applications that use layered and/or multi-description
codecs.<br>
          <br>
This memo updates the RTP Control Protocol (RTCP) timing rules to<br>
reduce the initial synchronisation delay for SSM sessions.&nbsp; A
new<br>
feedback packet is defined for use with the Extended RTP Profile for<br>
RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to<br>
rapidly request resynchronisation.&nbsp; Two new RTP header extensions
are<br>
defined to allow rapid synchronisation of late joiners, and
guarantee<br>
correct timestamp based decoding order recovery for layered codecs
in<br>
the presence of clock skew.<br>
          <br>
A URL for this Internet-Draft is:<br>
          <a moz-do-not-send="true"
 href="http://www.ietf.org/internet-drafts/draft-perkins-avt-rapid-rtp">
http://www.ietf.org/internet-drafts/draft-perkins-avt-rapid-rtp</a>-
sync-03.txt<br>
          <br>
Internet-Drafts are also available by anonymous FTP at:<br>
          <a moz-do-not-send="true"
 href="ftp://ftp.ietf.org/internet-drafts/">
ftp://ftp.ietf.org/internet-drafts/</a><br>
          <br>
Below is the data which will enable a MIME compliant mail reader<br>
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Audio/Video Transport Working Group<br>
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From john.mattsson@ericsson.com  Thu Apr  2 06:22:43 2009
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Thread-Topic: [AVT] SRTP store-and-forward
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From: "John Mattsson" <john.mattsson@ericsson.com>
To: "Randell Jesup" <rjesup@wgate.com>
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Subject: Re: [AVT] SRTP store-and-forward
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>Rolf Blom <rolf.j.blom@ericsson.com> writes:
>>To be able to handle an audio and a video stream using one security=20
>>context it is necessary to introduce a mechanism to give them separate

>>protection. We did that with the introduction of the SSS, the SRTP SaF

>>Source. It s also necessary to be able to indicate which e2e security=20
>>context that the media is using and thus a CCI, Crypto Context=20
>>Identifier is needed. So we believe that what we have presented is=20
>>general enough but still is a very thin design providing not more than
what is really needed.
>>
>>Cullen gave an example in which it was essential that what is=20
>>forwarded is the complete message that was received by the SaF=20
>>middlebox (do not launch the missile!). Even if I believe that such a=20
>>message wouldn't be left on an answering machine - it probably should=20
>>be delivered directly to eliminate the risk for mistakes - it points
at a possible threat.
>>This threat might perhaps be countered by some RTCP signaling messages

>>e2e. However, in the current draft-naslund-saf-00 we do not provide=20
>>e2e integrity protected RTCP. Is this a problem? Should it be added?=20
>>Or are there other application layer mechanisms that should be=20
>>applied? Note that with an e2e integrity transform and letting PUV be=20
>>a counter, it is possible to detect replays and re-orderings at the=20
>>application layer. However, it will not guarantee that all packets are
forwarded.
>>Note that the threat discussed is outside the trust model assumed for=20
>>SRTP store-and-forward, which assumes a honest but curious SaF
middlebox.

Randell Jesup <rjesup@wgate.com> writes:
>You need to at least include RTCP as well, or synchronization between=20
>streams (audio and video) cannot be done. Protecting it would be smart,

>though probably not strictly necessary. (If not protected, someone=20
>could mess with the stream sync, possibly tickling bugs in the player=20
>or causing one stream or the other to never play, to jump around, or to

>be unintelligible.)

Our intention have always been that the middlebox saves the relevant
parts of RTCP (e.g. SR, APP) and retransmits them to the final receiver.
As RTCP is protected hop-by-hop, the only one that could mess with the
synchronization is the middlebox, which according to our trust model is
trusted not to do so. Most of what can be achieved by messing with the
synchronization could be done by not forwarding chosen packets. We
thought about protecting RTCP e2e but felt that it creates more problems
than it solves. Parts of RTCP should be sent e2e and other parts only
hbh, for RTCP APP, this distinction is application dependant. If there
is a need to protect against a synchronization messing middlebox, the
audio and video should encoded in a format that can be sent in a single
stream.

/John

From zeevvax@microsoft.com  Wed Apr  1 10:08:15 2009
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From: Zeev Vax <zeevvax@microsoft.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Tom.Van_Caenegem@alcatel-lucent.be" <Tom.Van_Caenegem@alcatel-lucent.be>,  "Ali C. Begen (abegen)" <abegen@cisco.com>
Date: Wed, 1 Apr 2009 10:08:43 -0700
Thread-Topic: Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fields
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Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fields
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Bill,



Please send me the draft for review once you are done with your editorial p=
ath, before we send it out for review I would like to add my changes to the=
 draft.

Zeev

From: Bill Ver Steeg (versteb) [mailto:versteb@cisco.com]
Sent: Tuesday, March 31, 2009 11:46 AM
To: avt@ietf.org; Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Bege=
n (abegen)
Subject: Rapid Acquisition of Multicast Streams (RAMS) - Fixed and TLV fiel=
ds


One of the primary action items arising from the review of the Rapid Acquis=
ition of Multicast Streams  draft is to identify which fields are mandatory=
 (and thus fixed format) and which fields are optional (and thus TLV format=
). Here is my take on the matter. Once we reach consensus, I will update th=
e draft.

I have also included a description of the requested "motivating text" for t=
he fields that do not already have such text in the document. Please let me=
 know if I am on the right track.

There is currently no version number in the messages. Ali and I do not thin=
k that this is a problem because we can add new TLVs as needed. I want to b=
e sure we intentionally decide on this point rather than back into it accid=
ently. Any comments?

We have 3 messages -

RAMS-Request
 Max Receive BitRate - mandatory
 Min Buffer Fill Requirement - optional ( Add wording - The receiver may ha=
ve knowledge of it's buffering requirements. These requirements may be appl=
ication or device specific. For instance, the receiver may need to have a c=
ertain amount of data in it's application buffer to handle interdependency =
within the data. If the RS knows the buffering ability of the receiver, it =
may more precisely tailor the burst to the client. The methods used by the =
receiver to determine this value is application specific, and thus out of s=
cope. This mechanism can be used by the receiver to signal the sender, if t=
he receiver has such knowledge )
 Max Buffer Fill Requirement - optional ( Add wording - The receiver may ha=
ve knowledge of it's buffering requirements. These requirements may be appl=
ication or device specific. For instance, one receiver may have more physic=
al memory than another receiver, and thus can buffer more data. If the RS k=
nows the buffering ability of the receiver, it may more precisely tailor th=
e burst to the client. The methods used by the receiver to determine this v=
alue is application specific, and thus out of scope. This mechanism can be =
used by the receiver to signal the sender, if the receiver has such knowled=
ge )

RAMS- Inform
 message Sequence Number - mandatory
 Response - mandatory, will include an enumerated list of response codes - =
0 is succeed, non zero is an error code. We will enumerate a list of error =
codes, and note that additional error codes are possible using vendor-speci=
fic TLV fields. The enumerated errors will include [No server resources, No=
 network resources, bad RMS-R request, WHAT ELSE????] Note - we have remove=
d the "join now semantic, as that is captured in the "earliest join time" f=
ield.
 Reserved - mandatory - must be all 0s
 Max Burst Bitrate - mandatory
 Rapid Acquisition Duration - mandatory - 0 means "join now" (Add wording t=
o reflect that this is the server's current value, and can be changed by su=
bsequent RMS-I. Add wording to reflect that the RS can't consume resources =
forever and needs to deterministically stop the flow. The RS MUST terminate=
 the flow at this time, even if does not get RMS-T from the RR.  )
 Extended RTP Sequence Number of First Burst Packet - mandatory
 Earliest IGMP Join Time - optional (we should consider making this mandato=
ry in at least one RMS-I, as the client needs to eventually join)

Rams -Terminate -
  Extended RTP seqnum of First Multicast packet - optional (intent is to te=
ll the RS the seqnum at which to stop the burst, but leave the session open=
. If the field is not present, it means "stop now")



Bill VerSteeg

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<div class=3DSection1>

<p class=3DMsoPlainText><span style=3D'color:#244061'>Bill,<o:p></o:p></spa=
n></p>

<p class=3DMsoPlainText><span style=3D'color:#244061'><o:p>&nbsp;</o:p></sp=
an></p>

<p class=3DMsoPlainText><span style=3D'color:#244061'>Please send me the dr=
aft for
review once you are done with your editorial path, before we send it out fo=
r
review I would like to add my changes to the draft. &nbsp;<o:p></o:p></span=
></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#244061'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#244061'>Zeev<o:p></o:p></span></p>

<p class=3DMsoNormal><a name=3D"_MailEndCompose"><span style=3D'font-size:1=
1.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span><=
/a></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Bill Ver Stee=
g
(versteb) [mailto:versteb@cisco.com] <br>
<b>Sent:</b> Tuesday, March 31, 2009 11:46 AM<br>
<b>To:</b> avt@ietf.org; Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be; Ali =
C.
Begen (abegen)<br>
<b>Subject:</b> Rapid Acquisition of Multicast Streams (RAMS) - Fixed and T=
LV
fields<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>One
of the primary action items arising from the review of the Rapid Acquisitio=
n of
Multicast Streams&nbsp; draft is to identify which fields are mandatory (an=
d
thus fixed format) and which fields are optional (and thus TLV format).&nbs=
p;Here
is my take on the matter. Once we reach consensus, I will update the draft.=
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>I
have also included a description of the requested &quot;motivating text&quo=
t;
for the fields that do not already have such text in the document. Please l=
et
me know if I am on the right track.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>There
is currently no version number in the messages.&nbsp;Ali and I&nbsp;do not
think that this is a problem because we can&nbsp;add new TLVs as needed. I =
want
to be sure we intentionally decide on this point rather than back into it
accidently. Any comments?</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>We
have 3 messages - </span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>RAMS-Request</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Max
Receive BitRate - mandatory</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Min
Buffer Fill Requirement&nbsp;- optional ( Add wording - The receiver may ha=
ve
knowledge of it's buffering requirements. These requirements may be applica=
tion
or device&nbsp;specific. For instance, the receiver may need to have a cert=
ain
amount of data in it's application buffer to handle interdependency within =
the
data. If the RS knows the buffering ability of the receiver, it may more
precisely tailor the burst to the client. The&nbsp;methods used&nbsp;by the
receiver to determine this value is application specific, and thus out of
scope. This mechanism can be used by the receiver to signal the sender, if
the&nbsp;receiver has such knowledge&nbsp;)</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Max
Buffer Fill Requirement - optional ( Add wording - The receiver may have
knowledge of it's buffering requirements. These requirements may be applica=
tion
or device&nbsp;specific. For instance,&nbsp;one receiver may have more phys=
ical
memory than another receiver, and thus can buffer more data. If the RS know=
s
the buffering ability of the receiver, it may more precisely tailor the bur=
st
to the client.&nbsp;The&nbsp;methods used&nbsp;by the receiver to determine
this value is application specific, and thus out of scope. This mechanism c=
an
be used by the receiver to signal the sender, if the&nbsp;receiver has such
knowledge&nbsp;)</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>RAMS-
Inform</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;message
Sequence Number - mandatory</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Response
- mandatory, will include an enumerated list of response codes - 0 is succe=
ed,
non zero is an error code. We will enumerate a list of error codes,
and&nbsp;note that&nbsp;additional error codes are possible using
vendor-specific TLV fields. The enumerated errors will include [No server
resources, No network resources, bad RMS-R request, WHAT ELSE????] Note - w=
e
have removed the &quot;join now semantic, as that is captured in the
&quot;earliest join time&quot; field.&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Reserved
- mandatory - must be all&nbsp;0s</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Max
Burst Bitrate - mandatory </span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Rapid
Acquisition Duration - mandatory - 0 means &quot;join now&quot;&nbsp;(Add
wording to reflect that this is the server's current value, and can be
changed&nbsp;by subsequent RMS-I. Add wording to reflect that the RS can't
consume resources forever and needs to&nbsp;deterministically stop the flow=
.
The&nbsp;RS MUST terminate the flow at this time, even if does not get RMS-=
T
from the RR. &nbsp;)</span> <o:p></o:p></p>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;Extended
RTP Sequence Number of First Burst Packet - mandatory</span><o:p></o:p></p>

</div>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<span style=3D'font-size:10.0pt;font-family:"Ari=
al","sans-serif"'>Earliest
IGMP Join Time - optional (we should consider making this mandatory in at l=
east
one RMS-I, as the client needs to eventually join)</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>Rams
-Terminate - </span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;&nbsp;Extended
RTP seqnum of First Multicast packet - optional (intent is to tell the RS t=
he
seqnum at which to stop the burst, but leave the session open. If the field=
 is
not present, it means &quot;stop now&quot;)</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>&nbsp;</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif"'>Bill
VerSteeg</span><o:p></o:p></p>

</div>

</div>

</div>

</body>

</html>

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From zeevvax@microsoft.com  Wed Apr  1 19:50:40 2009
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From: Zeev Vax <zeevvax@microsoft.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>, "Tom.Van_Caenegem@alcatel-lucent.be" <Tom.Van_Caenegem@alcatel-lucent.be>,  "Ali C. Begen (abegen)" <abegen@cisco.com>
Date: Wed, 1 Apr 2009 19:51:34 -0700
Thread-Topic: Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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I agree with your conclusions, giving possible packet lose and the fact the=
 RS is not required to act on subsequent RMS-R, this would be the best way =
to keep it simple.
The statement I'm referring to is

"In order to limit the complexity, I suggest that we state that the first r=
eceived RMS-R triggers a RMS-I. Further RMS-R messages may flow. Further RM=
S-I messages may also flow, but there is no explicit requirement to send an=
 RMS-I in response to a subsequent RMS-R."
I don't think we need and monotonically increased  sequence number, relayin=
g on that will only complicate the RS decision making and error handling.

Zeev


From: Bill Ver Steeg (versteb) [mailto:versteb@cisco.com]
Sent: Wednesday, April 01, 2009 4:52 PM
To: Bill Ver Steeg (versteb); Zeev Vax; Tom.Van_Caenegem@alcatel-lucent.be;=
 Ali C. Begen (abegen)
Cc: avt@ietf.org
Subject: Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R me=
ssages for the same unicast flow




When we added the ability to send more than one RAMS-R message, we opened u=
p a few design points that we now need to consider.

We need to have some nomenclature for describing the "initial RMS-R" and "s=
ubsequent RMS-R", so I will start with using those words. Better words woul=
d be appreciated.

The first design point I would like to address is what the RS is required t=
o do when it receives an RAMS-R for a burst that is already in flow (aka "s=
ubsequent RMS-R"). We stated that the least it could do was "ignore it". Do=
es this imply that there is no requirement to send an RMS-I in response to =
RMS-R messages referencing an in-progress flow? If so, we should add clarif=
ying text in the document that cautions the RR not to assume the RAMS-R was=
 dropped. I briefly considered requiring the RS to send a RAMS-I for each r=
eceived RAMS-R. Unfortunately, this then leads to another set of corner cas=
es, and may result in a chatty exchange of messages.

We also need to consider how to deterministically differentiate between the=
 initial RMS-R message and subsequent RMS-R messages, particularly in the e=
vent of packet loss or (unlikely) re-ordering. This probably leads us to in=
cluding a monotonically increasing sequence number on the RMS-R. Dropped RM=
S-R messages should not be a protocol problem, as the first one received wi=
ll trigger the burst. Subsequent RMS-R messages either change the burst sha=
pe or they do not, and they do not trigger additional messages. Similarly, =
if the initial RMS-I is dropped it will probably trigger an additional RMS-=
R message. Subsequent RMS-I messages will either be received by the RR or t=
hey will not, and they will not trigger additional messages.

So, has anybody considered these design points in detail? I have started to=
 go down this path, but it gets complicated quickly.

In order to limit the complexity, I suggest that we state that the first re=
ceived RMS-R triggers a RMS-I. Further RMS-R messages may flow. Further RMS=
-I messages may also flow, but there is no explicit requirement to send an =
RMS-I in response to a subsequent RMS-R.



Bill VerSteeg

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<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>I agree with your conclusions, giving possible packet lose a=
nd
the fact the RS is not required to act on subsequent RMS-R, this would be t=
he best
way to keep it simple. <o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>The statement I&#8217;m referring to is<o:p></o:p></span></p=
>

<p><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color=
:#1F497D'>&#8220;</span><span
lang=3DEN>In order to limit the complexity, I suggest that we state that th=
e
first&nbsp;received RMS-R triggers a RMS-I. Further RMS-R messages may flow=
.
Further RMS-I messages may also flow, but there is no explicit requirement =
to
send an RMS-I in response to a subsequent RMS-R.&#8221;</span><span lang=3D=
EN
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'=
><o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>I don&#8217;t think we need and monotonically increased &nbs=
p;sequence
number, relaying on that will only complicate the RS decision making and er=
ror
handling.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>Zeev<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Calibri",=
"sans-serif";
color:#1F497D'>&nbsp;&nbsp;&nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><a name=3D"_MailEndCompose"><span style=3D'font-size:1=
1.0pt;
font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span><=
/a></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'>

<p class=3DMsoNormal><b><span style=3D'font-size:10.0pt;font-family:"Tahoma=
","sans-serif"'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Bill Ver Stee=
g
(versteb) [mailto:versteb@cisco.com] <br>
<b>Sent:</b> Wednesday, April 01, 2009 4:52 PM<br>
<b>To:</b> Bill Ver Steeg (versteb); Zeev Vax;
Tom.Van_Caenegem@alcatel-lucent.be; Ali C. Begen (abegen)<br>
<b>Cc:</b> avt@ietf.org<br>
<b>Subject:</b> Rapid Acquisition of Multicast Streams (RAMS) - multiple RA=
MS-R
messages for the same unicast flow<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p><span lang=3DEN>&nbsp;<o:p></o:p></span></p>

<p><span lang=3DEN>When we added the ability to send more than one RAMS-R
message, we opened up a few design points that we now need to consider.<o:p=
></o:p></span></p>

<p><span lang=3DEN>We need to have some nomenclature for describing the
&quot;initial RMS-R&quot; and &quot;subsequent RMS-R&quot;, so I will start
with using those words. Better words would be appreciated.<o:p></o:p></span=
></p>

<p><span lang=3DEN>The first design point I would like to address is what t=
he RS
is required to do when it receives an RAMS-R for a burst that is already in
flow (aka &quot;subsequent RMS-R&quot;). We stated that the least it could =
do
was &quot;ignore it&quot;. Does this imply that there is no requirement to =
send
an RMS-I in response to RMS-R messages referencing an in-progress flow? If =
so,
we should add clarifying text in the document that cautions the RR not to
assume the RAMS-R was dropped. I briefly considered requiring the RS to sen=
d a
RAMS-I for each received RAMS-R. Unfortunately, this then leads to another =
set
of corner cases, and may result in a chatty exchange of messages.<o:p></o:p=
></span></p>

<p><span lang=3DEN>We also need to consider how to deterministically
differentiate between the initial RMS-R message and subsequent RMS-R messag=
es,
particularly in the event of packet loss or (unlikely) re-ordering. This
probably leads us to including a monotonically increasing sequence number o=
n
the RMS-R. Dropped RMS-R messages should not be a protocol problem, as the
first one received will trigger the burst. Subsequent RMS-R messages either
change the burst shape or they do not, and they do not trigger additional
messages. Similarly, if the initial RMS-I is dropped it will probably trigg=
er
an additional RMS-R message. Subsequent RMS-I messages will either be recei=
ved
by the RR or they will not, and they will not trigger additional messages.<=
o:p></o:p></span></p>

<p><span lang=3DEN>So, has anybody considered these design points in detail=
? I
have started to go down this path, but it gets complicated quickly.<o:p></o=
:p></span></p>

<p><span lang=3DEN>In order to limit the complexity, I suggest that we stat=
e that
the first&nbsp;received RMS-R triggers a RMS-I. Further RMS-R messages may
flow. Further RMS-I messages may also flow, but there is no explicit
requirement to send an RMS-I in response to a subsequent RMS-R.<o:p></o:p><=
/span></p>

<p><span lang=3DEN>&nbsp;<o:p></o:p></span></p>

<p><span lang=3DEN>Bill VerSteeg<o:p></o:p></span></p>

</div>

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--_000_5177B2A348E4B64EAD7D589A027892D96A29EAB64FNAEXMSGC111re_--

From oran@cisco.com  Thu Apr  2 10:09:55 2009
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Subject: Re: [AVT] Rapid Acquisition of Multicast Streams (RAMS) - multiple RAMS-R messages for the same unicast flow
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On Apr 2, 2009, at 7:53 AM, Zou ZiXuan wrote:

>> The RS shall reply to each RAMS-R fom the viewpoint of state
>> machine.Nevertheless,
>> subsequent RAMS-I messages is not trigged by reception of RAMS-R
>> messages,
> [ZiXuan] If subsequent RAM-I message is not triggered by reception of
> RAMS-R, then why do we need a DIFFERENT field in the RAMS-I message? I
> guess you meant those RAMS-I messages generated unilaterally by RS, is
> my understanding correct?
Yes, sorry if that wasn't clear.
DaveO
>> so I also think that adding a DIFFERENT field for the values in the
> RAMS-I
>> should be a good choice.
>> In the meantime this can avoid having a seqnum on 2 of the 3  
>> messages.
>>
>> Regards,
>>
>> Peilin
>>
>>
>

From rjesup@wgate.com  Thu Apr  2 12:38:14 2009
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Date: Thu, 02 Apr 2009 15:39:41 -0400
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Subject: Re: [AVT] SRTP store-and-forward
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"John Mattsson" <john.mattsson@ericsson.com> writes:

>>Rolf Blom <rolf.j.blom@ericsson.com> writes:
>>>To be able to handle an audio and a video stream using one security 
>>>context it is necessary to introduce a mechanism to give them separate
>>>protection. We did that with the introduction of the SSS, the SRTP SaF
>>>Source. It s also necessary to be able to indicate which e2e security 
>>>context that the media is using and thus a CCI, Crypto Context 
>>>Identifier is needed. So we believe that what we have presented is 
>>>general enough but still is a very thin design providing not more than
>>>what is really needed.
>>>
>>>Cullen gave an example in which it was essential that what is forwarded
>>>is the complete message that was received by the SaF middlebox (do not
>>>launch the missile!). Even if I believe that such a message wouldn't be
>>>left on an answering machine - it probably should be delivered directly
>>>to eliminate the risk for mistakes - it points at a possible threat.
>>>This threat might perhaps be countered by some RTCP signaling messages
>>>e2e. However, in the current draft-naslund-saf-00 we do not provide e2e
>>>integrity protected RTCP. Is this a problem? Should it be added?  Or are
>>>there other application layer mechanisms that should be applied? Note
>>>that with an e2e integrity transform and letting PUV be a counter, it is
>>>possible to detect replays and re-orderings at the application
>>>layer. However, it will not guarantee that all packets are forwarded.
>>>Note that the threat discussed is outside the trust model assumed for
>>>SRTP store-and-forward, which assumes a honest but curious SaF
>>>middlebox.
>
>Randell Jesup <rjesup@wgate.com> writes:
>>You need to at least include RTCP as well, or synchronization between 
>>streams (audio and video) cannot be done. Protecting it would be smart,
>>though probably not strictly necessary. (If not protected, someone 
>>could mess with the stream sync, possibly tickling bugs in the player 
>>or causing one stream or the other to never play, to jump around, or to
>>be unintelligible.)
>
>Our intention have always been that the middlebox saves the relevant
>parts of RTCP (e.g. SR, APP) and retransmits them to the final receiver.

Ok, though I don't know how the midbox would know what RTCP packets are
relevant in all cases, especially if they're RTCP sub-packets that are
defined after the midbox is written/updated.

>As RTCP is protected hop-by-hop, the only one that could mess with the
>synchronization is the middlebox, which according to our trust model is
>trusted not to do so.

RTCP is protected end-to-end normally, not hop-to-hop (excluding B2BUAs -
perhaps you're assuming a B2BUA, but then RTP isn't end-to-end protected
either).  And I'm confused (probably by not reading the recent versions of
this): why are you assuming that the midbox is trusted?  And if it's
trusted, why are you not decrypting the message, then re-encrypting for
playback, avoiding the entirety (or most of it) of the need for this draft?

I'd assumed SRTP Store-and-forward was supposed to assume a lack of trust
in the box doing the storing, beyond that it would do the forwarding when
asked.  I.e. allow a message to be stored on an insecure server in a secure
manner; avoiding the possibility that the contents could be snooped or
modified.  (Obviously they could be deleted or truncated.)

>Most of what can be achieved by messing with the synchronization could be
>done by not forwarding chosen packets.

Simple refutation: set the current NTP time via RTCP for the one stream to
be 20 years in the future.  The user will probably hear the audio and see
no video (or a still image); or the other way around, depending on how the
client works.  (Some very dumb clients might ignore this.)  Alternate
refutation: cause the NTP time to jump around (forward and backwards); this
could cause all sorts of nasty effects depending on the client.

>We thought about protecting RTCP e2e but felt that it creates more
>problems than it solves. Parts of RTCP should be sent e2e and other parts
>only hbh, for RTCP APP, this distinction is application dependant. If
>there is a need to protect against a synchronization messing middlebox,
>the audio and video should encoded in a format that can be sent in a
>single stream.

You can do that, but then you lose the ability to leverage all the tools
that understand audio/video streams; it no longer is an extension to
existing abilities, but a new, separate tool for messages.

The question is: what is the reason(s) for SRTP usage, and how does that map
to the usecase you're trying to handle here (storing messages)?  Who are
you protecting against; what level of interference needs to be overcome?

-- 
Randell Jesup, Worldgate (developers of the Ojo videophone), ex-Amiga OS team
rjesup@wgate.com
"The fetters imposed on liberty at home have ever been forged out of the weapons
provided for defence against real, pretended, or imaginary dangers from abroad."
		- James Madison, 4th US president (1751-1836)


From tom.taylor@rogers.com  Thu Apr  2 19:14:57 2009
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Subject: [AVT] Inconsistent attribute name in RFC 3611
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RFC 3611 defines the attribute rcvr-rtt in the SDP section, but uses recv-rtt in 
the text and the IANA Considerations section. Alan Clark suggests that the SDP 
section should prevail, but it seems like a good idea to check on what has been 
implemented first. Do people have views one way or the other?

Tom Taylor

From yangpeilin@huawei.com  Thu Apr  2 19:53:46 2009
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Hi Bill=2C

I will review the draft and send comments later=2E

Peilin

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----- =D4=AD=D3=CA=BC=FE -----
=B7=A2=BC=FE=C8=CB=3A =22Bill Ver Steeg (versteb)=22 =3Cversteb=40cisco=2E=
com=3E
=C8=D5=C6=DA=3A =D0=C7=C6=DA=CE=E5=2C =CB=C4=D4=C2 3=C8=D5=2C 2009 =C9=CF=
=CE=E73=3A47
=D6=F7=CC=E2=3A Update to Rapid Acquisition of Multicast Sessions (RAMS) =
document from output of IETF break out session
=CA=D5=BC=FE=C8=CB=3A Roni Even =3Cron=2Eeven=2Etlv=40gmail=2Ecom=3E=2C I=
ngemar Johansson S =3Cingemar=2Es=2Ejohansson=40ericsson=2Ecom=3E=2C zeev=
vax=40microsoft=2Ecom=2C Xavier Marjou =3Cxavier=2Emarjou=40orange-ftgrou=
p=2Ecom=3E=2C =22Dave Oran (oran)=22 =3Coran=40cisco=2Ecom=3E=2C Orit Lev=
in =3Coritl=40microsoft=2Ecom=3E=2C Peilin YANG =3Cyangpeilin=40huawei=2E=
com=3E=2C Tom Taylor =3Ctom=2Etaylor=40rogers=2Ecom=3E
=B3=AD=CB=CD=3A =22Ali C=2E Begen (abegen)=22 =3Cabegen=40cisco=2Ecom=3E=2C=
 Ye-Kui Wang =3Cyekuiwang=40huawei=2Ecom=3E=2C Tom=2EVan=5FCaenegem=40alc=
atel-lucent=2Ebe=2C Haibin Song =3Cmelodysong=40huawei=2Ecom=3E=2C zongni=
ng 63316 =3Czongning=40huawei=2Ecom=3E=2C Jonathan Lennox =3Cjonathan=40v=
idyo=2Ecom=3E=2C sean=2Esheedy=40arrisi=2Ecom

=3E The following preliminary document captures the recommended =

=3E changes to
=3E the =22Rapid Acquisition of Multicast Sessions (RAMS)=22 draft from t=
he
=3E technical break out session in San Francisco last week=2E I think =

=3E that I
=3E captured the major points from the session=2C as well as the recent
=3E clarifications on the AVT list=2E There are still a few editor=27s no=
tes
=3E that need to be resolved=2C and the embedded ascii art needs =

=3E cleanup=2C but
=3E I think we made good progress=2E
=3E =

=3E Zeev is currently making some additional editorial changes=2C but we
=3E wanted to give the attendees plus the folks who have made suggestions=

=3E the opportunity to verify the major points in a timely fashion=2E
=3E =

=3E If you feel that the document is missing significant changes=2C =

=3E please let
=3E us know=2E If you have minor editorial comments=2C please hold them u=
ntil
=3E next week=2E We will be better able to word-smith once Zeev =

=3E completes his
=3E pass through the document=2E
=3E =

=3E =

=3E Bill VerSteeg
=3E 

From magnus.westerlund@ericsson.com  Fri Apr  3 02:38:50 2009
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Subject: Re: [AVT] [Editorial Errata Reported] RFC5404 (1754)
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Hi,

I reported this errata to keep track of the fact that the reference is
not correct. I don't think there is any issue with anyone not finding
G.719 due to this.

I would recommend a state of "Hold for document update".

Cheers

Magnus

RFC Errata System skrev:
> The following errata report has been submitted for RFC5404,
> "RTP Payload Format for G.719".
> 
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=5404&eid=1754
> 
> --------------------------------------
> Type: Editorial
> Reported by: Magnus Westerlund <magnus.westerlund@ericsson.com>
> 
> Section: 12.1
> 
> Original Text
> -------------
>    [ITU-T-G719]  ITU-T, "Specification : ITU-T G.719 extension for 20
>                  kHz fullband audio", April 2008.
> 
> 
> Corrected Text
> --------------
> [ITU-T-G719]  ITU-T, "Specification: ITU-T G.719 Low-complexity, full-band audio coding for high-quality, conversational applications", June 2008.
> 
> Notes
> -----
> Steven Botzko informed me that the Reference title is not correct, nor is the publication date for the G.719 reference. However, there should be little issues with finding the correct reference based on Recommendation number.
> 
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary. 
> 
> --------------------------------------
> RFC5404 (draft-ietf-avt-rtp-g719-04)
> --------------------------------------
> Title               : RTP Payload Format for G.719
> Publication Date    : January 2009
> Author(s)           : M. Westerlund, I. Johansson
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
> 


-- 

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'RFC Errata System'" <rfc-editor@rfc-editor.org>
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Subject: Re: [AVT] [Editorial Errata Reported] RFC5404 (1754)
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Hi,
I support Magnus on this.
Roni Even

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Friday, April 03, 2009 12:39 PM
To: RFC Errata System
Cc: ingemar.s.johansson@ericsson.com; rjsparks@nostrum.com;
jon.peterson@neustar.biz; fluffy@cisco.com; ron.even.tlv@gmail.com;
tom.taylor@rogers.com; avt@ietf.org
Subject: Re: [Editorial Errata Reported] RFC5404 (1754)

Hi,

I reported this errata to keep track of the fact that the reference is
not correct. I don't think there is any issue with anyone not finding
G.719 due to this.

I would recommend a state of "Hold for document update".

Cheers

Magnus

RFC Errata System skrev:
> The following errata report has been submitted for RFC5404,
> "RTP Payload Format for G.719".
>=20
> --------------------------------------
> You may review the report below and at:
> http://www.rfc-editor.org/errata_search.php?rfc=3D5404&eid=3D1754
>=20
> --------------------------------------
> Type: Editorial
> Reported by: Magnus Westerlund <magnus.westerlund@ericsson.com>
>=20
> Section: 12.1
>=20
> Original Text
> -------------
>    [ITU-T-G719]  ITU-T, "Specification : ITU-T G.719 extension for 20
>                  kHz fullband audio", April 2008.
>=20
>=20
> Corrected Text
> --------------
> [ITU-T-G719]  ITU-T, "Specification: ITU-T G.719 Low-complexity, =
full-band
audio coding for high-quality, conversational applications", June 2008.
>=20
> Notes
> -----
> Steven Botzko informed me that the Reference title is not correct, nor =
is
the publication date for the G.719 reference. However, there should be
little issues with finding the correct reference based on Recommendation
number.
>=20
> Instructions:
> -------------
> This errata is currently posted as "Reported". If necessary, please
> use "Reply All" to discuss whether it should be verified or
> rejected. When a decision is reached, the verifying party (IESG)
> can log in to change the status and edit the report, if necessary.=20
>=20
> --------------------------------------
> RFC5404 (draft-ietf-avt-rtp-g719-04)
> --------------------------------------
> Title               : RTP Payload Format for G.719
> Publication Date    : January 2009
> Author(s)           : M. Westerlund, I. Johansson
> Category            : PROPOSED STANDARD
> Source              : Audio/Video Transport
> Area                : Real-time Applications and Infrastructure
> Stream              : IETF
> Verifying Party     : IESG
>=20


--=20

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=E4r=F6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From oran@cisco.com  Fri Apr  3 05:51:25 2009
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Subject: Re: [AVT] Inconsistent attribute name in RFC 3611
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I just checked and the Cisco IPTV code uses rcvr-rtt.

Just goes to show that for implementers they copy stuff from the  
examples rather than the spec!

So, we'd obviously prefer that the errata change the IANA registration  
and the spec text rather than the example, but if given some time to  
switch (it takes 2 release to make such a change since this is a red/ 
green style packet problem) we can deal with either.

DaveO.


>>> -----Original Message-----
>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
>> Behalf Of
>>> Tom Taylor
>>> Sent: Thursday, April 02, 2009 7:16 PM
>>> To: IETF AVT WG
>>> Subject: [AVT] Inconsistent attribute name in RFC 3611
>>>
>>> RFC 3611 defines the attribute rcvr-rtt in the SDP section,
>> but uses
>>> recv-rtt in the text and the IANA Considerations section.
>> Alan Clark
>>> suggests that the SDP section should prevail, but it seems
>> like a good
>>> idea to check on what has been implemented first. Do people
>> have views
>>> one way or the other?
>>>
>>> Tom Taylor
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>
>>


From magnus.westerlund@ericsson.com  Fri Apr  3 06:07:28 2009
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Subject: [AVT] erratas on RFC 5117 (RTP Topologies)
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Hi,

I am looking on the filed Erratas for the RTP topologies (RFC 5117).

I think all fall into the category of "Hold for document update". I
would recommend that Cullen verify them into this state.

Cheers

Magnus



RFC5117, "RTP Topologies", January 2008

Source of RFC: avt (rai)

Errata ID: 1312

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2008-02-13

Section 3.3, 5th par says:

   Stand-alone Media Translators are rare.  Most commonly, a combination
|  of Transport and Media Translators are used to translate both the
   media stream and the transport aspects of a stream between two
   transport domains (or clouds).


It should say:

   Stand-alone Media Translators are rare.  Most commonly, a combination
|  of Transport and Media Translators is used to translate both the
   media stream and the transport aspects of a stream between two
   transport domains (or clouds).


Notes:

"A combination ... *is* used ..." !

Errata ID: 1313

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2008-02-13

Section 3.3,pg.7 says:

                    [...].  Therefore, if the Receiver Reports were
   forwarded without changes, the extended highest sequence number would
   indicate that B were substantially behind in reception, while it most
|  likely it would not be.  [...]
         ^^^^

It should say:

                    [...].  Therefore, if the Receiver Reports were
   forwarded without changes, the extended highest sequence number would
   indicate that B were substantially behind in reception, while it most
|  likely would not be.  [...]
         ^

Notes:

Spurious word replication; location is 4th-to-last line on page 7.

Errata ID: 1314

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2008-02-13

Section 3.4, pg.9 says:

         [...].  The CSRC Count (CC) and CSRC fields in the RTP header
|  are used to indicate the contributors of to the newly generated
   stream.  The SSRCs of the to-be-mixed streams on the Mixer input
   appear as the CSRCs at the Mixer output.  That output stream uses a
|  unique SSRC that identifies the Mixer's stream.  The CSRC are
   forwarded between the two domains to allow for loop detection and
   identification of sources that are part of the global session.  [...]


It should say:

         [...].  The CSRC Count (CC) and CSRC fields in the RTP header
|  are used to indicate the contributors to the newly generated
   stream.  The SSRCs of the to-be-mixed streams on the Mixer input
   appear as the CSRCs at the Mixer output.  That output stream uses a
|  unique SSRC that identifies the Mixer's stream.  The CSRCs are
   forwarded between the two domains to allow for loop detection and
   identification of sources that are part of the global session.  [...]


Notes:

Near the bottom of page 9:
a) s/of to/to/
^^^
b) s/The CSRC are/The CSRCs are/
^

Errata ID: 1315

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2008-02-13

Section 3.4, pg.10 says:

   A Mixer is responsible for receiving RTCP feedback messages and
   handling them appropriately.  The definition of "appropriate" depends
   on the message itself and the context.  In some cases, the reception
   of a codec-control message may result in the generation and
   transmission of RTCP feedback messages by the Mixer to the
|  participants in the other domain.  In other cases, a message is
   handled by the Mixer itself and therefore not forwarded to any other
   domain.


It should say:

   A Mixer is responsible for receiving RTCP feedback messages and
   handling them appropriately.  The definition of "appropriate" depends
   on the message itself and the context.  In some cases, the reception
   of a codec-control message may result in the generation and
   transmission of RTCP feedback messages by the Mixer to the
|  participants in the other domain(s).  In other cases, a message is
   handled by the Mixer itself and therefore not forwarded to any other
   domain.


Notes:

Location is 4th paragraph on page 10.
Rationale: There may be more than one "other" domain;
in particular, this *is* the case in the example discussed
in the text (cf. Figure 5 on page 9).

Errata ID: 1316

Status: Reported
Type: Editorial

Reported By: Alfred Hoenes
Date Reported: 2008-02-13

Section 4.1.5, p.16 says:

  ... handled correctly in domain bridging function.  [...]

It should say:

Either:

  ... handled correctly in domain bridging functions.  [...]

Or (less preferable):

  ... handled correctly in a domain bridging function.  [...]



-- 

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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Subject: [AVT] [Editorial Errata Reported] RFC5404 (1754)
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The following errata report has been submitted for RFC5404,
"RTP Payload Format for G.719".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=5404&eid=1754

--------------------------------------
Type: Editorial
Reported by: Magnus Westerlund <magnus.westerlund@ericsson.com>

Section: 12.1

Original Text
-------------
   [ITU-T-G719]  ITU-T, "Specification : ITU-T G.719 extension for 20
                 kHz fullband audio", April 2008.


Corrected Text
--------------
[ITU-T-G719]  ITU-T, "Specification: ITU-T G.719 Low-complexity, full-band audio coding for high-quality, conversational applications", June 2008.

Notes
-----
Steven Botzko informed me that the Reference title is not correct, nor is the publication date for the G.719 reference. However, there should be little issues with finding the correct reference based on Recommendation number.

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
rejected. When a decision is reached, the verifying party (IESG)
can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC5404 (draft-ietf-avt-rtp-g719-04)
--------------------------------------
Title               : RTP Payload Format for G.719
Publication Date    : January 2009
Author(s)           : M. Westerlund, I. Johansson
Category            : PROPOSED STANDARD
Source              : Audio/Video Transport
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From john.mattsson@ericsson.com  Fri Apr  3 09:41:09 2009
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From: "John Mattsson" <john.mattsson@ericsson.com>
To: "Randell Jesup" <rjesup@wgate.com>
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Randell Jesup <rjesup@wgate.com> writes:
>>>You need to at least include RTCP as well, or synchronization between

>>>streams (audio and video) cannot be done. Protecting it would be=20
>>>smart, though probably not strictly necessary. (If not protected,=20
>>>someone could mess with the stream sync, possibly tickling bugs in
the=20
>>>player or causing one stream or the other to never play, to jump=20
>>>around, or to be unintelligible.)
=20
>>Our intention have always been that the middlebox saves the relevant=20
>>parts of RTCP (e.g. SR, APP) and retransmits them to the final
receiver.
=20
>Ok, though I don't know how the midbox would know what RTCP packets are
>relevant in all cases, especially if they're RTCP sub-packets that are
>defined after the midbox is written/updated.
=20
This is a general problem for store-and-forward of RTP and not specific
to SRTP SaF. You are right in that the middlebox cannot know which RTCP
packets are relevant in all future cases and might need to be updated. I
think that a simple policy were the middlebox forwards the information
it knows should be forwarded e2e (e.g. synch info) and the information
it does not recognize (e.g. RTCP APP) goes a long way.
=20
>>As RTCP is protected hop-by-hop, the only one that could mess with the

>>synchronization is the middlebox, which according to our trust model
is=20
>>trusted not to do so.
=20
>RTCP is protected end-to-end normally, not hop-to-hop (excluding B2BUAs
-
>perhaps you're assuming a B2BUA, but then RTP isn't end-to-end
protected
>either).  And I'm confused (probably by not reading the recent versions
of
>this): why are you assuming that the midbox is trusted?  And if it's
>trusted, why are you not decrypting the message, then re-encrypting for
>playback, avoiding the entirety (or most of it) of the need for this
>draft?
=20
>I'd assumed SRTP Store-and-forward was supposed to assume a lack of
trust
>in the box doing the storing, beyond that it would do the forwarding
when
>asked.  I.e. allow a message to be stored on an insecure server in a
>secure manner; avoiding the possibility that the contents could be
>snooped or modified.  (Obviously they could be deleted or truncated.)
=20
The middlebox is exactly as you say only trusted to do the forwarding
when asked (including unmodified synchronization info). We call this
semi-trusted in the draft. If the middlebox is totally untrusted it
would be meaningless to use for anything. We do not trust the middlebox
with cleartext data.
=20
>>Most of what can be achieved by messing with the synchronization could

>>be done by not forwarding chosen packets.
=20
>Simple refutation: set the current NTP time via RTCP for the one stream
>to be 20 years in the future.  The user will probably hear the audio
and
>see no video (or a still image); or the other way around, depending on
>how the client works.  (Some very dumb clients might ignore this.) =20
>Alternate refutation: cause the NTP time to jump around (forward and
>backwards); this could cause all sorts of nasty effects depending on
>the client.
=20
I agree, the middlebox could cause the the NTP time to jump around. The
semi-trusted middlebox is however trusted to forward the packets without
modifing them.
=20
>>We thought about protecting RTCP e2e but felt that it creates more=20
>>problems than it solves. Parts of RTCP should be sent e2e and other=20
>>parts only hbh, for RTCP APP, this distinction is application=20
>>dependant. If there is a need to protect against a synchronization=20
>>messing middlebox, the audio and video should encoded in a format that

>>can be sent in a single stream.
=20
>You can do that, but then you lose the ability to leverage all the
tools
>that understand audio/video streams; it no longer is an extension to
>existing abilities, but a new, separate tool for messages.
=20
>The question is: what is the reason(s) for SRTP usage, and how does
that
>map to the usecase you're trying to handle here (storing messages)?
Who
>are you protecting against; what level of interference needs to be
overcome?
=20
The main thing that we are protecting is the confidentiality of the
data. If you look at the use cases described in the drafts, I think you
will agree that in these use cases it is reasonable to assume that the
middleboxes are what we call semi-trusted. If one absolutely needs to
use a middlebox which is not trusted to not modify the synchronization
info and if modification of the synchronization info is seen as an
unacceptable risk, one might be limited to a smaller set of formats and
tools.

/ John Mattsson

From tom.taylor@rogers.com  Fri Apr  3 15:54:56 2009
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Subject: Re: [AVT] Inconsistent attribute name in RFC 3611
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Maybe I'm being hasty, but the general opinion seems to be to go with rcvr-rtt.

David R Oran wrote:
> I just checked and the Cisco IPTV code uses rcvr-rtt.
> 
> Just goes to show that for implementers they copy stuff from the 
> examples rather than the spec!
> 
> So, we'd obviously prefer that the errata change the IANA registration 
> and the spec text rather than the example, but if given some time to 
> switch (it takes 2 release to make such a change since this is a 
> red/green style packet problem) we can deal with either.
> 
> DaveO.
> 
> 
>>>> -----Original Message-----
>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
>>> Behalf Of
>>>> Tom Taylor
>>>> Sent: Thursday, April 02, 2009 7:16 PM
>>>> To: IETF AVT WG
>>>> Subject: [AVT] Inconsistent attribute name in RFC 3611
>>>>
>>>> RFC 3611 defines the attribute rcvr-rtt in the SDP section,
>>> but uses
>>>> recv-rtt in the text and the IANA Considerations section.
>>> Alan Clark
>>>> suggests that the SDP section should prevail, but it seems
>>> like a good
>>>> idea to check on what has been implemented first. Do people
>>> have views
>>>> one way or the other?
>>>>
>>>> Tom Taylor
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/avt
>>>
>>>
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
> 

From fluffy@cisco.com  Sat Apr  4 04:45:53 2009
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Cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] erratas on RFC 5117 (RTP Topologies)
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On Apr 3, 2009, at 7:07 AM, Magnus Westerlund wrote:

> Hi,
>
> I am looking on the filed Erratas for the RTP topologies (RFC 5117).
>
> I think all fall into the category of "Hold for document update". I
> would recommend that Cullen verify them into this state.
>
> Cheers
>
> Magnus

Seemed clear for these so I did as Magnus suggested.


From ron.even.tlv@gmail.com  Mon Apr  6 10:10:21 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Subject: [AVT] comments on draft-schmidt-avt-rfc3016bis-01
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Hi,

I reviewed the draft and have some comments.

 

It looks to me that the draft has all the text from RFC 3016 and I suggest
that it would obsolete RFC 3016 and not update it.

 

Others

 

1.       We are using media subtype for RTP payload instead of MIME type,
this need to be updated in the whole draft

2.       You are using the term VOP starting in section 1.1, please define
what is this abbreviation of.

3.       In section 1.2 you discuss scalable stream. In the second line of
the 4 paragraph you are referring to RTP packets but from the rest of the
paragraph it looks like you meant RTP streams (with different PT)

4.       For the scalable audio and video you are using the PT values to
define the dependency. I assume that you need it for backward compatibility
but why not also suggest using the SDP decoding dependency draft.

5.       In section 3 you mention RFC 4629 and 4628. Please add references.

6.       The media subtype registration should be updated to be inline with
RFC485

7.       For mpeg4 -video you mention in Applications which use this media
type IM and email yet there is no file format specified or referenced here.

8.       To the SDP usage add offer/answer consideration including also the
negotiation for profile-level, what is considered an interoperability mode,
do both sides need to have the same profile and level or the offer is
rejected and so on.

9.       The security section can you the security consideration from
draft-ieft-avt-srtp-not-mandatory

10.   The IANA consideration should say that it updates the media type
registration from 3016 and point at the media subtype definition sections in
the current draft.

 

 

Regards

Roni Even

 


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<div class=3DSection1>

<p class=3DMsoNormal>Hi,<o:p></o:p></p>

<p class=3DMsoNormal>I reviewed the draft and have some =
comments.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>It looks to me that the draft has all the text from =
RFC 3016
and I suggest that it would obsolete RFC 3016 and not update =
it.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Others<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>We are using media =
subtype for RTP
payload instead of MIME type, this need to be updated in the whole =
draft<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>You are using the term =
VOP
starting in section 1.1, please define what is this abbreviation =
of.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 1.2 you =
discuss scalable
stream. In the second line of the 4 paragraph you are referring to RTP =
packets
but from the rest of the paragraph it looks like you meant RTP streams =
(with different
PT)<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>For the scalable audio =
and video
you are using the PT values to define the dependency. I assume that you =
need it
for backward compatibility but why not also suggest using the SDP =
decoding dependency
draft.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>5.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 3 you mention =
RFC 4629
and 4628. Please add references.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>6.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>The media subtype =
registration
should be updated to be inline with RFC485<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>7.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>For mpeg4 &#8211;video =
you mention
in Applications which use this media type IM and email yet there is no =
file format
specified or referenced here.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>8.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>To the SDP usage add =
offer/answer
consideration including also the negotiation for profile-level, what is
considered an interoperability mode,&nbsp; do both sides need to have =
the same
profile and level or the offer is rejected and so on.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>9.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>The security section can =
you the
security consideration from =
draft-ieft-avt-srtp-not-mandatory<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>10.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>The IANA consideration =
should say
that it updates the media type registration from 3016 and point at the =
media
subtype definition sections in the current draft.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards<o:p></o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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From: Eric Rescorla <ekr@networkresonance.com>
To: avt@ietf.org, draft-mattsson-srtp-store-and-forward@tools.ietf.org
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Subject: [AVT] Review of store and forward drafts
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$Id: draft-mattson-srtp-store-and-forward-02-rev.txt,v 1.1 2009/04/06 22:43:10 ekr Exp $

Also draft-naslund-srtp-saf-00.txt

These documents describe a set of use cases and a protocol for a
modification to SRTP to do "store and forward". I have several
concerns:

- Are these use cases actually compelling?
- Do these use cases all need the same set of mechanisms?
- How is key management really going to be done for these.
- How realistic is the threat model?

The use cases here fall into two major categories:

- media distribution
- recording media on answering machines/voicemail servers
- untrusted conference bridges

It's not clear to me that these are really that similar problems.  In
particular, the media distribution cases mostly seem to me to be cases
in which what's needed is to temporally decouple message integrity
from confidentiality so that you can avoid storing bogus data which is
only determined to be bogus when you go to replay it. This is
particularly noticeable in the "Recording Encrypted Media at Home"
example (S 4.2.4) where there's no real need to treat the DVR as
untrusted--you just want to be sure that the media is correct at the
time you record it even though you don't have a license for the media
yet.

I agree that the voice mail case requires something more substantial,
however it's also much less compelling. First, it's not clear to me
that people really don't trust their voice mail servers. Second, voice
mail servers do a fair amount of mixing (for instance, prompts) with
the media. As I said on the call, I'm skeptical that existing phones
will handle this well. Third, it requires a key management scheme
we don't have: in order for this use case to work, you need a way
for the sender of the voice mail to do offline key exchange with
the ultimate receiver. If this worked well, we wouldn't have had 
any need for RTPSEC to solve the special case of online key exchange!

The untrusted conference bridge case seems even less compelling, 
because real bridges do a lot of fancy audio stuff.

Another concern I have is that the threat model seems fairly
unrealistic. Why is it safe to assume that the middlebox which
is not allowed to see the media is nevertheless allowed to retime
it and mount generic cut-and-paste attacks? I appreciate you
have a countermeasure for that, but that countermeasure relies
on a separate integrity check and basically precludes much of
the point of this work, it seems to me.

At the end of the day, I wonder whether there is a simpler scheme in
which we simply partition the authentication and encryption
transforms so that disconnected keys can be used for each. ISTM
that this would allow the most interesting set of use cases
(those concerned with media distribution) with significantly
less effort. Yes, I realize that it wouldn't provide quite
the same level of integrity protection, but it's not clear
that that's really necessary. To the extent to which you *do*
want to double-wrap the data, it's not clear to me that doing
it again in SRTP is really that useful: a lot of the point of
SRTP is that it leverages existing headers for maximum data
compactness. Once you're doublewrapping and adding new headers
like e2e PUV, maybe you could just get away with an outer
DTLS wrapper without any SRTP extensions









 








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From: Eric Rescorla <ekr@networkresonance.com>
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$Id: draft-mattson-srtp-store-and-forward-02-rev.txt,v 1.1 2009/04/06 22:43:10 ekr Exp $

Also draft-naslund-srtp-saf-00.txt

These documents describe a set of use cases and a protocol for a
modification to SRTP to do "store and forward". I have several
concerns:

- Are these use cases actually compelling?
- Do these use cases all need the same set of mechanisms?
- How is key management really going to be done for these.
- How realistic is the threat model?

The use cases here fall into two major categories:

- media distribution
- recording media on answering machines/voicemail servers
- untrusted conference bridges

It's not clear to me that these are really that similar problems.  In
particular, the media distribution cases mostly seem to me to be cases
in which what's needed is to temporally decouple message integrity
from confidentiality so that you can avoid storing bogus data which is
only determined to be bogus when you go to replay it. This is
particularly noticeable in the "Recording Encrypted Media at Home"
example (S 4.2.4) where there's no real need to treat the DVR as
untrusted--you just want to be sure that the media is correct at the
time you record it even though you don't have a license for the media
yet.

I agree that the voice mail case requires something more substantial,
however it's also much less compelling. First, it's not clear to me
that people really don't trust their voice mail servers. Second, voice
mail servers do a fair amount of mixing (for instance, prompts) with
the media. As I said on the call, I'm skeptical that existing phones
will handle this well. Third, it requires a key management scheme
we don't have: in order for this use case to work, you need a way
for the sender of the voice mail to do offline key exchange with
the ultimate receiver. If this worked well, we wouldn't have had 
any need for RTPSEC to solve the special case of online key exchange!

The untrusted conference bridge case seems even less compelling, 
because real bridges do a lot of fancy audio stuff.

Another concern I have is that the threat model seems fairly
unrealistic. Why is it safe to assume that the middlebox which
is not allowed to see the media is nevertheless allowed to retime
it and mount generic cut-and-paste attacks? I appreciate you
have a countermeasure for that, but that countermeasure relies
on a separate integrity check and basically precludes much of
the point of this work, it seems to me.

At the end of the day, I wonder whether there is a simpler scheme in
which we simply partition the authentication and encryption
transforms so that disconnected keys can be used for each. ISTM
that this would allow the most interesting set of use cases
(those concerned with media distribution) with significantly
less effort. Yes, I realize that it wouldn't provide quite
the same level of integrity protection, but it's not clear
that that's really necessary. To the extent to which you *do*
want to double-wrap the data, it's not clear to me that doing
it again in SRTP is really that useful: a lot of the point of
SRTP is that it leverages existing headers for maximum data
compactness. Once you're doublewrapping and adding new headers
like e2e PUV, maybe you could just get away with an outer
DTLS wrapper without any SRTP extensions









 








From ekr@networkresonance.com  Mon Apr  6 16:48:04 2009
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From: Eric Rescorla <ekr@networkresonance.com>
To: Rolf Blom <rolf.j.blom@ericsson.com>
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Subject: Re: [AVT] SRTP store-and-forward
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> When you refer to MIKEY-RSA having some mechanism to acquire such
> information (shared key / certificate) I believe you refer to
> MIKEY-RSA-R. In MIKEY-RSA-R the initiator attaches his certificate in
> the first message and the responder uses it to return the key to be
> used. This works well when there is an on-line connection between the
> two but&nbsp; wouldn't work for SRTP&nbsp; Store-and-Forward applications. Here
> we must rely on either predistributed keys or certificates or that keys
> or certificates can be acquired from an online third party. As I said
> before this situation is exactly as for secure email. And secure email
> is used!

I would say extraordinarily rarely. Indeed, this is precisely one
of the challenges that led to RTPSEC.

-Ekr


From rolf.j.blom@ericsson.com  Tue Apr  7 06:47:08 2009
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Subject: Re: [AVT] SRTP store-and-forward
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Hi Eric,<br>
<br>
My experience is that secure email becomes of more and more interest
for enterprises and government authorities. Such organizations already
now invest in the needed infrastructure to enable handling of user
certificates and other types of user credentials for secure access and
communication. I also expect that in the future, secure email will be
used more and more by the general public for privacy reasons. Look e.g.
at the success for PGP.<br>
<br>
/Rolf<br>
<br>
<br>
<br>
Eric Rescorla wrote:
<blockquote
 cite="mid:20090406234956.9616E1244C6@kilo.networkresonance.com"
 type="cite">
  <blockquote type="cite">
    <pre wrap="">When you refer to MIKEY-RSA having some mechanism to acquire such
information (shared key / certificate) I believe you refer to
MIKEY-RSA-R. In MIKEY-RSA-R the initiator attaches his certificate in
the first message and the responder uses it to return the key to be
used. This works well when there is an on-line connection between the
two but&amp;nbsp; wouldn't work for SRTP&amp;nbsp; Store-and-Forward applications. Here
we must rely on either predistributed keys or certificates or that keys
or certificates can be acquired from an online third party. As I said
before this situation is exactly as for secure email. And secure email
is used!
    </pre>
  </blockquote>
  <pre wrap=""><!---->
I would say extraordinarily rarely. Indeed, this is precisely one
of the challenges that led to RTPSEC.

-Ekr


  </pre>
</blockquote>
<br>
<pre class="moz-signature" cols="72">-- 
Rolf Blom, Ph.D. Expert, Mobile Communications Security, Ericsson Research
Postal address: Ericsson AB, SE-164 80 STOCKHOLM, Sweden
Tel: +46 10 713 17 07,  GSM: +46 70 757 2092,  Fax: +46 8 757 01 35


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From ekr@networkresonance.com  Tue Apr  7 08:46:47 2009
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At Tue, 07 Apr 2009 15:47:46 +0200,
Rolf Blom wrote:
> My experience is that secure email becomes of more and more interest
> for enterprises and government authorities. Such organizations already
> now invest in the needed infrastructure to enable handling of user
> certificates and other types of user credentials for secure access and
> communication.

My experience is the contrary. Some organizations try and they're
almost universally miserable.


> I also expect that in the future, secure email will be
> used more and more by the general public for privacy reasons. Look e.g.
> at the success for PGP.<br>

If that's a success, I'd hate to see failure. Compare the deployment of
TLS to the deployment of PGP.

-Ekr

From ron.even.tlv@gmail.com  Wed Apr  8 07:00:13 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Colin Perkins'" <csp@csperkins.org>, <Thomas.Schierl@hhi.fraunhofer.de>
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Hi,

There was support for having this work as a working group item, so AVT will
start working on this topic.

Colin or Thomas please submit the draft as an AVT draft.

Roni Even

 

From: Roni Even [mailto:ron.even.tlv@gmail.com] 
Sent: Sunday, March 29, 2009 11:14 PM
To: 'avt@ietf.org'
Cc: 'Tom Taylor'
Subject: making draft-perkins-avt-rapid-rtp-sync-03 a WG item

 

Hi,

I would like to ask if there is interest in making Rapid Synchronisation of
RTP Flows, draft-perkins-avt-rapid-rtp-sync-03.txt
<http://tools.ietf.org/id/draft-perkins-avt-rapid-rtp-sync-03.txt>   a
working group meeting. The draft was presented in San Francisco and is
offering a solutions to synchronize multiple RTP streams. It is a suggested
solution to DVB TM-AVC who would like to have by June.
This work is not in the AVT charter and the chairs intend to submit an
updated charter.
Please send your respond to the mailing list till April 8th . If there will
be no objections we will accept this topic and have the above draft as the
initial document
 
Roni Even
 

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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>There was support for =
having this
work as a working group item, so AVT will start working on this =
topic.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Colin or Thomas =
please submit
the draft as an AVT draft.<o:p></o:p></span></p>

<p class=3DMsoNormal><span style=3D'color:#1F497D'>Roni =
Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Roni Even
[mailto:ron.even.tlv@gmail.com] <br>
<b>Sent:</b> Sunday, March 29, 2009 11:14 PM<br>
<b>To:</b> 'avt@ietf.org'<br>
<b>Cc:</b> 'Tom Taylor'<br>
<b>Subject:</b> making draft-perkins-avt-rapid-rtp-sync-03 a WG =
item<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt'>Hi,<o:p></o:p></span></p>

<pre><span style=3D'font-family:"Calibri","sans-serif"'>I would like to =
ask if there is interest in making Rapid Synchronisation of RTP Flows, =
</span><a
href=3D"http://tools.ietf.org/id/draft-perkins-avt-rapid-rtp-sync-03.txt"=
>draft-perkins-avt-rapid-rtp-sync-03.txt</a> &nbsp;<span
style=3D'font-family:"Calibri","sans-serif"'>a working group meeting. =
The draft was presented in San Francisco and is offering a solutions to =
synchronize multiple RTP streams. It is a suggested solution to DVB =
TM-AVC who would like to have by June.<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Calibri","sans-serif"'>This work is not in the AVT =
charter and the chairs intend to submit an updated =
charter.<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Calibri","sans-serif"'>Please send your respond to =
the mailing list till April 8<sup>th</sup> . If there will be no =
objections we will accept this topic and have the above draft as the =
initial document<o:p></o:p></span></pre><pre><span
style=3D'font-family:"Calibri","sans-serif"'><o:p>&nbsp;</o:p></span></pr=
e><pre><span
style=3D'font-family:"Calibri","sans-serif"'>Roni =
Even<o:p></o:p></span></pre><pre><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></pre></div>

</body>

</html>

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From web-usrn@ISI.EDU  Tue Apr  7 11:00:28 2009
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Subject: [AVT] [Technical Errata Reported] RFC3611 (1759)
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The following errata report has been submitted for RFC3611,
"RTP Control Protocol Extended Reports (RTCP XR)".

--------------------------------------
You may review the report below and at:
http://www.rfc-editor.org/errata_search.php?rfc=3611&eid=1759

--------------------------------------
Type: Technical
Reported by: Timur Friedman <timur.friedman@lip6.fr>

Section: 6.3

Original Text
-------------
"recv-rtt"

Corrected Text
--------------
"rcvr-rtt"

Notes
-----
Sec. 6.3 describes the SDP attribute in Sec. 5.1, but erroneously calls it "recv-rtt" whereas it is in fact "rcvr-rtt". The IANA, following Sec. 6.3, had recorded "recv-rtt" but has corrected this and now records "rcvr-rtt". I am one of the editors of this RFC, and I apologize for not having caught this error before publication.

Instructions:
-------------
This errata is currently posted as "Reported". If necessary, please
use "Reply All" to discuss whether it should be verified or
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can log in to change the status and edit the report, if necessary. 

--------------------------------------
RFC3611 (draft-ietf-avt-rtcp-report-extns-06)
--------------------------------------
Title               : RTP Control Protocol Extended Reports (RTCP XR)
Publication Date    : November 2003
Author(s)           : T. Friedman, Ed., R. Caceres, Ed., A. Clark, Ed.
Category            : PROPOSED STANDARD
Source              : Audio/Video Transport
Area                : Real-time Applications and Infrastructure
Stream              : IETF
Verifying Party     : IESG

From ingemar.s.johansson@ericsson.com  Wed Apr  8 13:49:54 2009
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Hi

A new draft that addresses the work on Rapid Acquisition for Multicast
based Streaming has been submitted=20
http://tools.ietf.org/id/draft-johansson-avt-mcast-based-rams-00.txt

Comments are welcome=20

Regards
Ingemar Johansson
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D

Abstract:
This document proposes an improvement to the unicast based=20
Rapid Acquisition for Multicast based Streaming discussed in=20
[ID-Versteeg].
The outline of the improvement is to gather up Rapid=20
Acquisition requests for many users and transmit them in=20
dedicated multicast streams.  With this technique the peak=20
load on the retransmission server and on the outgoing link=20
from the retransmission server can be reduced.  For a problem=20
description of the channel change problem in multicast based=20
IPTV the reader is encouraged to read [ID-Versteeg].

[ID-Versteeg] =3D
http://tools.ietf.org/id/draft-versteeg-avt-rapid-synchronization-for-rt
p-02.txt

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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		: The SEED Cipher Algorithm and Its Use with the Secure Real-time Transport Protocol (SRTP
	Author(s)	: S. Yoon, H. Park, J. Kim, H. Jeong, Y. Won
	Filename	: draft-ietf-avt-seed-srtp-10.txt
	Pages		: 12
	Date		: 2009-4-8
	
This document describes the use of the SEED block cipher algorithm in 
   the Secure Real-time Transport Protocol (SRTP) for providing 
   confidentiality for the Real-time Transport Protocol (RTP) traffic 
   and for the control traffic for RTP, the Real-time Transport Control 
   Protocol (RTCP).

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From csp@csperkins.org  Wed Apr  8 14:41:11 2009
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From: Colin Perkins <csp@csperkins.org>
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On 8 Apr 2009, at 14:56, Roni Even wrote:
> There was support for having this work as a working group item, so  
> AVT will start working on this topic.
> Colin or Thomas please submit the draft as an AVT draft.

I submitted draft-ietf-avt-rapid-rtp-sync-00.txt - just waiting for WG  
chair approval. This is identical to draft-perkins-avt-rapid-rtp- 
sync-03.txt apart from the filename and date.

-- 
Colin Perkins
http://csperkins.org/




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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           : Rapid Synchronisation of RTP Flows
	Author(s)       : C. Perkins, T. Schierl
	Filename        : draft-ietf-avt-rapid-rtp-sync-00.txt
	Pages           : 17
	Date            : 2009-04-08

This memo outlines how RTP multimedia sessions are synchronised, and
discusses how rapidly such synchronisation can occur.  We show that
most RTP sessions can be synchronised immediately, but that the use
of video switching multipoint conference units (MCUs) or large source
specific multicast (SSM) groups can greatly increase the initial
synchronisation delay.  This increase in delay can be unacceptable to
some applications that use layered and/or multi-description codecs.

This memo updates the RTP Control Protocol (RTCP) timing rules to
reduce the initial synchronisation delay for SSM sessions.  A new
feedback packet is defined for use with the Extended RTP Profile for
RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to
rapidly request resynchronisation.  Two new RTP header extensions are
defined to allow rapid synchronisation of late joiners, and guarantee
correct timestamp based decoding order recovery for layered codecs in
the presence of clock skew.

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From john.mattsson@ericsson.com  Wed Apr  8 15:39:45 2009
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Hi Eric,

=20

Thanks very much for providing these comments; it is much appreciated to =
have this analysis of the problems and solutions in the draft.

=20

> These documents describe a set of use cases and a protocol for a=20

> modification to SRTP to do "store and forward". I have several

> concerns:

=20

> - Are these use cases actually compelling?

=20

The use cases are motivated by applications used and needed by the =
intended primary users and their major concern is data confidentiality. =
Given that, the use cases are real and compelling. In media distribution =
use cases, the confidentiality requirement is asymmetric; it (mainly) =
comes from the content owner. In the other use cases the requirement is =
symmetric and comes from both sender and receiver. Typical users in the =
answering machine and conferencing use cases are as stated in the draft =
organizations such as public safety and enterprises. Or for that matter =
anyone that uses transport protection and wants a similar level of =
protection in the store-and-forward case.

=20

> - Do these use cases all need the same set of mechanisms?

=20

Maybe not, but the draft proposes a common solution. And if all use =
cases can be easily solved with a single set of mechanisms, there seems =
to be no need for several solutions.

=20

> - How is key management really going to be done for these.

> - How realistic is the threat model?

=20

These questions are answered below.

=20

> The use cases here fall into two major categories:

=20

> - media distribution

> - recording media on answering machines/voicemail servers

> - untrusted conference bridges

=20

> It's not clear to me that these are really that similar problems.  In=20

> particular, the media distribution cases mostly seem to me to be cases

> in which what's needed is to temporally decouple message integrity

> from confidentiality so that you can avoid storing bogus data which is

> only determined to be bogus when you go to replay it. This is=20

> particularly noticeable in the "Recording Encrypted Media at Home"

> example (S 4.2.4) where there's no real need to treat the DVR as=20

> untrusted--you just want to be sure that the media is correct at the=20

> time you record it even though you don't have a license for the media

> yet.

=20

You are right in that the problems are a bit different. In this use case =
it might be sufficient to separate the keys. However, the solution then =
becomes severely limited in that the media cannot be pre-encrypted and =
the stored media cannot be forwarded. To get away from such limitations, =
a more important requirement in the media distribution use cases is to =
decouple the encryption from the RTP header to make the protection =
transport independent (See Section 5 in draft-mattsson). When it comes =
to trusting the DVR in the use case you mention we agree that the owner =
of the DVR would as you write probably trust it. However, the major =
security requirement in all use cases is to protect the confidentiality =
of the data and a content owner would have (very) limited trust in an =
arbitrary DVR.

=20

While we initially say that the trust model does not need to be =
symmetric in that both sender and receiver needs to have the same level =
of trust in the middlebox, we have probably been sloppy in pointing out =
this difference in a few places (like here, for instance). Thanks for =
identifying this issue.

=20

> I agree that the voice mail case requires something more substantial,=20

> however it's also much less compelling. First, it's not clear to me=20

> that people really don't trust their voice mail servers. Second, voice

> mail servers do a fair amount of mixing (for instance, prompts) with=20

> the media. As I said on the call, I'm skeptical that existing phones=20

> will handle this well. Third, it requires a key management scheme we=20

> don't have: in order for this use case to work, you need a way for the

> sender of the voice mail to do offline key exchange with the ultimate=20

> receiver. If this worked well, we wouldn't have had any need for=20

> RTPSEC to solve the special case of online key exchange!

=20

Well, in the general case with two random peers without any relation, =
the key management has no general solution today, and it is probably =
such peers that you think of when you write that the use case is not =
compelling. The typical users we have in mind, public-safety and =
enterprise users, would normally have key management systems already. =
Note that the de-coupling of the key management and SRTP SaF was a =
deliberate choice to allow use of different existing and/or future key =
management solutions. The fact that a general key management scheme does =
not exist is to me not an argument to stop looking at the use case, it =
is rather an argument to start looking also at the key management =
problem. In Appendix B in draft-mattsson we illustrate how the key =
management can be achieved with small changes to DTLS-SRTP or MIKEY, but =
some work surely needs to be done. Finally, you can of course not mix =
encrypted data, but the needed functionality for prompting etc can be =
achieved by switching of security contexts within one stream or simply =
sending the prompts in a separate stream.

=20

> The untrusted conference bridge case seems even less compelling,=20

> because real bridges do a lot of fancy audio stuff.

=20

One example where no mixing is done is Push-to-talk, which is a major =
use case for the expected users such as public safety and enterprises. =
Ordinary conference bridges would need some type of floor control.

=20

> Another concern I have is that the threat model seems fairly=20

> unrealistic. Why is it safe to assume that the middlebox which is not=20

> allowed to see the media is nevertheless allowed to retime it and=20

> mount generic cut-and-paste attacks? I appreciate you have a=20

> countermeasure for that, but that countermeasure relies on a separate=20

> integrity check and basically precludes much of the point of this=20

> work, it seems to me.

=20

The trust in the middlebox may sometimes be very high (as you yourself =
point out above, e.g. the end user likely has strong trust in his own =
PVR). In other cases, it may be weak(er). We looked for a single trust =
model that is reasonable (neither too strong nor too weak) to apply to =
all our use cases. The main concern (from end-user's point of view) in =
all our uses cases is the confidentiality of sensitive/valuable data. We =
claim that from this (confidentiality) point of view it _is_ safe to use =
the proposed solution in the defined trust model. The middlebox is =
trusted to deliver media as requested, nothing else, and to this end we =
used the well-known "honest-by-curious" model. But this does not imply =
that the middlebox is "allowed" to do the copy-paste attacks and similar =
things. The deliberately malicious middlebox threat is in fact not =
realistic in the intended operational environment. Nevertheless, we =
always leave control of data confidentiality to the endpoints.

=20

In summary:

*	In the media distribution use cases, the main concern of the content =
creator is probably not what a DVR does with the media as long as it =
cannot access cleartext data. In the same case, the owner of the DVR =
would trust it.
*	In the answering machine and conferencing use cases, there is no =
realistic case were the owner of the service (e.g. an operator) would =
have something to gain by launching such copy-paste attacks in general. =
(Note that such attacks would anyhow be much harder to do in a =
controlled way, when the data is e2e encrypted.) The privacy of the data =
is the fundamental requirement for public safety and enterprise users.
*	The trust model has been chosen to be simple but at the same time it =
also "allows" some attacks which are not realistic (but they are still =
detectable).

=20

When we update the draft, we will include a section about copy-and-paste =
attacks in the security considerations.

=20

> At the end of the day, I wonder whether there is a simpler scheme in=20

> which we simply partition the authentication and encryption transforms

> so that disconnected keys can be used for each. ISTM that this would=20

> allow the most interesting set of use cases (those concerned with=20

> media distribution) with significantly less effort. Yes, I realize=20

> that it wouldn't provide quite the same level of integrity protection,

> but it's not clear that that's really necessary. To the extent to=20

> which you *do* want to double-wrap the data, it's not clear to me that

> doing it again in SRTP is really that useful: a lot of the point of=20

> SRTP is that it leverages existing headers for maximum data=20

> compactness. Once you're doublewrapping and adding new headers like=20

> e2e PUV, maybe you could just get away with an outer DTLS wrapper=20

> without any SRTP extensions

=20

Key separation is a very nice and simple alternative that would work in =
some of the uses cases. Unfortunately, it has several shortcomings. It =
does not work well in the media distribution use cases as rewind and =
random seek functionality would be impossible. For that functionality to =
be possible, the protection has to be transport independent. To solve =
this, we introduced the PUV. To identify the SA we needed the CCI, and =
to protect several streams with a single SA, we added the SSS. We did =
consider alternatives, but the requirements in the identified use cases =
cannot be solved with any "simpler" changes to SRTP.

=20

/John


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Subject: [AVT] RFC 5506 on Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences
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A new Request for Comments is now available in online RFC libraries.

        
        RFC 5506

        Title:      Support for Reduced-Size Real-Time Transport 
                    Control Protocol (RTCP): Opportunities and
                    Consequences 
        Author:     I. Johansson, M. Westerlund
        Status:     Standards Track
        Date:       April 2009
        Mailbox:    ingemar.s.johansson@ericsson.com, 
                    magnus.westerlund@ericsson.com
        Pages:      17
        Characters: 41011
        Updates:    RFC3550, RFC3711, RFC4585

        I-D Tag:    draft-ietf-avt-rtcp-non-compound-09.txt

        URL:        http://www.rfc-editor.org/rfc/rfc5506.txt

This memo discusses benefits and issues that arise when allowing
Real-time Transport Protocol (RTCP) packets to be transmitted with
reduced size.  The size can be reduced if the rules on how to create
compound packets outlined in RFC 3550 are removed or changed.  Based
on that analysis, this memo defines certain changes to the rules to
allow feedback messages to be sent as Reduced-Size RTCP packets under
certain conditions when using the RTP/AVPF (Real-time Transport
Protocol / Audio-Visual Profile with Feedback) profile (RFC 4585).
This document updates RFC 3550, RFC 3711, and RFC 4585.  [STANDARDS TRACK]

This document is a product of the Audio/Video Transport Working Group of the IETF.

This is now a Proposed Standard Protocol.

STANDARDS TRACK: This document specifies an Internet standards track
protocol for the Internet community,and requests discussion and suggestions
for improvements.  Please refer to the current edition of the Internet
Official Protocol Standards (STD 1) for the standardization state and
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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 Format for ITU-T Recommendation G.722.1
	Author(s)       : P. Luthi, R. Even
	Filename        : draft-ietf-avt-rfc3047-bis-09.txt
	Pages           : 17
	Date            : 2009-04-11

International Telecommunication Union (ITU-T) Recommendation G.722.1
is a wide-band audio codec.  This document describes the payload
format for including G.722.1 generated bit streams within an RTP
packet.  The document also describes the syntax and semantics of the
SDP parameters needed to support G.722.1 audio codec.

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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 Format for the Speex Codec
	Author(s)       : G. Herlein, et al.
	Filename        : draft-ietf-avt-rtp-speex-06.txt
	Pages           : 23
	Date            : 2009-04-13

Speex is an open-source voice codec suitable for use in Voice over IP
(VoIP) type applications.  This document describes the payload format
for Speex generated bit streams within an RTP packet.  Also included
here are the necessary details for the use of Speex with the Session
Description Protocol (SDP).Editors Note

All references to RFC XXXX are to be replaced by references to the
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Colin Perkins'" <csp@csperkins.org>, <Thomas.Schierl@hhi.fraunhofer.de>
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Subject: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00
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Hi guys,

I think Thomas had a request to move this draft forward. 

I suggest that you do another revision finishing the missing parts like the
IANA consideration section. 

Please use fmt=5 for the message since this is the next available one in the
IANA registry and the other work an rapid acquisition will use the following
numbers when if and when it becomes a working group document.

 

I noticed that there was a lot of support for this work, so all those who
support the work please help by reviewing the work so we can start WGLC ASAP

 

Roni Even

 

 

 


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<p class=3DMsoNormal>Hi guys,<o:p></o:p></p>

<p class=3DMsoNormal>I think Thomas had a request to move this draft =
forward. <o:p></o:p></p>

<p class=3DMsoNormal>I suggest that you do another revision finishing =
the missing
parts like the IANA consideration section. <o:p></o:p></p>

<p class=3DMsoNormal>Please use fmt=3D5 for the message since this is =
the next
available one in the IANA registry and the other work an rapid =
acquisition will
use the following numbers when if and when it becomes a working group =
document.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I noticed that there was a lot of support for this =
work, so
all those who support the work please help by reviewing the work so we =
can
start WGLC ASAP<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

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From ron.even.tlv@gmail.com  Tue Apr 14 03:09:43 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Zeev Vax'" <zeevvax@microsoft.com>, "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>, "'Peilin YANG'" <yangpeilin@huawei.com>
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Subject: [AVT] FMT numbers in your draft
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Hi guys,

Since we have the RTP rapid synchronization as a WG item I suggest they use
the fmt=5 and that for new RTPFB that will be specified in any new work will
use the following numbers when it is a WG item

 

Regards

Roni Even

AVT co-chair 


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<p class=3DMsoNormal>Hi guys,<o:p></o:p></p>

<p class=3DMsoNormal>Since we have the RTP rapid synchronization as a WG =
item I
suggest they use the fmt=3D5 and that for new RTPFB that will be =
specified in any
new work will use the following numbers when it is a WG =
item<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Regards<o:p></o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

<p class=3DMsoNormal>AVT co-chair <o:p></o:p></p>

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Subject: Re: [AVT] FMT numbers in your draft
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Sure, we will make the necessary changes in the revision. We are also =
working on a format that will need only one FMT number for the protocol, =
and then we will use a field something like "sub fmt" to further specify =
which specific message we are using.

Thanks for the heads-up.

-acbegen=20

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Tuesday, April 14, 2009 3:09 AM
> To: 'Zeev Vax'; Ali C. Begen (abegen); Bill Ver Steeg=20
> (versteb); 'Peilin YANG'
> Cc: avt@ietf.org; 'Tom Taylor'
> Subject: FMT numbers in your draft
>=20
> Hi guys,
>=20
> Since we have the RTP rapid synchronization as a WG item I=20
> suggest they use the fmt=3D5 and that for new RTPFB that will=20
> be specified in any new work will use the following numbers=20
> when it is a WG item
>=20
> =20
>=20
> Regards
>=20
> Roni Even
>=20
> AVT co-chair=20
>=20
>=20

From Thomas.Schierl@hhi.fraunhofer.de  Tue Apr 14 06:20:00 2009
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Subject: Re: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00
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Hi Roni, all,<br>
<br>
We will do so soon. I hope we can submit a new version of the draft
early next week.<br>
<br>
Thanks,<br>
Thomas<br>
<pre class="moz-signature" cols="72">______________________________________________
Thomas Schierl, Dipl.-Ing.
Fraunhofer HHI
Senior Research Engineer
Image Processing Dept. / Multimedia Transport
Einsteinufer 37, 10587 Berlin, Germany
PHONE: +49 30 31002 227 , FAX: +49 30 392 7200
WEB: <a class="moz-txt-link-freetext" href="http://ip.hhi.de">http://ip.hhi.de</a> , SKYPE: thomas.schierl
EMAIL: <a class="moz-txt-link-abbreviated" href="mailto:thomas.schierl@hhi.fraunhofer.de">thomas.schierl@hhi.fraunhofer.de</a></pre>
<br>
<br>
Roni Even wrote:
<blockquote cite="mid:49e46079.25e2660a.30b0.ffffb5f3@mx.google.com"
 type="cite">
  <meta http-equiv="Context-Type" content="text/html; charset=US-ASCII">
  <div>
  <p>Hi guys, </p>
  <p>I think Thomas had a request to move this draft forward. </p>
  <p>I suggest that you do another revision finishing the missing
parts like the IANA consideration section. </p>
  <p>Please use fmt=5 for the message since this is the next
available one in the IANA registry and the other work an rapid
acquisition will
use the following numbers when if and when it becomes a working group
document. </p>
  <p> &nbsp; </p>
  <p>I noticed that there was a lot of support for this work, so
all those who support the work please help by reviewing the work so we
can
start WGLC ASAP </p>
  <p> &nbsp; </p>
  <p>Roni Even </p>
  <p> &nbsp; </p>
  <p> &nbsp; </p>
  <p> &nbsp; </p>
  </div>
</blockquote>
<BR>
<br>
----<br>
Visit us at<br><br>

NAB SHOW 2009 / April 18-23 / Las Vegas, Nevada USA / South Hall (Upper), Booth SU9624F<br>
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From yekuiwang@huawei.com  Tue Apr 14 07:09:14 2009
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Date: Tue, 14 Apr 2009 09:58:06 -0400
From: Ye-Kui Wang <yekuiwang@huawei.com>
In-reply-to: <49e46079.25e2660a.30b0.ffffb5f3@mx.google.com>
To: 'Roni Even' <ron.even.tlv@gmail.com>, 'Colin Perkins' <csp@csperkins.org>,  Thomas.Schierl@hhi.fraunhofer.de
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Subject: Re: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00
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Colin said the following a few days ago, 
 
"I submitted draft-ietf-avt-rapid-rtp-sync-00.txt - just waiting for WG
chair approval. This is identical to draft-perkins-avt-rapid-rtp-
sync-03.txt apart from the filename and date."
 
Therefore, it seems that all the comments sent to the list before the San
Francisco meeting should be addressed first (and there were requests that
these should be addressed in the first WG version), before the "supporters"
draft to do another round of review.  
 
BR, YK


  _____  

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Roni
Even
Sent: Tuesday, April 14, 2009 6:06 AM
To: 'Colin Perkins'; Thomas.Schierl@hhi.fraunhofer.de
Cc: avt@ietf.org; avt-chairs@tools.ietf.org
Subject: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00



Hi guys,

I think Thomas had a request to move this draft forward. 

I suggest that you do another revision finishing the missing parts like the
IANA consideration section. 

Please use fmt=5 for the message since this is the next available one in the
IANA registry and the other work an rapid acquisition will use the following
numbers when if and when it becomes a working group document.

 

I noticed that there was a lot of support for this work, so all those who
support the work please help by reviewing the work so we can start WGLC ASAP

 

Roni Even

 

 

 


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<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>Colin said the following a few days 
ago,&nbsp;</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>"I submitted draft-ietf-avt-rapid-rtp-sync-00.txt - just 
waiting for WG chair approval. This is identical to draft-perkins-avt-rapid-rtp- 
sync-03.txt apart from the filename and date."</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>Therefore, it seems that all the comments sent to the list 
before the San Francisco meeting should be addressed first (and there were 
requests that these should be addressed in the first WG version), before the 
"supporters" draft to do another round of review. &nbsp;</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=ltr align=left><SPAN class=390225413-14042009><FONT face=&#23435;&#20307; 
color=#0000ff size=2>BR, YK</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=ltr 
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
  <DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
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  <FONT face=Tahoma size=2><B>From:</B> avt-bounces@ietf.org 
  [mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Roni Even<BR><B>Sent:</B> 
  Tuesday, April 14, 2009 6:06 AM<BR><B>To:</B> 'Colin Perkins'; 
  Thomas.Schierl@hhi.fraunhofer.de<BR><B>Cc:</B> avt@ietf.org; 
  avt-chairs@tools.ietf.org<BR><B>Subject:</B> [AVT] going forward with 
  draft-ietf-avt-rapid-rtp-sync-00<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV class=Section1>
  <P class=MsoNormal>Hi guys,<o:p></o:p></P>
  <P class=MsoNormal>I think Thomas had a request to move this draft forward. 
  <o:p></o:p></P>
  <P class=MsoNormal>I suggest that you do another revision finishing the 
  missing parts like the IANA consideration section. <o:p></o:p></P>
  <P class=MsoNormal>Please use fmt=5 for the message since this is the next 
  available one in the IANA registry and the other work an rapid acquisition 
  will use the following numbers when if and when it becomes a working group 
  document.<o:p></o:p></P>
  <P class=MsoNormal><o:p>&nbsp;</o:p></P>
  <P class=MsoNormal>I noticed that there was a lot of support for this work, so 
  all those who support the work please help by reviewing the work so we can 
  start WGLC ASAP<o:p></o:p></P>
  <P class=MsoNormal><o:p>&nbsp;</o:p></P>
  <P class=MsoNormal>Roni Even<o:p></o:p></P>
  <P class=MsoNormal><o:p>&nbsp;</o:p></P>
  <P class=MsoNormal><o:p>&nbsp;</o:p></P>
  <P class=MsoNormal><o:p>&nbsp;</o:p></P></DIV></BLOCKQUOTE></BODY></HTML>

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From Thomas.Schierl@hhi.fraunhofer.de  Tue Apr 14 07:19:09 2009
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Cc: 'Colin Perkins' <csp@csperkins.org>, avt-chairs@tools.ietf.org, avt@ietf.org
Subject: Re: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00
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Ye-Kui,<br>
<br>
This is the plan for the new version to be submitted early next week.<br>
<br>
Thanks,<br>
Thomas<br>
<pre class="moz-signature" cols="72">______________________________________________
Thomas Schierl, Dipl.-Ing.
Fraunhofer HHI
Senior Research Engineer
Image Processing Dept. / Multimedia Transport
Einsteinufer 37, 10587 Berlin, Germany
PHONE: +49 30 31002 227 , FAX: +49 30 392 7200
WEB: <a class="moz-txt-link-freetext" href="http://ip.hhi.de">http://ip.hhi.de</a> , SKYPE: thomas.schierl
EMAIL: <a class="moz-txt-link-abbreviated" href="mailto:thomas.schierl@hhi.fraunhofer.de">thomas.schierl@hhi.fraunhofer.de</a></pre>
<br>
<br>
Ye-Kui Wang wrote:
<blockquote cite="mid:B5D32C3224974AE7B838F1946352C513@china.huawei.com"
 type="cite">
  <meta http-equiv="Context-Type" content="text/html; charset=us-ascii">
  <div><span> Colin said the following a few days ago,&nbsp; </span></div>
  <div><span> </span>&nbsp;</div>
  <div><span> "I submitted draft-ietf-avt-rapid-rtp-sync-00.txt - just
waiting for WG chair approval. This is identical to
draft-perkins-avt-rapid-rtp- sync-03.txt apart from the filename and
date." </span></div>
  <div><span> </span>&nbsp;</div>
  <div><span> Therefore, it seems that all the comments sent to the
list before the San Francisco meeting should be addressed first (and
there were requests that these should be addressed in the first WG
version), before the "supporters" draft to do another round of review.
&nbsp; </span></div>
  <div><span> </span>&nbsp;</div>
  <div><span> BR, YK </span></div>
  <br>
  <blockquote>
    <div lang="en-us"> <b>From:</b> <a class="moz-txt-link-abbreviated" href="mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a>
[<a class="moz-txt-link-freetext" href="mailto:avt-bounces@ietf.org">mailto:avt-bounces@ietf.org</a>] <b>On Behalf Of </b>Roni Even<br>
    <b>Sent:</b> Tuesday, April 14, 2009 6:06 AM<br>
    <b>To:</b> 'Colin Perkins'; <a class="moz-txt-link-abbreviated" href="mailto:Thomas.Schierl@hhi.fraunhofer.de">Thomas.Schierl@hhi.fraunhofer.de</a><br>
    <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:avt@ietf.org">avt@ietf.org</a>; <a class="moz-txt-link-abbreviated" href="mailto:avt-chairs@tools.ietf.org">avt-chairs@tools.ietf.org</a><br>
    <b>Subject:</b> [AVT] going forward with
draft-ietf-avt-rapid-rtp-sync-00<br>
    <br>
    </div>
    <div>
    <p>Hi guys, </p>
    <p>I think Thomas had a request to move this draft forward. </p>
    <p>I suggest that you do another revision finishing the missing
parts like the IANA consideration section. </p>
    <p>Please use fmt=5 for the message since this is the next
available one in the IANA registry and the other work an rapid
acquisition will use the following numbers when if and when it becomes
a working group document. </p>
    <p> &nbsp; </p>
    <p>I noticed that there was a lot of support for this work, so all
those who support the work please help by reviewing the work so we can
start WGLC ASAP </p>
    <p> &nbsp; </p>
    <p>Roni Even </p>
    <p> &nbsp; </p>
    <p> &nbsp; </p>
    <p> &nbsp; </p>
    </div>
  </blockquote>
</blockquote>
<BR>
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From petithug@acm.org  Tue Apr 14 08:42:55 2009
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Subject: [AVT] RFC 4103 - RTP Payload for Text Conversation
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I hope that this is the right place for questions on RFC 4103.

- Section 5.2, 3rd paragraph:  "The last packet before an idle
period will contain only one non-empty T140block as redundant data..."

I am not sure how to reconcile this with the first paragraph in the
same section that states that the beginning of an idle period is
when the empty T140block is sent.  Does it means that the idle
period starts with the empty T140block when no redundancy is used,
but with the last retransmitted packet when redundancy is used?

This also does not fit well with the next sentence; "Any empty
T140block sent as primary data MUST be included as redundant
T140block in subsequent packets, just as normal text."  If it was
the case, wouldn't the last redundant packet sent entirely be
composed of empty T140block?

- Section 7.1: "As a follow-on to the previous example, the example
belows shows the next RTP packet in the sequence..."

The example that follow does not seems to be correct.  The previous
packet contains an empty primary block and data in the secondary
block.  The next packet contains 3 blocks so I understand that the
third block in the previous packet was removed because it contained
an empty block that has a too old timestamp.  So the previous packet
contains (older first): {empty, data, empty}, but the next packet
contains {empty, data, data}, which does not make sense.  If I
understand correctly the protocol, it should be {data, empty, data}.
 Is my understanding correct?

- T-140 says that the byte order mark should be inserted at the
beginning of the session, which I understand as the first packet
sent (as RFC 4103 does not say anything about it).  But the byte
order mark is useless after been converted to UTF-8 so why a sender
should send it?


Thanks.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

From md.1@newtbt.com  Tue Apr 14 09:02:16 2009
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Subject: Re: [AVT] going forward with draft-ietf-avt-rapid-rtp-sync-00
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Roni-

My comments from the earlier draft have not been addressed, but I see 
the authors are preparing another revision.

But before I continued the thread with the authors, I had asked you 
your intentions for modifying the charter.  The new ID is now listed, 
but there is no discussion.  It would be helpful to understand the 
scope of this work and if this is intended to be a general and 
complete guidance for rapid sync between RTP sessions, or just one 
way to improve sync in some cases?

Thanks,

         Mike

At 01:06 PM 4/14/2009 +0300, Roni Even wrote:
>Content-Type: multipart/alternative;
>         boundary="----=_NextPart_000_0086_01C9BD01.D0454F70"
>Content-language: en-us
>
>Hi guys,
>I think Thomas had a request to move this draft forward.
>I suggest that you do another revision finishing the missing parts 
>like the IANA consideration section.
>Please use fmt=5 for the message since this is the next available 
>one in the IANA registry and the other work an rapid acquisition 
>will use the following numbers when if and when it becomes a working 
>group document.
>
>I noticed that there was a lot of support for this work, so all 
>those who support the work please help by reviewing the work so we 
>can start WGLC ASAP
>
>Roni Even
>
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www.ietf.org/mailman/listinfo/avt


From gunnar.hellstrom@omnitor.se  Tue Apr 14 10:44:13 2009
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To: "'Marc Petit-Huguenin'" <petithug@acm.org>, <avt@ietf.org>
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Marc,

I will start answering your questions in the AVT list.

In parallell, we can have a side discussion about details if you want.=20
I have guided some implementors through implementations and interop =
tests.

Until then, take a look at www.realtimetext.org
and sourceforge.net search for tipcon1=20

Regards

Gunnar=20


-------------------------------------------------------------------
Gunnar Hellstr=F6m
Omnitor
gunnar.hellstrom@omnitor.se
Tel: +46708204288
www.omnitor.se

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Marc
Petit-Huguenin
Sent: Tuesday, April 14, 2009 5:44 PM
To: avt@ietf.org
Subject: [AVT] RFC 4103 - RTP Payload for Text Conversation

I hope that this is the right place for questions on RFC 4103.

- Section 5.2, 3rd paragraph:  "The last packet before an idle period =
will
contain only one non-empty T140block as redundant data..."

I am not sure how to reconcile this with the first paragraph in the same
section that states that the beginning of an idle period is when the =
empty
T140block is sent.  Does it means that the idle period starts with the =
empty
T140block when no redundancy is used, but with the last retransmitted =
packet
when redundancy is used?

This also does not fit well with the next sentence; "Any empty T140block
sent as primary data MUST be included as redundant T140block in =
subsequent
packets, just as normal text."  If it was the case, wouldn't the last
redundant packet sent entirely be composed of empty T140block?

- Section 7.1: "As a follow-on to the previous example, the example =
belows
shows the next RTP packet in the sequence..."

The example that follow does not seems to be correct.  The previous =
packet
contains an empty primary block and data in the secondary block.  The =
next
packet contains 3 blocks so I understand that the third block in the
previous packet was removed because it contained an empty block that has =
a
too old timestamp.  So the previous packet contains (older first): =
{empty,
data, empty}, but the next packet contains {empty, data, data}, which =
does
not make sense.  If I understand correctly the protocol, it should be =
{data,
empty, data}.
 Is my understanding correct?

- T-140 says that the byte order mark should be inserted at the =
beginning of
the session, which I understand as the first packet sent (as RFC 4103 =
does
not say anything about it).  But the byte order mark is useless after =
been
converted to UTF-8 so why a sender should send it?


Thanks.

--
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

__________ NOD32 4006 (20090414) Information __________

Detta meddelande dr genomsvkt av NOD32 Antivirus.
http://www.nod32.com



From gunnar.hellstrom@omnitor.se  Tue Apr 14 13:52:27 2009
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Marc,
See answers inline.


-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Marc
Petit-Huguenin
Sent: Tuesday, April 14, 2009 5:44 PM
To: avt@ietf.org
Subject: [AVT] RFC 4103 - RTP Payload for Text Conversation

I hope that this is the right place for questions on RFC 4103.

- Section 5.2, 3rd paragraph:  "The last packet before an idle period will
contain only one non-empty T140block as redundant data..."

I am not sure how to reconcile this with the first paragraph in the same
section that states that the beginning of an idle period is when the empty
T140block is sent.  Does it means that the idle period starts with the empty
T140block when no redundancy is used, but with the last retransmitted packet
when redundancy is used?

<GH: Read te first paragraph of 5.2 as valid for the non-redundancy case. In
that case you transmit one packet with an empty T140 block at the sample
time after the last valid data. Then there will be an equal opportunity to
detect if valid data is missing regardless of if the data is within a text
burst or at the end of it. The idle period starts after transmission of that
packet.
Next multi-line paragraph is for the redundancy case.  Then you send in this
pattern: 
data1, data2, data3.   

data2, data3, empty.  

data3, empty, empty 

-----  idle period     ----           

empty, empty, data4    

This matches your assumption, and the text in the RFC.>


This also does not fit well with the next sentence; "Any empty T140block
sent as primary data MUST be included as redundant T140block in subsequent
packets, just as normal text."  If it was the case, wouldn't the last
redundant packet sent entirely be composed of empty T140block?
<GH: Yes, it fits. If the idle period is short, the latest transmitted empty
T140 block that has not been transmitted the full number of redundant
generations, will be transmitted again, stepped up in redundancy
generations, and followed by the new primary data. But if the idle period
was longer, the old empty blocks will be dropped. >

- Section 7.1: "As a follow-on to the previous example, the example belows
shows the next RTP packet in the sequence..."

The example that follow does not seems to be correct.  The previous packet
contains an empty primary block and data in the secondary block.  The next
packet contains 3 blocks so I understand that the third block in the
previous packet was removed because it contained an empty block that has a
too old timestamp.  So the previous packet contains (older first): {empty,
data, empty}, but the next packet contains {empty, data, data}, which does
not make sense.  If I understand correctly the protocol, it should be {data,
empty, data}.
 Is my understanding correct?
<GH: Yes, it seems that you found an error. The data block in the last
picture should be named "R2" instead of "R1" and in the text above the
picture, "The value of the "R2 block length"" should be "The value of the
"R1 block length"" 

- T-140 says that the byte order mark should be inserted at the beginning of
the session, which I understand as the first packet sent (as RFC 4103 does
not say anything about it).  But the byte order mark is useless after been
converted to UTF-8 so why a sender should send it?
<GH: One good reason for this is that some network equipment requires
outgoing packets in order to be able to receive incoming packets. So, one
reason for this transmission is to "prime the channel". 
Anoter reason is that you do not know what treatment the T.140 text gets
when it is received. It may be converted back to UCS-16 form, and then it is
very valid to have a byte order mark first in the text to verify that bytes
do not get pairwise swapped by mistake at some big endian - little endian
interface. So, therefore it is best to stick to what the standard says about
including it. 

There may even be a reason to keep sending byte order marks or empty t140
blocks at regular intervals if nothing else is transmitted, in order to keep
the RTP stream alive through NAT and firewalls. 

The keep-alive draft
http://tools.ietf.org/html/draft-ietf-avt-app-rtp-keepalive-05 
recommends use of empty t140blocks for this purpose, but if you need to
create the keep-alive packets from an application on top of an existing RTP
stack, it might not be possible to cause transmission of empty blocks, but
very well blocks with byte order mark.>

<GH: Please note that there is errata available for RFC 4103. It is just
swapping of the order of the payload types in the m-line of the second SDP
example in section 7.2, so that they come in the right priority order with
redundancy first and plain T140 second.
http://www.rfc-editor.org/errata_search.php?rfc=4103

>  

Thanks
Gunnar Hellstrom.
--------------------

 


Thanks.

--
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt

__________ NOD32 4006 (20090414) Information __________

Detta meddelande dr genomsvkt av NOD32 Antivirus.
http://www.nod32.com



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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [AVT] RFC 4103 - RTP Payload for Text Conversation
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Gunnar Hellstrom wrote:
> Marc,
> See answers inline.
> 
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Marc
> Petit-Huguenin
> Sent: Tuesday, April 14, 2009 5:44 PM
> To: avt@ietf.org
> Subject: [AVT] RFC 4103 - RTP Payload for Text Conversation
> 
> I hope that this is the right place for questions on RFC 4103.
> 
> - Section 5.2, 3rd paragraph:  "The last packet before an idle period will
> contain only one non-empty T140block as redundant data..."
> 
> I am not sure how to reconcile this with the first paragraph in the same
> section that states that the beginning of an idle period is when the empty
> T140block is sent.  Does it means that the idle period starts with the empty
> T140block when no redundancy is used, but with the last retransmitted packet
> when redundancy is used?
> 
> <GH: Read te first paragraph of 5.2 as valid for the non-redundancy case. In
> that case you transmit one packet with an empty T140 block at the sample
> time after the last valid data. Then there will be an equal opportunity to
> detect if valid data is missing regardless of if the data is within a text
> burst or at the end of it. The idle period starts after transmission of that
> packet.
> Next multi-line paragraph is for the redundancy case.  Then you send in this
> pattern: 
> data1, data2, data3.   
> 
> data2, data3, empty.  
> 
> data3, empty, empty 
> 
> -----  idle period     ----           
> 
> empty, empty, data4    
> 
> This matches your assumption, and the text in the RFC.>

So if I understand correctly the idle period starts 300 ms after
sending the last text when there is no redundancy, and 600 ms after
sending the last text in the primary block when using redundancy.

> 
> 
> This also does not fit well with the next sentence; "Any empty T140block
> sent as primary data MUST be included as redundant T140block in subsequent
> packets, just as normal text."  If it was the case, wouldn't the last
> redundant packet sent entirely be composed of empty T140block?
> <GH: Yes, it fits. If the idle period is short, the latest transmitted empty
> T140 block that has not been transmitted the full number of redundant
> generations, will be transmitted again, stepped up in redundancy
> generations, and followed by the new primary data. But if the idle period
> was longer, the old empty blocks will be dropped. >

OK.

> 
> - T-140 says that the byte order mark should be inserted at the beginning of
> the session, which I understand as the first packet sent (as RFC 4103 does
> not say anything about it).  But the byte order mark is useless after been
> converted to UTF-8 so why a sender should send it?
> <GH: One good reason for this is that some network equipment requires
> outgoing packets in order to be able to receive incoming packets. So, one
> reason for this transmission is to "prime the channel". 

Right.  NATs.


Thanks.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

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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 format for GSM-HR
	Author(s)       : X. Duan, et al.
	Filename        : draft-ietf-avt-rtp-gsm-hr-00.txt
	Pages           : 18
	Date            : 2009-04-15

This document specifies the RTP payload format for packetization of
the GSM Half-Rate speech codec.

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http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-gsm-hr-00.txt

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Subject: [AVT] Fwd: [Fecframe] WGLC on draft-ietf-fecframe-interleaved-fec-scheme
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Forwarding to AVT, since this is an RTP payload format.
Colin



Begin forwarded message:
> From: Marshall Eubanks <tme@multicasttech.com>
> Date: 1 April 2009 13:36:50 BST
> To: fecframe@ietf.org
> Subject: [Fecframe] WGLC on draft-ietf-fecframe-interleaved-fec-scheme
>
> As was discussed in the meeting last week, I would like to start a
> Working Group Last Call on
>
> draft-ietf-fecframe-interleaved-fec-scheme . This draft can be found  
> at
>
> http://tools.ietf.org/html/draft-ietf-fecframe-interleaved-fec-scheme-02
>
> This WGLC will run until midnight, UTC, on April 16th.  Please read
> this draft and send your comments and questions to the list.
>
> Regards
> Marshall
>
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe


-- 
Colin Perkins
http://csperkins.org/




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

	Title		: RTP Payload Format for SPIRIT IP-MR Speech Codec
	Author(s)	: A. Setsko
	Filename	: draft-ietf-avt-rtp-ipmr-03.txt
	Pages		: 13
	Date		: 2009-4-15
	
This document specifies the payload format for packetization of SPIRIT 
IP-MR encoded speech signals into the Real-time Transport Protocol 
(RTP). The payload format supports transmission of multiple frames per 
payload and introduced redundancy for robustness against packet loss.

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Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions (RAMS)" available
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There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniza
tion-for-rtp-03.txt
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniz
ation-for-rtp-03.txt>=20
=20
We have incorporated the changes from the technical breakout session in
San Francisco. The major changes in this version of the draft include=20
1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts
2- changing the message names to reflect the title change
3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages
4- additional description/motivation for the various message flows has
been added
5- RTP/RTCP muxing has been added
=20
=20
We hope to make this a Working Group item, and will change the name of
the draft to avoid conflicts with other "synchronization" drafts at that
time.
=20
Bill VerSteeg
=20
=20

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<BODY>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>There =
is a new draft=20
of the "Rapid Acquisition of Multicast RTP Sessions (RAMS)" draft =
available at=20
<A=20
href=3D"http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-sync=
hronization-for-rtp-03.txt"><FONT=20
face=3D"Times New Roman"=20
size=3D3>http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-syn=
chronization-for-rtp-03.txt</FONT></A></FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>We=20
have&nbsp;incorporated the changes from the technical breakout session =
in San=20
Francisco. The major changes in this version of the draft include=20
</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>1- =
Changing the=20
document title to avoid confusion with other ongoing "synchronization"=20
drafts</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>2- =
changing the=20
message names to reflect the title change</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>3- =
clarification of=20
the RTCP message semantics, including&nbsp;changes to the&nbsp;"Request" =
and=20
"Inform" messages</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>4- =
additional=20
description/motivation for the various message flows has been=20
added</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>5- =
RTP/RTCP muxing=20
has been added</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>We =
hope to make this=20
a Working Group item, and will change the name of the draft to avoid =
conflicts=20
with other "synchronization" drafts at that time.</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial size=3D2>Bill=20
VerSteeg</FONT></SPAN></DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D084481216-16042009><FONT face=3DArial=20
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From wwwrun@core3.amsl.com  Thu Apr 16 13:57:40 2009
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Subject: [AVT] Posting of IPR Disclosure related to Cisco System's Statement of IPR related to draft-ietf-avt-rtcp-xr-loss-conceal-01
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Dear Geoff Hunt, Alan Clark:

An IPR disclosure that pertains to your Internet-Draft entitled "RTCP XR Report
Block for Loss Concealment metric Reporting"
(draft-ietf-avt-rtcp-xr-loss-conceal) was submitted to the IETF Secretariat on
2009-04-15 and has been posted on the "IETF Page of Intellectual Property Rights
Disclosures" (https://datatracker.ietf.org/ipr/1131/). The title of the IPR
disclosure is "Cisco System's Statement of IPR related to
draft-ietf-avt-rtcp-xr-loss-conceal-01."

The IETF Secretariat



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Dear Geoff Hunt, Alan Clark:

An IPR disclosure that pertains to your Internet-Draft entitled "RTCP XR Report
Block for Concealed Seconds metric Reporting" (draft-ietf-avt-rtcp-xr-concsec)
was submitted to the IETF Secretariat on 2009-04-15 and has been posted on the
"IETF Page of Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/1132/). The title of the IPR disclosure is
"Cisco System's Statement of IPR related to draft-ietf-avt-rtcp-xr-concsec-01."

The IETF Secretariat



From tom.taylor@rogers.com  Fri Apr 17 17:48:24 2009
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Subject: [AVT] IETF 74 AVT Minutes first cut
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I'll improve the minutes for the second session before submitting, but in the 
meantime, please check everything for accuracy.

Tom Taylor


AVT  Minutes
============

AVT met for two sessions at IETF 74: Monday 1300-1500, and Thursday 1510-1610.
Roni Even and Tom Taylor chaired the sessions.

Monday 1300-1500
================

Dave Oran <oran@cisco.com> took notes, with some help from Joerg Ott 
<jo@netlab.tkk.fi>. Jonathan Lennox <jonathan@vidyo.com> acted as Jabber scribe. 
There were 75 attendees at this session.

Chair report (Roni Even)
========================
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-0.pdf

- Slight change to agenda: moved security issues to Thursday, RTCP-XR moved to
today.

- NOTE WELL was presented.

- Chairs gave document status - see agenda/intro slides for individual document
details. Noted that SSM is blocking some key documents. Apparently a production 
problem -- using Word. In addition to drafts listed, noted that SVC and 3984bis 
are ready to go to the IESG.

- Received a liaison from ITU-T Q18 of WP1/16. Desire is to avoid tandem media
processing entities on a bearer path (e.g. echo cancellers). See chair slides
for discussion and details. WG is asked to give chairs input (via mailing list)
on how to respond.
Colin Perkins on Jabber: do we need a joint BoF to discuss requirements?

- Need charter update for RTCP extensions.

- There is an issue with codec references for media sub-type registration in the
standards tree when the codec itself is proprietary but the payload type is 
defined by an IETF document. Registry rules are unclear. Topic is on agenda for 
a meeting between the Chairs and ADs Thursday morning.

- RFC3016 mainly used by 3GPP, but some misalignments with 3GPP PSS service.
draft-schmidt-avt-rfc3016bis-01 provides updates. The authors want it adopted as 
a WG work item.

RTP Payload for IP-MR Speech codec
==================================
draft-ietf-avt-rtp-ipmr-02
Roni presented on behalf of the authors
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-1.pdf

- on-the-fly adaptable wideband speech codec, VBR, etc.
- see slides for details of updates from previous version.
- Need review to ascertain whether it is ready for last call.

No one indicated that they had read the draft.

Rapid Synchronization of RTP flows
==================================
draft-perkins-avt-rapid-rtp-sync-03
Roni presented on behalf of the authors
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-2.pdf

- general agreement to change name of the rapid multicast synch work to avoid
terminology confusion. This draft will keep the "rapid synch" terminology.

- current draft is a merge of draft-perkins... and draft-schierl...

- see slides for details on draft technical content and updates since -00.

- proposed solution to jitter problem is to use the NTP header extension for all
streams in the session

- trying to encourage DVB to use this spec instead of rolling their own for TM-AVC.

Chairs polled the room as to whether it should be a working group item. Hum
agreement it should be - will confirm on mailing list.

Ye-Kui Wang <yekuiwang@huawei.com> noted that "SSM" was spelled out in different 
ways. It should be "Source-specific multicast".

Rapid Synchronization with RTP Multicast Sessions
=================================================

(1) draft-versteeg-avt-rapid-synchronization-for-rtp-02
     ---------------------------------------------------
Presented by Bill Ver Steeg <versteb@cisco.com>
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-3.pdf

The drafts presented at IETF 73 were merged into this one. However, there is 
another draft dealing with this topic (see below). To avoid confusion with the 
Perkins-Schierl work item it is proposed to rename this draft to "Unicast-based 
Rapid Acquisition of Multicast RTP Sessions".

- See slides for background and enumeration of updates from last version.

- Current draft is a merge of -01 and draft=-levin-rtp-burst. They were similar
and not a big deal to combine

- Went to TLV encoding of the fields. The earlier dependence on MPEG-2 transport 
stream transport has been removed. Adding capability for vendor-specific fields.

Roni Even remarked that there was no text explaining how the TLVs are used and 
what is the behavior associated with them. Bill Ver Steeg said they would work 
to clarify the semantics. Dave Oran noted that in most cases these fields are 
used strictly for optimizations.  Roni argued that designers will not implement 
them if they see no gain from doing so. Authors will continue to consider the 
matter.

- issue around when and whether we can use extended RTP sequence numbers in the
control messages.

Breakout scheduled for Tuesday 9 am to discuss open issues.

Roni: once a it is a WG item we can think about fixing FMT items.

Xavier Marjou <xavier.marjou@orange-ftgroup.com>: was there thought as to 
whether everything should be in RTP header extensions rather than RTCP messages? 
He had NAT interference in mind. Dave Oran responded that most of the messaging 
fits with the AVPF control channel model, but it might be a good idea to 
piggyback RS->RR control information for burst rate changes. Zeev Vax 
<zeevvax@microsoft.com> added that you can derive burst rate from the received 
data, but the idea of using header extensions is worth considering.

Roni observed that we still need to iron out the details of the mechanism.
He would try to present the outcome of the breakout session on Thursday so we 
can get this to be a WG item as soon as possible.

(2) draft-peilin-avt-rtp-burst-01
     -----------------------------
Ye-Kui Wang <yekuiwang@huawei.com> presented on behalf of the authors.
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-7.pdf.

- there may be Huawei IPR on the draft.

- General architecture adopts the draft-versteeg... approach

- Two new items beyond what is in the other draft
-- 1: allows burst rate to be controlled by RR, RS or both together.
-- 2: Method to terminate a burst based on initiation of a different burst
- See slides and draft for details.

Yunfei Zhang [??]: Requirements OK, but (1) not sure how receiver can know what 
bit rate to provide to the sender, and (2) are these mandatory
requirements of the solution?

- Roni: clarification of (1) how does the RR know the peak rate it can use?
Answer: receiver just tells what it may know. Followup: should IETF be
specifying optimal bitrates? YeKui: not looking to optimize. No need for IETF to 
specify -- situations differ. Roni: look at RFC 3550. Zeev: Burst typically 5-10 
seconds. Can't adapt faster than RTT allows. Need stable jitter and loss 
estimate. Takes longer than time available. Risk of increased overhead without 
adding value.

- Yunfei second question: is it implied that support of rapid channel change is 
mandatory? No - depends on provider, but question of making system work better. 
Dave Oran: bear in mind that this is an optimization of an optimization. Second 
point: if message fails, don't want to worsen subscriber experience.

RTCP-XR* Drafts
===============
Geoff Hunt <geoff.hunt@bt.com> presented.
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-4.pdf.

- There is a large menagerie of these - see slides for a list. Split of drafts
due to a "re-architecting" of the XR drafts to separate the individual blocks to
be specific rather than packaged together in arbitrary sets.

- see slides for changes from prior versions.

A Cisco IPR claim affecting -xr-concsec was noted.

Al Morton: on PDV block draft, 2 points - (1) would good to reference an RFC on
PDV guidance (RFC5481), (2) Naming conventions in this draft no longer 
correspond to the terms we use in RFC3550 and elsewhere. He would be glad to 
help nail the names down, particularly before they get put into an IANA 
registry. There is now general convergence amongst SDOs.

Reviews requested.
Milestones still have to be firmed up. Cullen understands the logic of creating 
so many drafts, but suggested we discuss the prioritization of these drafts on 
the list.

RTP payload format for the CELT codec
=====================================
draft-valin-celt-rtp-profile-00
Presented by Greg Maxwell <gmaxwell@juniper.net>
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-5.pdf

- Presented background of codec characteristics (see slides). On chart 4, Dave 
Oran what the Y-axis was for the bar chart on the right -- scores from ITU-T 
methodology.

- To solve early media problems - could burn a bunch of payload types
(combinatorics kill it) so need to choose some common combinations.

- Future work consideration is signaling configuration data, also need to freeze
the CELT bit-stream.

RTP Payload format for DV (IEC 61834) Video
===========================================
draft-ietf-avt-rfc3189bis-02
Roni provided a quick review.
Slides at http://www.ietf.org/proceedings/09mar/slides/avt-6.pdf.

- Minor changes from last draft (see slides)

Need opinions on whether this is ready for WGLC.

AVT Thursday Mar 26
===================
Roni and Tom chairing

Status
=============
Roni

Perkins synch
In favour of making WG item? No objections

Summary from burst breakout session (chart)
Revision then go to list to make WG item

DTLS-SRTP Key Transport (Dan)
=============================
Apologies, material to be uploaded after presentation.

Supports interworking with sdescriptions.

David McGrew states values to this independently of EKT -- interworking needed

?? can add description of interworking with MIKEY? Yes. Please indicate which 
variety of MIKEY is of interest.

Roni: verify that the call flows work correctly in interworking.


Encrypted key transport (EKT) David McGrew
==========================================

Master key conveyed to many participants.
Layer of indirection separating key management (who should have it?) from 
conferencing details.

Group controller sets up DTLS-SRTP key transport
EKT uses 1 to many RTP channel

Defined two years ago, premature. Essential for scaling to large groups -- 
avoids layer violations.
Currently ses sdescriptions

Eric Rescorla: what alternatives to EKT.
A couple of DTLS-SRTP extensions give almost as good results.

How would you know it is safe to send EKT messages? SDP.
Looks quite useful.

Roni: Do more discussion of tradeoffs before merging documents. Think about 
multi-unicast cast, which scales differently from multicast.

?? need reliability, RTP not reliable. David: can lose some packets. Dan: can 
send in either RTCP or RTP. In latter case, common fate. Sender can use more 
replication initially.

AES-192 and AES-256 David McGrew
================================

128 bit keys won't expire for decades, but might want to be ready for larger keys.
Possibilities: future advances in cryptanalysis, larger computers.
(Competition/compatibility ZRTP). SWEETPEA compliance.

Should we make AES-192 optional, make AES-256 the target.

Definitely add DTLS-SRTP binding.

Agreed, make standard

Jonathan: what about SHA-1? Offline discussion.

Cullen: make sure SecArea gets involved.

AES-GCM and AES-CCM autheticated encryption David McGrew
========================================================

AES-GCM

Encryption + authen -- no need for HMAC

Arguably efficient, esp. for short packets

AES-CCM more compact code.

Draft uses interface defined by RFC 5116

Eric: supports making WG doc. But why CCM? Roni: see if existing usage mandates use
Dan xx: might consider SIV instead of CCM.

SRTP Store and forward Blom
===========================

Draft refocussed on use cases and requirements.

Cullen only reader. Didn't understand it all yet.
Eric: wondered why all the layers. Response: other scenarios.

Continuing discussion. To the list.






From petithug@acm.org  Sat Apr 18 08:29:50 2009
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RFC 4103 does not give any advice on what to do when a t140block to
send will create an RTP packet that will exceed the PMTU.  Should
the PMTU be considered as an implicit maximum character transmission
rate, or should multiple RTP packets with the same timestamp be
sent? (for the latter there is the additional issue that RFC 2198
does not provide any advice on the same problem).

For example let say that the cps is 30 and the PMTU is unknown so
576 bytes is the maximum IPv4 packet size that can be sent.  With 30
cps and a language that makes use of three bytes per character, the
maximum block size is 900 bytes which is more than the maximum IPv4
packet size without even counting the overhead.  In this case the
maximum implicit cps would be 17 cps.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

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Marc,
My calculations end up a factor 10 lower than yours.

You are right that we did not think there was any reason to go deep in a
discussion on MTU sizes in RFC 4103.

Instead we specify the intended use, that is for human input, and human
reading in a real-time conversation. 

The most rapid input I can imagine is from a speech-to-text application,
that maybe can reach 20 characters per second from a rapid speaker. 
That is why the maximum required rate to support is specified to be 30
characters per second.

The recommended sample time is 300 ms, resulting in sending 3.3 packets per
second when there is text to send. With an even caracter rate, that results
in 9 new characters per packet. 

With the recommended two generations of redundancy, it will mean that 27
characters are packed in each packet.

If they are from a language that make use of three UTF-8 bytes per
character, there will be a mean of 81 bytes text per packet.

Add the IPv4, UDP, RTP and RFC2198 overhead of -say 70 bytes, the packet
size that will be close to maximum is 151 bytes. That is so far from the
maximum MTU size 

This is a maximum. In reality, text is usually entered by typing, resulting
in only 1-4 new characters per packet. 

Even if the reasoning above tells that there is no risk at all for exceeding
the maximum MTU-size, the application should follow a reasonable strategy to
be sure: The application should check so that the maximum MTU is not
exceeded. If there is a risk for that, excess characters should be delayed
and transmitted in next regular packet to transmit. 

Gunnar

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Marc
Petit-Huguenin
Sent: Saturday, April 18, 2009 5:31 PM
To: avt@ietf.org
Subject: [AVT] Another question on RFC 4103

RFC 4103 does not give any advice on what to do when a t140block to send
will create an RTP packet that will exceed the PMTU.  Should the PMTU be
considered as an implicit maximum character transmission rate, or should
multiple RTP packets with the same timestamp be sent? (for the latter there
is the additional issue that RFC 2198 does not provide any advice on the
same problem).

For example let say that the cps is 30 and the PMTU is unknown so
576 bytes is the maximum IPv4 packet size that can be sent.  With 30 cps and
a language that makes use of three bytes per character, the maximum block
size is 900 bytes which is more than the maximum IPv4 packet size without
even counting the overhead.  In this case the maximum implicit cps would be
17 cps.

--
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org
_______________________________________________
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From petithug@acm.org  Sat Apr 18 13:13:48 2009
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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [AVT] Another question on RFC 4103
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Gunnar Hellstrom wrote:
> Marc,
> My calculations end up a factor 10 lower than yours.

Section 6 of RFC 4103 states that "The [cps] value shall be used as
a mean value over any 10-second internal."  So at 30 cps, one can
send nothing for 10 seconds then send a block of 300 characters
without triggering the rate limiter.  This can be the case when a
text (e.g a long URL) is copied and pasted in the UI.

> 
> You are right that we did not think there was any reason to go deep in a
> discussion on MTU sizes in RFC 4103.
> 
> Instead we specify the intended use, that is for human input, and human
> reading in a real-time conversation. 
> 
> The most rapid input I can imagine is from a speech-to-text application,
> that maybe can reach 20 characters per second from a rapid speaker. 
> That is why the maximum required rate to support is specified to be 30
> characters per second.

RFC 4103 says that the _default_ value is 30, but where is it said
that the _maximum_ is 30?

> 
> The recommended sample time is 300 ms, resulting in sending 3.3 packets per
> second when there is text to send. With an even caracter rate, that results
> in 9 new characters per packet. 
> 
> With the recommended two generations of redundancy, it will mean that 27
> characters are packed in each packet.
> 
> If they are from a language that make use of three UTF-8 bytes per
> character, there will be a mean of 81 bytes text per packet.
> 
> Add the IPv4, UDP, RTP and RFC2198 overhead of -say 70 bytes, the packet
> size that will be close to maximum is 151 bytes. That is so far from the
> maximum MTU size 
> 
> This is a maximum. In reality, text is usually entered by typing, resulting
> in only 1-4 new characters per packet. 
> 
> Even if the reasoning above tells that there is no risk at all for exceeding
> the maximum MTU-size, the application should follow a reasonable strategy to
> be sure: The application should check so that the maximum MTU is not
> exceeded. If there is a risk for that, excess characters should be delayed
> and transmitted in next regular packet to transmit. 

OK, so your answer is that the PMTU must be considered as an
implicit maximum character transmission rate, and must override the
provided cps if it is higher than the value calculated from the PMTU.

Thanks.

> 
> Gunnar
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Marc
> Petit-Huguenin
> Sent: Saturday, April 18, 2009 5:31 PM
> To: avt@ietf.org
> Subject: [AVT] Another question on RFC 4103
> 
> RFC 4103 does not give any advice on what to do when a t140block to send
> will create an RTP packet that will exceed the PMTU.  Should the PMTU be
> considered as an implicit maximum character transmission rate, or should
> multiple RTP packets with the same timestamp be sent? (for the latter there
> is the additional issue that RFC 2198 does not provide any advice on the
> same problem).
> 
> For example let say that the cps is 30 and the PMTU is unknown so
> 576 bytes is the maximum IPv4 packet size that can be sent.  With 30 cps and
> a language that makes use of three bytes per character, the maximum block
> size is 900 bytes which is more than the maximum IPv4 packet size without
> even counting the overhead.  In this case the maximum implicit cps would be
> 17 cps.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

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Extended draft paper submission: HPCNCS-09 call for papers
=A0
This Extended Call for Papers for the 2009 International Conference on High=
 Performance Computing, Networking and Communication Systems (HPCNCS-09) (w=
ebsite: http://www.PromoteResearch.org ) is for those who didn't get a chan=
ce to submit the papers for the earlier call for papers. The papers receive=
d and accepted in response to this extended call for papers will be include=
d in the final version of the respective conference proceedings. These proc=
eedings will be either ready by the time of the conference (i.e., they will=
 be available during the conference) or soon after the conference (before t=
he end of August 2009), based how fast the proceedings can be prepared.=20
Note: If you have already submitted a paper (whether accepted or rejected o=
r currently under review) for MULTICONF-09, please DO NOT submit that paper=
 again to this extended call for papers.=20

IMPORTANT DATES:=20
Draft paper submission date: May 11, 2009=20
Acceptance/rejection decision: May 21, 2009
Camera ready paper and copyright and pre-registration due: May 28, 2009
Conference dates: July 13-16, 2009
=A0
=A0
HPCNCS-09 will be held during July 13-16 2009 in Orlando, FL, USA. We invit=
e draft paper submissions. The conference will take place at the same time =
and venue where several other international conferences are taking place. T=
he other conferences include:
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Artificial Intellig=
ence and Pattern Recognition (AIPR-09)=20
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Automation, Robotic=
s and Control Systems (ARCS-09)
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Bioinformatics, Com=
putational Biology, Genomics and Chemoinformatics (BCBGC-09)=20
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Enterprise Informat=
ion Systems and Web Technologies (EISWT-09)
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Information Securit=
y and Privacy (ISP-09)
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Recent Advances in =
Information Technology and Applications (RAITA-09)
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Software Engineerin=
g Theory and Practice (SETP-09)=20
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Theory and Applicat=
ions of Computational Science (TACS-09)
=B7=A0=A0=A0=A0=A0=A0=A0=A0 International Conference on Theoretical and Mat=
hematical Foundations of Computer Science (TMFCS-09)
=A0
The website http://www.PromoteResearch.org contains more details.
=A0
Sincerely
John Edward
Publicity committee
=A0
=A0
=A0=0A=0A=0A      
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<table cellspacing=3D"0" cellpadding=3D"0" border=3D"0" ><tr><td valign=3D"=
top" style=3D"font: inherit;"><P style=3D"MARGIN: 0in 0in 0pt"><A name=3DOL=
E_LINK4></A><A name=3DOLE_LINK3><SPAN style=3D"mso-bookmark: OLE_LINK4"><FO=
NT face=3D"Times New Roman" size=3D3>Extended draft paper submission: HPCNC=
S-09 call for papers</FONT></SPAN></A></DIV>
<P style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookmark: OLE_LINK3"><S=
PAN style=3D"mso-bookmark: OLE_LINK4"><?xml:namespace prefix =3D o ns =3D "=
urn:schemas-microsoft-com:office:office" /><o:p><FONT face=3D"Times New Rom=
an" size=3D3>&nbsp;</FONT></o:p></SPAN></SPAN></DIV>
<DIV><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-bookmark: O=
LE_LINK4"><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN class=3Dstyle=
4>This Extended Call for Papers for the </SPAN>2009 <SPAN style=3D"COLOR: b=
lack">International Conference on High Performance Computing, Networking an=
d Communication Systems (HPCNCS-09) </SPAN>(website: </FONT></FONT></SPAN><=
/SPAN><A href=3D"http://www.promoteresearch.org/"><FONT size=3D3><FONT face=
=3D"Times New Roman"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D=
"mso-bookmark: OLE_LINK4">http://www.PromoteResearch.org</SPAN></SPAN><SPAN=
 style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4">=
</SPAN></SPAN></FONT></FONT></A><SPAN style=3D"mso-bookmark: OLE_LINK3"><SP=
AN style=3D"mso-bookmark: OLE_LINK4"><FONT size=3D3><FONT face=3D"Times New=
 Roman"> ) <SPAN class=3Dstyle4>is for those who didn't get a chance to sub=
mit the papers for the earlier call for papers. The papers received and acc=
epted in response to this
 extended call for papers will be included in the final version of the resp=
ective conference proceedings. These proceedings will be either ready by th=
e time of the conference (i.e., they will be available during the conferenc=
e) or soon after the conference (before the end of August 2009), based how =
fast the proceedings can be prepared. </SPAN></FONT></FONT></SPAN></SPAN></=
DIV>
<P class=3Dstyle41 style=3D"MARGIN: auto 0in"><SPAN style=3D"mso-bookmark: =
OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><FONT size=3D3><FONT fac=
e=3D"Times New Roman"><STRONG>Note:</STRONG> If you have already submitted =
a paper (whether accepted or rejected or currently under review) for MULTIC=
ONF-09, please DO NOT submit that paper again to this extended call for pap=
ers. </FONT></FONT></SPAN></SPAN></DIV>
<P class=3Dstyle41 style=3D"MARGIN: auto 0in"><SPAN style=3D"mso-bookmark: =
OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><BR><STRONG><FONT face=
=3D"Times New Roman" size=3D3>IMPORTANT DATES: </FONT></STRONG></SPAN></SPA=
N></DIV>
<P class=3Dstyle41 style=3D"MARGIN: auto 0in"><SPAN style=3D"mso-bookmark: =
OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><FONT face=3D"Times New =
Roman" size=3D3>Draft paper submission date: May 11, 2009 <BR>Acceptance/re=
jection decision: May 21, 2009<BR>Camera ready paper and copyright and pre-=
registration due: May 28, 2009<BR>Conference dates: July 13-16, 2009</FONT>=
</SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookmark: OLE_LINK3"><S=
PAN style=3D"mso-bookmark: OLE_LINK4"><o:p><FONT face=3D"Times New Roman" s=
ize=3D3>&nbsp;</FONT></o:p></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookmark: OLE_LINK3"><S=
PAN style=3D"mso-bookmark: OLE_LINK4"><o:p><FONT face=3D"Times New Roman" s=
ize=3D3>&nbsp;</FONT></o:p></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookmark: OLE_LINK3"><S=
PAN style=3D"mso-bookmark: OLE_LINK4"><FONT size=3D3><FONT face=3D"Times Ne=
w Roman"><SPAN style=3D"COLOR: black">HPCNCS-09 </SPAN>will be held during =
July 13-16 2009 in Orlando, FL, USA. We invite draft paper submissions. The=
 conference will take place at the same time and venue where several other =
international conferences are taking place. The other conferences include:<=
SPAN style=3D"COLOR: black"><o:p></o:p></SPAN></FONT></FONT></SPAN></SPAN><=
/DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN style=3D"COLOR: black=
">International Conference on Artificial Intelligence and Pattern Recogniti=
on (AIPR-09)</SPAN> <SPAN style=3D"COLOR: black"><o:p></o:p></SPAN></FONT><=
/FONT></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Automation, Robotics and Control Systems (ARC=
S-09)<o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Bioinformatics, Computational Biology, Genomi=
cs and Chemoinformatics (BCBGC-09) <o:p></o:p></FONT></FONT></SPAN></SPAN><=
/SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Enterprise Information Systems and Web Techno=
logies (EISWT-09)<o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Information Security and Privacy (ISP-09)<o:p=
></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Recent Advances in Information Technology and=
 Applications (RAITA-09)<o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV=
>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Software Engineering Theory and Practice (SET=
P-09) <o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Theory and Applications of Computational Scie=
nce (TACS-09)<o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></DIV>
<P style=3D"MARGIN: 0in 0in 0pt 0.5in; TEXT-INDENT: -0.25in; mso-list: l0 l=
evel1 lfo1"><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-book=
mark: OLE_LINK4"><SPAN style=3D"COLOR: black; FONT-FAMILY: Symbol; mso-fare=
ast-font-family: Symbol; mso-bidi-font-family: Symbol"><SPAN style=3D"mso-l=
ist: Ignore"><FONT size=3D3>=B7</FONT><SPAN style=3D"FONT: 7pt 'Times New R=
oman'">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </SPAN></SPAN></SPA=
N><SPAN style=3D"COLOR: black"><FONT size=3D3><FONT face=3D"Times New Roman=
">International Conference on Theoretical and Mathematical Foundations of C=
omputer Science (TMFCS-09)<o:p></o:p></FONT></FONT></SPAN></SPAN></SPAN></D=
IV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookm=
ark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><o:p><FONT face=3D"=
Times New Roman" size=3D3>&nbsp;</FONT></o:p></SPAN></SPAN></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookm=
ark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><FONT face=3D"Times=
 New Roman" size=3D3>The website </FONT></SPAN></SPAN><A href=3D"http://www=
..promoteresearch.org/"><FONT size=3D3><FONT face=3D"Times New Roman"><SPAN =
style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4">h=
ttp://www.PromoteResearch.org</SPAN></SPAN><SPAN style=3D"mso-bookmark: OLE=
_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"></SPAN></SPAN></FONT></FONT=
></A><SPAN style=3D"mso-bookmark: OLE_LINK3"><SPAN style=3D"mso-bookmark: O=
LE_LINK4"><FONT face=3D"Times New Roman" size=3D3> contains more details.</=
FONT></SPAN></SPAN></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookm=
ark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><o:p><FONT face=3D"=
Times New Roman" size=3D3>&nbsp;</FONT></o:p></SPAN></SPAN></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><SPAN style=3D"mso-bookm=
ark: OLE_LINK3"><SPAN style=3D"mso-bookmark: OLE_LINK4"><FONT face=3D"Times=
 New Roman" size=3D3>Sincerely</FONT></SPAN></SPAN></DIV><SPAN style=3D"mso=
-bookmark: OLE_LINK4"></SPAN><SPAN style=3D"mso-bookmark: OLE_LINK3"></SPAN=
>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><FONT face=3D"Times New =
Roman" size=3D3>John Edward</FONT></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><FONT face=3D"Times New =
Roman" size=3D3>Publicity committee</FONT></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><o:p><FONT face=3D"Times=
 New Roman" size=3D3>&nbsp;</FONT></o:p></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><o:p><FONT face=3D"Times=
 New Roman" size=3D3>&nbsp;</FONT></o:p></DIV>
<P class=3DMsoNormal style=3D"MARGIN: 0in 0in 0pt"><o:p><FONT face=3D"Times=
 New Roman" size=3D3>&nbsp;</FONT></o:p></DIV></td></tr></table><br>=0A=0A=
=0A=0A      
--0-1221683748-1240166184=:5722--


From gunnar.hellstrom@omnitor.se  Sun Apr 19 14:02:18 2009
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From: "Gunnar Hellstrom" <gunnar.hellstrom@omnitor.se>
To: "'Marc Petit-Huguenin'" <petithug@acm.org>
References: <49E9F234.7000400@acm.org> <00ad01c9c05c$82028fe0$211ea8c0@GunnarH> <49EA34BC.8080600@acm.org>
Date: Sun, 19 Apr 2009 23:03:37 +0200
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Cc: avt@ietf.org
Subject: Re: [AVT] Another question on RFC 4103
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Marc,
See answers inline:=20

Gunnar
-----Original Message-----
From: Marc Petit-Huguenin [mailto:petithug@acm.org]=20
Sent: Saturday, April 18, 2009 10:15 PM
To: Gunnar Hellstrom
Cc: avt@ietf.org
Subject: Re: [AVT] Another question on RFC 4103

Gunnar Hellstrom wrote:
> Marc,
> My calculations end up a factor 10 lower than yours.

Section 6 of RFC 4103 states that "The [cps] value shall be used as a =
mean
value over any 10-second internal."  So at 30 cps, one can send nothing =
for
10 seconds then send a block of 300 characters without triggering the =
rate
limiter.  This can be the case when a text (e.g a long URL) is copied =
and
pasted in the UI.
<GH: The application is allowed to smear the transmission over a number =
of
packets so that any known MTU size limit can be obeyed. >
>=20
> You are right that we did not think there was any reason to go deep in =

> a discussion on MTU sizes in RFC 4103.
>=20
> Instead we specify the intended use, that is for human input, and=20
> human reading in a real-time conversation.
>=20
> The most rapid input I can imagine is from a speech-to-text=20
> application, that maybe can reach 20 characters per second from a =
rapid
speaker.
> That is why the maximum required rate to support is specified to be 30 =

> characters per second.

RFC 4103 says that the _default_ value is 30, but where is it said that =
the
_maximum_ is 30?
<GH: No, that is right. But in a couple of places it is said that it is
intended for human conversation and that automatic transmission of large
texts shall be avoided. Your example though, of pasting a URL into the =
text
stream is a good and valid example of actions that are common between
real-time text users, and one of the great benefits of complementing the
voice channel with a real-time text channel in the call. (Your example =
of a
URL with 300 bytes is a bit unusually long but can appear.)>
>=20
> The recommended sample time is 300 ms, resulting in sending 3.3=20
> packets per second when there is text to send. With an even caracter=20
> rate, that results in 9 new characters per packet.
>=20
> With the recommended two generations of redundancy, it will mean that=20
> 27 characters are packed in each packet.
>=20
> If they are from a language that make use of three UTF-8 bytes per=20
> character, there will be a mean of 81 bytes text per packet.
>=20
> Add the IPv4, UDP, RTP and RFC2198 overhead of -say 70 bytes, the=20
> packet size that will be close to maximum is 151 bytes. That is so far =

> from the maximum MTU size
>=20
> This is a maximum. In reality, text is usually entered by typing,=20
> resulting in only 1-4 new characters per packet.
>=20
> Even if the reasoning above tells that there is no risk at all for=20
> exceeding the maximum MTU-size, the application should follow a=20
> reasonable strategy to be sure: The application should check so that=20
> the maximum MTU is not exceeded. If there is a risk for that, excess=20
> characters should be delayed and transmitted in next regular packet to
transmit.

OK, so your answer is that the PMTU must be considered as an implicit
maximum character transmission rate, and must override the provided cps =
if
it is higher than the value calculated from the PMTU.

<GH: No, if you really have a low limit in allowed datagram size, I =
suggest
that you occasionally reduce the packet interval. Not to zero as you
proposed, because that is not allowed in RTP, but just as much shorter =
than
300 ms as you need to get your long text transmitted.
Or do you have any other proposal for how to handle your case? >   =20

Gunnar
--------------

Thanks.

>=20
> Gunnar
>=20
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of=20
> Marc Petit-Huguenin
> Sent: Saturday, April 18, 2009 5:31 PM
> To: avt@ietf.org
> Subject: [AVT] Another question on RFC 4103
>=20
> RFC 4103 does not give any advice on what to do when a t140block to=20
> send will create an RTP packet that will exceed the PMTU.  Should the=20
> PMTU be considered as an implicit maximum character transmission rate, =

> or should multiple RTP packets with the same timestamp be sent? (for=20
> the latter there is the additional issue that RFC 2198 does not=20
> provide any advice on the same problem).
>=20
> For example let say that the cps is 30 and the PMTU is unknown so
> 576 bytes is the maximum IPv4 packet size that can be sent.  With 30=20
> cps and a language that makes use of three bytes per character, the=20
> maximum block size is 900 bytes which is more than the maximum IPv4=20
> packet size without even counting the overhead.  In this case the=20
> maximum implicit cps would be
> 17 cps.

--
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

__________ NOD32 4018 (20090418) Information __________

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From petithug@acm.org  Sun Apr 19 14:58:49 2009
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Gunnar Hellstrom wrote:
> Marc,
> See answers inline: 
> 
> Gunnar
> -----Original Message-----
> From: Marc Petit-Huguenin [mailto:petithug@acm.org] 
> Sent: Saturday, April 18, 2009 10:15 PM
> To: Gunnar Hellstrom
> Cc: avt@ietf.org
> Subject: Re: [AVT] Another question on RFC 4103
> 

[...]

> OK, so your answer is that the PMTU must be considered as an implicit
> maximum character transmission rate, and must override the provided cps if
> it is higher than the value calculated from the PMTU.
> 
> <GH: No, if you really have a low limit in allowed datagram size, I suggest
> that you occasionally reduce the packet interval. Not to zero as you
> proposed, because that is not allowed in RTP, but just as much shorter than
> 300 ms as you need to get your long text transmitted.
> Or do you have any other proposal for how to handle your case? >    

I am not proposing anything, I am just trying to understand how this
is supposed to work.  My current understanding is that the cps is an
average of the 10 last seconds but that text blocks that creates RTP
packets exceeding the path MTU must be smoothed off multiple RTP
packets.

Thanks.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

From spoorthi.ka@globaledgesoft.com  Tue Apr 21 01:50:24 2009
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Subject: [AVT] Three way and Conference calls
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Hi,

Plz provide info on Role of RTP/RTCP in three way and conference calls.
During a conference call, RTP mixing has to be performed by the 
endpoints or servers. Plz clarify.

Thnx and Rgs,
spoorthi


From ingemar.s.johansson@ericsson.com  Tue Apr 21 05:53:56 2009
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions (RAMS)" available
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Hi

I have read through the updated draft and have a few comments and =
requests for your consideration.

In general (as I hinted before) I would like this draft to not just =
discuss the unicast version of RAMS, instead I would like it to also =
incorporate the multicast version into the framework.=20
I have looked into this and I would say that support for the multicast =
alternative as described in=20
http://tools.ietf.org/id/draft-johansson-avt-mcast-based-rams-00.txt
is fairly straightforward to add to the existing framework, I will =
describe in brief below the key additions that I see today.
The title of the draft would then become "Rapid Acquisition of Multicast =
RTP Sessions". I can offer to add the necessary parts in the document =
that describes the multicast based version of RAMS.

Packet structure:
In the draft, three FMT's are defined for the RAMS R,I and T messages. I =
wonder if it would be better to use one just one FMT for this "family" =
of messages and use a subtype to define R, I, T (or any other message). =
This was also brought up during the break-out session in SF.=20
Also I would consider a version field as well as there is a chance that =
things need to be changed in the future and TLV extensions may not solve =
all backwards compatibility issues in a satisfactory way (changing =
version number should not be too easy though, TLV elements should be the =
first choice).=20
With this in mind the first 32 bits may look like:
 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | ver |subtype| flags   .... |reserved                          |       =
  =20
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
In this example 3 bits are used for the version number and 5 bits are =
used for the subtype but there may be a better distribution here. The =
number of bits for the flags and reserved field depends on the subtype
The subtypes would be encoded as
RAMS-R =3D 0
RAMS-I =3D 1
RAMS-T =3D 2


Figure 2:=20
A flow diagram that describes the multicast version to be added.

Figure 3:=20
A description that describes the multicast version to be added.

Section 6.4: Failure cases
An alternative to RAMS-T in the multicast version of RAMS is the use of =
IGMP-Leave (details about this are needed)

Section 7.1 RAMS Request
Here the 32 bit header is added (with subtype=3D0) and possibly the =
preference and or support of the multicast and the unicast version of =
RAMS can be defined in the flags field. The latter is likely needed as =
the SDP's are likely only declarative.

Section 7.2 RAMS Information
Here the 32 bit header is added (with subtype=3D1) and the MSN,S and =
Response field is put in the flags field.
An additional TLV element is added that includes a multicast address and =
possibly other information such as expected delay until the multicast =
RAMS starts. This field is mandatory if the multicast address for RAMS =
is described in this way.

Section 7.2 RAMS Termination
Here the 32 bit header is added (with subtype=3D2)=20
Note that RAMS-T may not be needed in the multicast version of RAMS =
(T.B.D)

Comments to my comments are welcome
Regards
Ingemar
*******************************************=20
Ingemar Johansson=20
Senior Research Engineer, IETF "nethead"=20
EAB/TVP - Multimedia Technologies=20
Ericsson Research Ericsson AB=20
Box 920 S-971 28 Lule=E5, Sweden=20
Tel: +46 (0)10 7143042=20
ECN: 852-43042=20
ECC: 852-19042
Mobile: +46 (0)730 783289=20
Visit http://labs.ericsson.com !
*******************************************=20

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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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Hi Ingemar,=20

> Hi
>=20
> I have read through the updated draft and have a few comments=20
> and requests for your consideration.

Thanks for the review.
=20
> In general (as I hinted before) I would like this draft to=20
> not just discuss the unicast version of RAMS, instead I would=20
> like it to also incorporate the multicast version into the framework.=20

While I see the common goal for these two methods, I believe the one you =
proposed is an additional enhancement on the unicast approach. Regarding =
whether they need to be combined or not, I have some comments below.

> I have looked into this and I would say that support for the=20
> multicast alternative as described in=20
> http://tools.ietf.org/id/draft-johansson-avt-mcast-based-rams-00.txt
> is fairly straightforward to add to the existing framework, I=20
> will describe in brief below the key additions that I see today.

I would not say it would be "fairly straighforward." To make the =
multicast bursting effective, many parameters need to be adjusted =
properly and optimized. Ow, it can easily degrade the performance even =
at the expense of additional complexity.

A concern I have which is not addressed in your document is that to make =
the proposal make sense, Td needs to be small enough. But, to ensure =
that all receivers join the burst mcast session before the burst starts, =
Td cannot be very small and RS should not admit last-minute RAMS =
requests for that multicast burst (they need to wait for the next =
batch). This is because if some receivers miss the first few packets of =
the burst, their acquisition will fail. Also, how many clients can the =
retransmission server aggregate in a single burst batch in a practical =
scenario? The eligible requests should be received within +/- a few =
hundereds of milliseconds. How practical is this, I wonder.

The selection of a common burst bitrate is also critical. If everybody =
supports a different burst bitrate and there is a large variation, =
multicast bursting will perform poorly for the well-connected clients. =
One solution could be to include the receivers with an available bw that =
is smaller than R1 and larger than R2 in the multicast pool. The very =
fast and very slow receivers should be still served via unicast.

My personal view is that the multicast bursting proposal will be a nice =
solution when the issues above (and possible others) are addressed, =
however, I think it should be a separate draft that normatively =
references the unicast draft. This way, we can keep the unicast draft =
more focused.

> The title of the draft would then become "Rapid Acquisition=20
> of Multicast RTP Sessions". I can offer to add the necessary=20
> parts in the document that describes the multicast based=20
> version of RAMS.
>=20
> Packet structure:
> In the draft, three FMT's are defined for the RAMS R,I and T=20
> messages. I wonder if it would be better to use one just one=20
> FMT for this "family" of messages and use a subtype to define=20
> R, I, T (or any other message). This was also brought up=20
> during the break-out session in SF.=20

This was indeed a good recommendation and we already took that into =
account, but we did not want to modify the packet formats too much to =
avoid any confusion in this version. If you look at the open items =
section, it is already there.

> Also I would consider a version field as well as there is a=20
> chance that things need to be changed in the future and TLV=20
> extensions may not solve all backwards compatibility issues=20
> in a satisfactory way (changing version number should not be=20
> too easy though, TLV elements should be the first choice).=20

In a later version, the meanings of the mandatory fields would not =
change and TLVs are optional anyway. So, why do you think new TLVs would =
not give you what you want? Not that I am against using a version field, =
but I would rather see a need for it.

> With this in mind the first 32 bits may look like:
>  0                   1                   2                   3
>  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>  | ver |subtype| flags   .... |reserved                      =20
>    |         =20
>  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> In this example 3 bits are used for the version number and 5=20
> bits are used for the subtype but there may be a better=20
> distribution here. The number of bits for the flags and=20
> reserved field depends on the subtype
> The subtypes would be encoded as
> RAMS-R =3D 0
> RAMS-I =3D 1
> RAMS-T =3D 2
>=20
>=20
> Figure 2:=20
> A flow diagram that describes the multicast version to be added.
>=20
> Figure 3:=20
> A description that describes the multicast version to be added.
>=20
> Section 6.4: Failure cases
> An alternative to RAMS-T in the multicast version of RAMS is=20
> the use of IGMP-Leave (details about this are needed)
>=20
> Section 7.1 RAMS Request
> Here the 32 bit header is added (with subtype=3D0) and possibly=20
> the preference and or support of the multicast and the=20
> unicast version of RAMS can be defined in the flags field.=20
> The latter is likely needed as the SDP's are likely only declarative.
>
> Section 7.2 RAMS Information
> Here the 32 bit header is added (with subtype=3D1) and the=20
> MSN,S and Response field is put in the flags field.
> An additional TLV element is added that includes a multicast=20
> address and possibly other information such as expected delay=20
> until the multicast RAMS starts. This field is mandatory if=20
> the multicast address for RAMS is described in this way.
>=20
> Section 7.2 RAMS Termination
> Here the 32 bit header is added (with subtype=3D2)=20
> Note that RAMS-T may not be needed in the multicast version=20
> of RAMS (T.B.D)

These could be all addressed if/when we decide to combine the mcast =
bursting support.

Thanks again for the detail review.

BR,
-acbegen
=20
> Comments to my comments are welcome
> Regards
> Ingemar
> *******************************************=20
> Ingemar Johansson=20
> Senior Research Engineer, IETF "nethead"=20
> EAB/TVP - Multimedia Technologies=20
> Ericsson Research Ericsson AB=20
> Box 920 S-971 28 Lule=E5, Sweden=20
> Tel: +46 (0)10 7143042=20
> ECN: 852-43042=20
> ECC: 852-19042
> Mobile: +46 (0)730 783289=20
> Visit http://labs.ericsson.com !
> *******************************************=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>=20

From br@brianrosen.net  Tue Apr 21 14:01:52 2009
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Subject: Re: [AVT] Three way and Conference calls
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RTP and RTPC are always used in a single end to end leg.  The mixer has a
point to point connection with each leg.  The mixer can be in an endpoint or
in a server.  If a phone, for example, can mix a three way, it may simply
have point to point RTP connections with the other two parties.  If it
can't, it may need an external mixer to do that.  That mixer would have 3
point to point RTP connections.


     B                 S
   /   \     or     /  |  \
  A     C          A   B   C

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Spoorthi
Sent: Tuesday, April 21, 2009 4:50 AM
To: avt@ietf.org
Subject: [AVT] Three way and Conference calls

Hi,

Plz provide info on Role of RTP/RTCP in three way and conference calls.
During a conference call, RTP mixing has to be performed by the 
endpoints or servers. Plz clarify.

Thnx and Rgs,
spoorthi

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt


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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 Format for H.264 Video
	Author(s)       : Y. Wang, et al.
	Filename        : draft-ietf-avt-rtp-rfc3984bis-05.txt
	Pages           : 97
	Date            : 2009-04-22

This memo describes an RTP Payload format for the ITU-T 
Recommendation H.264 video codec and the technically identical 
ISO/IEC International Standard 14496-10 video codec, excluding the 
Scalable Video Coding (SVC) extension and the Multivew Video Coding 
extension, for which the RTP payload formats are defined elsewhere.  
The RTP payload format allows for packetization of one or more 
Network Abstraction Layer Units (NALUs), produced by an H.264 video 
encoder, in each RTP payload.  The payload format has wide 
applicability, as it supports applications from simple low bit-rate 
conversational usage, to Internet video streaming with interleaved 
transmission, to high bit-rate video-on-demand. 

This memo obsoletes RFC 3984.  Changes from RFC 3984 are summarized 
in section 18.  Issues on backward compatibility to RFC 3984 are 
discussed in section 17.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-rfc3984bis-05.txt

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From yekuiwang@huawei.com  Wed Apr 22 18:04:12 2009
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The changes compared to -04 include the change of SDP reference from RFC
2327 to RFC 4566, as well as some purely editorial changes, all suggested by
Tom Taylor.  Many thanks to Tom!

BR, YK


> -----Original Message-----
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> Behalf Of Internet-Drafts@ietf.org
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> Subject: [AVT] I-D Action:draft-ietf-avt-rtp-rfc3984bis-05.txt
> 
> 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 Format for H.264 Video
> 	Author(s)       : Y. Wang, et al.
> 	Filename        : draft-ietf-avt-rtp-rfc3984bis-05.txt
> 	Pages           : 97
> 	Date            : 2009-04-22
> 
> This memo describes an RTP Payload format for the ITU-T 
> Recommendation H.264 video codec and the technically 
> identical ISO/IEC International Standard 14496-10 video 
> codec, excluding the Scalable Video Coding (SVC) extension 
> and the Multivew Video Coding extension, for which the RTP 
> payload formats are defined elsewhere.  
> The RTP payload format allows for packetization of one or 
> more Network Abstraction Layer Units (NALUs), produced by an 
> H.264 video encoder, in each RTP payload.  The payload format 
> has wide applicability, as it supports applications from 
> simple low bit-rate conversational usage, to Internet video 
> streaming with interleaved transmission, to high bit-rate 
> video-on-demand. 
> 
> This memo obsoletes RFC 3984.  Changes from RFC 3984 are 
> summarized in section 18.  Issues on backward compatibility 
> to RFC 3984 are discussed in section 17.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-rfc3984
> bis-05.txt
> 
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> 
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> 



From timur.friedman@gmail.com  Wed Apr 22 20:31:14 2009
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Subject: [AVT] misleading RFC 3611 statement on RTCP packet stacking
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--001636163c5f09cf0f0468308698
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Some people are reading RFC 3611 as overriding RFC 3550 on RTCP packet
stacking.

This is what Sec. 1.1 of RFC 3611 says:

"XR packets supplement the existing RTCP packets, and may be stacked with
other RTCP packets to form compound RTCP packets [9, Section 6]."

Reference 9 is RFC 3550, which says:

"all RTCP packets MUST be sent in a compound packet of at least two
individual packets"

That is, RFC 3550 says that RTCP packets MUST be stacked into a compound
packet, and some people are reading RFC 3611 as weakening that mandate to
say only that RTCP packets MAY be stacked into a compound packet (but also
MAY NOT, at least in the case of the XR packet).

As one of the authors of RFC 3611, I can say that it was not my intention t=
o
override RFC 3550. Nor was it the intention of the two of my eight
co-authors who I have discussed this with. Rather, the purpose of this
sentence was to remind the reader that XR packets, like any RTCP packets,
are designed to be stacked, and to helpfully point the reader to the RFC
3550 section that deals with RTCP packet stacking for further details.

To the best of my recollection, no discussion in AVT leading up to approval
of RFC 3611 indicates that any group participants at the time believed that
RFC 3611 would override RFC 3550 on RTCP packet stacking.

Still, I understand how people are reading RFC 3611 in this way. It does sa=
y
"may be stacked", and though "may" is not capitalized, it is nonetheless an
RFC 2119 keyword, meaning that the item is truly optional.

Co-authors Nick Duffield, Ramon Caceres, and myself would like to open
discussion on how to address this situation. We start by suggesting that we
file an RFC Errata report, indicating that the sentence should read instead=
:

=93XR packets supplement the existing RTCP packets, and stack with other RT=
CP
packets to form compound RTCP packets, as described in Sec. 6 of RFC 3550
[9].=94

By dropping the "may", we drop any hint that this RFC overrides the RFC 355=
0
stacking mandate. And by adding "as described in Sec. 6 of RFC 3550" we mak=
e
clear the informational nature of this sentence.

Timur Friedman

--001636163c5f09cf0f0468308698
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Some people are reading RFC 3611 as overriding RFC 3550 on RTCP packet stac=
king.<br><br>This is what Sec. 1.1 of RFC 3611 says:<br><br>&quot;XR packet=
s supplement the existing RTCP packets, and may be stacked
with other RTCP packets to form compound RTCP packets [9, Section 6].&quot;=
<br><br>Reference 9 is RFC 3550, which says:<br><br>&quot;all RTCP packets =
MUST be sent in a compound packet of at least two individual packets&quot;<=
br>

<br>That is, RFC 3550 says that RTCP packets MUST be stacked into a compoun=
d packet, and some people are reading RFC 3611 as weakening that mandate to=
 say only that RTCP packets MAY be stacked into a compound packet (but also=
 MAY NOT, at least in the case of the XR packet).<br>

<br>As one of the authors of RFC 3611, I can say that it was not my intenti=
on to override RFC 3550. Nor was it the intention of the two of my eight co=
-authors who I have discussed this with. Rather, the purpose of this senten=
ce was to remind the reader that XR packets, like any RTCP packets, are des=
igned to be stacked, and to helpfully point the reader to the RFC 3550 sect=
ion that deals with RTCP packet stacking for further details.<br>

<br>To the best of my recollection, no discussion in AVT leading up to appr=
oval of RFC 3611 indicates that any group participants at the time believed=
 that RFC 3611 would override RFC 3550 on RTCP packet stacking.<br><br>

Still, I understand how people are reading RFC 3611 in this way. It does sa=
y &quot;may be stacked&quot;, and though &quot;may&quot; is not capitalized=
, it is nonetheless an RFC 2119 keyword, meaning that the item is truly opt=
ional.<br>

<br>Co-authors Nick Duffield, Ramon Caceres, and myself would like to open =
discussion on how to address this situation. We start by suggesting that we=
 file an RFC Errata report, indicating that the sentence should read instea=
d:<br>

<br>=93XR
packets supplement the existing RTCP packets, and stack with other RTCP
packets to form compound RTCP packets, as described in Sec. 6 of RFC
3550 [9].=94<br><br>By dropping the &quot;may&quot;, we drop any hint that =
this RFC overrides the RFC 3550 stacking mandate. And by adding &quot;as de=
scribed in Sec. 6 of RFC 3550&quot; we make clear the informational nature =
of this sentence.<br>

<br>Timur Friedman<br><br>

--001636163c5f09cf0f0468308698--

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Date: Thu, 23 Apr 2009 11:45:18 -0400
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"Brian Rosen" <br@brianrosen.net> writes:

>RTP and RTPC are always used in a single end to end leg.  The mixer has a
>point to point connection with each leg.  The mixer can be in an endpoint or
>in a server.  If a phone, for example, can mix a three way, it may simply
>have point to point RTP connections with the other two parties.  If it
>can't, it may need an external mixer to do that.  That mixer would have 3
>point to point RTP connections.
>
>
>     B                 S
>   /   \     or     /  |  \
>  A     C          A   B   C


There's one other configuration for 3-way and conference calls:

    B
  /   \
 A --- C

I.e. a no-central-mixer solution, or an "all to all media" topology as
defined in RFC 5117.  Each endpoint would be responsible for doing it's own
mixing as needed, and the upstream and downstream bandwidth of A and C (for
the first diagram above) and A, B and C (for the second diagram) would be
roughly doubled compared to a mixer solution.  A & C (or A, B and C) also
probably need to be capable enough to decode two incoming streams, though
they could implement a hard switch behavior (instead of mixing) and only
decode one stream at a time.

This configuration isn't often used in the unicast world, I believe; it
fits better in a multicast world where the upstream bandwidth of A & C (or
A, B & C) wouldn't have to increase.  Also, as you add more participants,
the bandwidth and processing multiplication effect quickly will cause
problems. 

>-----Original Message-----
>Plz provide info on Role of RTP/RTCP in three way and conference calls.
>During a conference call, RTP mixing has to be performed by the 
>endpoints or servers. Plz clarify.

-- 
Randell Jesup, Worldgate (developers of the Ojo videophone), ex-Amiga OS team
rjesup@wgate.com
"The fetters imposed on liberty at home have ever been forged out of the weapons
provided for defence against real, pretended, or imaginary dangers from abroad."
		- James Madison, 4th US president (1751-1836)


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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		: The SEED Cipher Algorithm and Its Use with the Secure Real-time Transport Protocol (SRTP
	Author(s)	: S. Yoon, H. Park, J. Kim, H. Jeong, Y. Won
	Filename	: draft-ietf-avt-seed-srtp-11.txt
	Pages		: 12
	Date		: 2009-4-23
	
This document describes the use of the SEED block cipher algorithm in 
   the Secure Real-time Transport Protocol (SRTP) for providing 
   confidentiality for the Real-time Transport Protocol (RTP) traffic 
   and for the control traffic for RTP, the Real-time Transport Control 
   Protocol (RTCP).

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-avt-seed-srtp-11.txt

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From ron.even.tlv@gmail.com  Fri Apr 24 00:04:23 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Spoorthi'" <spoorthi.ka@globaledgesoft.com>, <avt@ietf.org>
References: <49ED88C5.1010802@globaledgesoft.com>
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Subject: Re: [AVT] Three way and Conference calls
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Hi,
You should look at http://tools.ietf.org/html/rfc5117 that discuss the
different RTP topologies. 
Still you can always do a full mesh and let every endpoint mix the streams.

Roni Even

-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Spoorthi
Sent: Tuesday, April 21, 2009 11:50 AM
To: avt@ietf.org
Subject: [AVT] Three way and Conference calls

Hi,

Plz provide info on Role of RTP/RTCP in three way and conference calls.
During a conference call, RTP mixing has to be performed by the 
endpoints or servers. Plz clarify.

Thnx and Rgs,
spoorthi

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt


From spoorthi.ka@globaledgesoft.com  Fri Apr 24 02:54:07 2009
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From: Spoorthi <spoorthi.ka@globaledgesoft.com>
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Subject: Re: [AVT] Three way and Conference calls
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A endpoint involved a three way call will have point to point RTP 
connections with the other two parties as shown below

*
     B    
   /   \    
  A     C       
*


Now, the endpoint the 'B' has to mix the audio received from the 
endpoint 'A' and transmit the same to endpoint 'C'.
The endpoint 'B' has to mix the audio received from the endpoint 'C' and 
transmit the same to endpoint 'A'.
To perform the mixing should the endpoint B deployed with a mixer or 
endpoint B has to do the audio mixing???
Plz clarify...

Thnx and Rgs,
spoorthi


Brian Rosen wrote:
> RTP and RTPC are always used in a single end to end leg.  The mixer has a
> point to point connection with each leg.  The mixer can be in an endpoint or
> in a server.  If a phone, for example, can mix a three way, it may simply
> have point to point RTP connections with the other two parties.  If it
> can't, it may need an external mixer to do that.  That mixer would have 3
> point to point RTP connections.
>
>
>      B                 S
>    /   \     or     /  |  \
>   A     C          A   B   C
>
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
> Spoorthi
> Sent: Tuesday, April 21, 2009 4:50 AM
> To: avt@ietf.org
> Subject: [AVT] Three way and Conference calls
>
> Hi,
>
> Plz provide info on Role of RTP/RTCP in three way and conference calls.
> During a conference call, RTP mixing has to be performed by the 
> endpoints or servers. Plz clarify.
>
> Thnx and Rgs,
> spoorthi
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>
>   


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Date: Fri, 24 Apr 2009 18:02:54 +0800
From: Zou ZiXuan <tendyntu@huawei.com>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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ÕâÊÇÒ»·â MIME ¸ñÊ½µÄ¶à²¿·ÖÓÊ¼þ¡£

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Hi,

         The burst duration (8 octets) in RAMS-I message is in a single
RTP ticks. And I note that from draft-01 on, the solution is
payload-independent (is my understanding correct?). So, for non-MEPG2-TS
payload, for example, H.264-NAL where audio and video content are
encapsulated separately in two RTP streams with different ticks, does
the duration indicated by this single ticks still work? Which stream
does this tick belong to?

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Bill Ver Steeg (versteb)
Sent: Friday, April 17, 2009 12:39 AM
To: avt@ietf.org
Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

 

There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniz
ation-for-rtp-03.txt>
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniza
tion-for-rtp-03.txt

 

We have incorporated the changes from the technical breakout session in
San Francisco. The major changes in this version of the draft include 

1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts

2- changing the message names to reflect the title change

3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages

4- additional description/motivation for the various message flows has
been added

5- RTP/RTCP muxing has been added

 

 

We hope to make this a Working Group item, and will change the name of
the draft to avoid conflicts with other "synchronization" drafts at that
time.

 

Bill VerSteeg

 

 


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<p class=MsoNormal><span lang=EN-US style='font-size:9.0pt;font-family:"Arial","sans-serif";
color:blue'>Hi,<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='font-size:9.0pt;font-family:"Arial","sans-serif";
color:blue'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The burst duration
(8 octets) in RAMS-I message is in a single RTP ticks. And I note that from draft-01
on, the solution is payload-independent (is my understanding correct?). So, for
non-MEPG2-TS payload, for example, H.264-NAL where audio and video content are
encapsulated separately in two RTP streams with different ticks, does the
duration indicated by this single ticks still work? Which stream does this tick
belong to?<o:p></o:p></span></p>

<p class=MsoNormal><span lang=EN-US style='font-size:9.0pt;font-family:"Arial","sans-serif";
color:blue'><o:p>&nbsp;</o:p></span></p>

<div style='border:none;border-left:solid blue 1.5pt;padding:0cm 0cm 0cm 4.0pt'>

<div>

<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>

<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> avt-bounces@ietf.org
[mailto:avt-bounces@ietf.org] <b>On Behalf Of </b>Bill Ver Steeg (versteb)<br>
<b>Sent:</b> Friday, April 17, 2009 12:39 AM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> [AVT] New draft of &quot;Rapid Acquisition of Multicast RTP
Sessions (RAMS)&quot; available<o:p></o:p></span></p>

</div>

</div>

<p class=MsoNormal><span lang=EN-US><o:p>&nbsp;</o:p></span></p>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>There
is a new draft of the &quot;Rapid Acquisition of Multicast RTP Sessions
(RAMS)&quot; draft available at <a
href="http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt"><span
style='font-size:12.0pt;font-family:"Times New Roman","serif"'>http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt</span></a></span><span
lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>We
have&nbsp;incorporated the changes from the technical breakout session in San
Francisco. The major changes in this version of the draft include </span><span
lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>1-
Changing the document title to avoid confusion with other ongoing
&quot;synchronization&quot; drafts</span><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>2-
changing the message names to reflect the title change</span><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>3-
clarification of the RTCP message semantics, including&nbsp;changes to
the&nbsp;&quot;Request&quot; and &quot;Inform&quot; messages</span><span
lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>4-
additional description/motivation for the various message flows has been added</span><span
lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>5-
RTP/RTCP muxing has been added</span><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>We
hope to make this a Working Group item, and will change the name of the draft
to avoid conflicts with other &quot;synchronization&quot; drafts at that time.</span><span
lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Bill
VerSteeg</span><span lang=EN-US><o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

<div>

<p class=MsoNormal><span lang=EN-US>&nbsp;<o:p></o:p></span></p>

</div>

</div>

</div>

</body>

</html>

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Hi,

I have two questions/comments mostly regarding the RTCP-SSM draft
(http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18). The background
of my questions is the use case of "broadcast channels" in an IPTV
context, where the channels are distributed over IP multicast (SSM). In
this context, the concept of feedback target(s) is obviously very
useful, and actually necessary given today's restrictions with multicast
in upstream in the "WAN".=20

For the IPTV use case, it also makes sense to have multiple feedback
targets, each responsible for a subset of RTP receivers, because of the
large number of RTP receivers involved (100.000s is possible). The way
to signal this by means of SDP is that each local subset of RTP
receivers sees a different IP address in the "a=3Drtcp: "portnumber" IN
IP4 "IP address"  line.  So, multiple "localised" FT  is covered by the
draft IMO.

However, in that case, the maximum bandwidth that can be used for RTCP
reporting by RTP receivers only needs to be determined by the size of
the local group of RTP receivers (sending RTCP to the same localised
Feedback Target), especially in the summary FB model. But in my
interpretation of the architecture, the actual RTCP bandwith limit
information can only be retrieved by the RTP receivers from the
information sent by the distribution source in the ssm session. Either
by refected RTCP reports from RTP receivers (not a feasible solution for
the given use case) or from a RSI message -in the summary FB model-
indicating the session size or explicitly the RTCP reporting bandwidth
limit. The fact is that this information is sent to ALL RTP receivers,
which may be reporting to different Feedback targets where each Feedback
target is collecting RTCP messages from a different number of RTP
receivers, and hence may have different "requirements" for total RTCP
reporting bandwidth limit.  My question is whether local Feedback
targets have been considered with respect to this issue?

My second question again has to do with multiple feedback targets for
the same use case.=20

In the IPTV use case for rtcpssm, Feedback Targets can also be used as
retransmission sources (RFC 4588) and in
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-
rtp-03, the feedback target architecture is used to allow fast
acquisition of SSM RTP sessions. The draft proposes that  a "burst
source" is co-located with the FT which transmits unicast data (starting
with a random access point of the video channel) from a cache that holds
the most recent data distributed over the RTP SSM. It does transmit the
RTP packets at an accelerated speed, allowing the RTP receiver to join
the SSM after some time.

The idea I would like to bring forward is to allow separate Feedback
Targets  for the same RTP receiver (population): one Feedback target
acts as RTP retransmission server (see
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-
rtp-03) and collects RTCP FB messages, while the other Feedback Target
is solely "responsible" for the non RTCP-FB messages, such as RTCP XR
messages. I realise this idea is not necessarily linked with  the
rtcp-ssm draft, but IMO only for a massive scale application that makes
use of RTP and leverages rtcp ssm such a separation is justified. The
possibility for separation of the monitoring service -associated with a
server function basically responsible for collecting RTCP receiver
reports for the purpose of monitoring- and the retransmission service
requiring a server function that is responsible for processing and
responding to RTCP FB messages, is driven from the point of view of
server function optimisation (different requirements on the server
platform).=20

In terms of signaling two Feedback Targets, we could make use of the
following lines:

a=3Drtcp:40001 IN IP4 192.0.2.1 (This is for retransmission)
a=3Drtcp-fb:96 nack
a=3Drtcp:40001 IN IP4 192.0.2.2 (This is for XR collection)


However, that is probably quite ambiguous..?
Furthermore I also realise such a solution introduces other questions
than just signaling syntax including the fact that we may need guiding
rules on how RTCP reports are issued by the RTP receiver with two RTCP
targets in place, as in the example above. The new RFC 5506 is certainly
helpful here, but this is probably not sufficient.=20
Comments appreciated on this idea.

Kind regards,
Tom Van Caenegem



=20


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<BR>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Hi,</FONT></SPAN>
</P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">I have two =
questions/comments mostly regarding the RTCP-SSM draft (</FONT></SPAN><A =
HREF=3D"http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18"><SPAN =
LANG=3D"en-us"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18</FONT=
></U></SPAN></A><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">). =
The background of my questions is the use case of &quot;broadcast =
channels&quot; in an IPTV context, where the channels are distributed =
over IP multicast (SSM). In this context, the concept of feedback =
target(s) is obviously very useful, and actually necessary given today's =
restrictions with multicast in upstream in the &quot;WAN&quot;. =
</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">For the IPTV use =
case, it also makes sense to have multiple feedback targets, each =
responsible for a subset of RTP receivers, because of the large number =
of RTP receivers involved (100.000s is possible). The way to signal this =
by means of SDP is that each local subset of RTP receivers sees a =
different IP address in the &quot;</FONT></SPAN><SPAN =
LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Courier New">a=3Drtcp: =
&quot;portnumber&quot; IN IP4 &quot;IP =
address&quot;&nbsp;</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 =
FACE=3D"Arial">line.&nbsp; So, multiple &quot;localised&quot; FT&nbsp; =
is covered by the draft IMO.</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">However, in that =
case, the maximum bandwidth that can be used for RTCP reporting by RTP =
receivers only needs to be determined by the size of the local group of =
RTP receivers (sending RTCP to the same localised Feedback Target), =
especially in the summary FB model. But in my interpretation of the =
architecture, the actual RTCP bandwith limit information can only be =
retrieved by the RTP receivers from the information sent by the =
distribution source in the ssm session. Either by refected RTCP reports =
from RTP receivers (not a feasible solution for the given use case) or =
from a RSI message -in the summary FB model- indicating the session size =
or explicitly the RTCP reporting bandwidth limit. The fact is that this =
information is sent to ALL RTP receivers, which may be reporting to =
different Feedback targets where each Feedback target is collecting RTCP =
messages from a different number of RTP receivers, and hence may have =
different &quot;requirements&quot; for total RTCP reporting bandwidth =
limit.&nbsp; My question is whether local Feedback targets have been =
considered with respect to this issue?</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">My second question =
again has to do with multiple feedback targets for the same use case. =
</FONT></SPAN>
</P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">In the IPTV use =
case for rtcpssm, Feedback Targets can also be used as retransmission =
sources (RFC 4588) and in </FONT></SPAN><A =
HREF=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizati=
on-for-rtp-03"><SPAN LANG=3D"en-us"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchr=
onization-for-rtp-03</FONT></U></SPAN></A><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Arial">, the feedback target architecture is used to =
allow fast acquisition of SSM RTP sessions. The draft proposes =
that&nbsp; a &quot;burst source&quot; is co-located with the FT which =
transmits unicast data (starting with a random access point of the video =
channel) from a cache that holds the most recent data distributed over =
the RTP SSM. It does transmit the RTP packets at an accelerated speed, =
allowing the RTP receiver to join the SSM after some =
time.</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">The idea I would =
like to bring forward is to allow separate Feedback Targets&nbsp; for =
the same RTP receiver (population): one Feedback target acts as RTP =
retransmission server (see </FONT></SPAN><A =
HREF=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizati=
on-for-rtp-03"><SPAN LANG=3D"en-us"><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Arial">http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchr=
onization-for-rtp-03</FONT></U></SPAN></A><SPAN LANG=3D"en-us"><FONT =
SIZE=3D2 FACE=3D"Arial">) and collects RTCP FB messages, while the other =
Feedback Target is solely &quot;responsible&quot; for the non RTCP-FB =
messages</FONT><FONT SIZE=3D2 FACE=3D"Arial">, such as RTCP XR messages. =
I realise this idea is not necessarily linked with&nbsp; the rtcp-ssm =
draft, but IMO only for a massive scale application that makes use of =
RTP and leverages rtcp ssm such a separation is justified. The =
possibility for separation of the monitoring service</FONT> <FONT =
SIZE=3D2 FACE=3D"Arial">&#8211;</FONT><FONT SIZE=3D2 =
FACE=3D"Arial">associated with a server function basically responsible =
for collecting RTCP receiver reports for the purpose of monitoring- and =
the retransmission service requiring a server function that is =
responsible for processing and responding to RTCP FB messages, is driven =
from the point of view of server function optimisation (different =
requirements on the server platform). </FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">In terms of =
signaling two Feedback Targets, we could make use of the following =
lines:</FONT></SPAN>
</P>

<P><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Courier =
New">a=3Drtcp:40001 IN IP4 192.0.2.1 (This is for =
retransmission)</FONT></SPAN>

<BR><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Courier =
New">a=3Drtcp-fb:96 nack</FONT></SPAN>

<BR><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Courier =
New">a=3Drtcp:40001 IN IP4 192.0.2.2 (This is for XR =
collection)</FONT></SPAN><SPAN LANG=3D"en-us"></SPAN>
</P>
<BR>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">However, that is =
probably quite ambiguous..?</FONT></SPAN>

<BR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Furthermore I =
also realise such a solution introduces other questions than just =
signaling syntax including the fact that we may need guiding rules on =
how RTCP reports are issued by the RTP receiver with two RTCP targets in =
place, as in the example above. The new RFC 5506 is certainly helpful =
here, but this is probably not sufficient. </FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Comments =
appreciated on this idea.</FONT></SPAN>
</P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Kind =
regards,</FONT></SPAN>

<BR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Tom Van =
Caenegem</FONT></SPAN>
</P>
<BR>
<BR>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;</FONT></SPAN>
</P>

</BODY>
</HTML>
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From ron.even.tlv@gmail.com  Fri Apr 24 04:24:59 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com>
In-Reply-To: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com>
Date: Fri, 24 Apr 2009 14:24:16 +0300
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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Hi,

I think that the current draft does not give a description of a system that
works since there is no text explaining how the RS knows the unicast
transport address on the RR to where to send the stream. 

If you mandate the use of RTP/RTCP mux it should say so otherwise the RAMS-R
should have an optional parameter that supplies this information and a flag
for using RTP/RTCP mux.

Thanks

Roni Even

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Bill
Ver Steeg (versteb)
Sent: Thursday, April 16, 2009 7:39 PM
To: avt@ietf.org
Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

 

There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronizatio
n-for-rtp-03.txt>
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronization
-for-rtp-03.txt

 

We have incorporated the changes from the technical breakout session in San
Francisco. The major changes in this version of the draft include 

1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts

2- changing the message names to reflect the title change

3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages

4- additional description/motivation for the various message flows has been
added

5- RTP/RTCP muxing has been added

 

 

We hope to make this a Working Group item, and will change the name of the
draft to avoid conflicts with other "synchronization" drafts at that time.

 

Bill VerSteeg

 

 


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think that the current draft does not give a =
description of a
system that works since there is no text explaining how the RS knows the
unicast transport address on the RR to where to send the stream. =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If you mandate the use of RTP/RTCP mux it should say so =
otherwise
the RAMS-R should have an optional parameter that supplies this =
information and
a flag for using RTP/RTCP mux.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of =
</b>Bill Ver
Steeg (versteb)<br>
<b>Sent:</b> Thursday, April 16, 2009 7:39 PM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> [AVT] New draft of &quot;Rapid Acquisition of Multicast =
RTP
Sessions (RAMS)&quot; available<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There
is a new draft of the &quot;Rapid Acquisition of Multicast RTP Sessions
(RAMS)&quot; draft available at <a
href=3D"http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-sync=
hronization-for-rtp-03.txt"><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>http://www.ietf.org/internet-drafts/draft-versteeg-avt-ra=
pid-synchronization-for-rtp-03.txt</span></a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We
have&nbsp;incorporated the changes from the technical breakout session =
in San
Francisco. The major changes in this version of the draft include =
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>1-
Changing the document title to avoid confusion with other ongoing
&quot;synchronization&quot; drafts</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>2-
changing the message names to reflect the title =
change</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>3-
clarification of the RTCP message semantics, including&nbsp;changes to
the&nbsp;&quot;Request&quot; and &quot;Inform&quot; =
messages</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>4-
additional description/motivation for the various message flows has been =
added</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>5-
RTP/RTCP muxing has been added</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We
hope to make this a Working Group item, and will change the name of the =
draft
to avoid conflicts with other &quot;synchronization&quot; drafts at that =
time.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Bill
VerSteeg</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

</body>

</html>

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From ron.even.tlv@gmail.com  Fri Apr 24 07:31:18 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <fecframe@ietf.org>
References: <04CAD96D4C5A3D48B1919248A8FE0D54091BAA88@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54091BAA88@xmb-sjc-215.amer.cisco.com>
Date: Fri, 24 Apr 2009 17:30:29 +0300
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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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Hi,
I am sorry that I did not review the draft before since I am not subscribed
to fecframe and AVT was not cc'ed on the WGLC.

I have some editorial comments:

1. In section 1 you reference RFC2733, I think it was obsolete by RFC 5109 ,
do you need this downref?

2. In the registration section 5.1, the comments applies to all subtypes:  
  2.1  You have for security considerations and published specification
"this document" it is better to write RFC xxxx and have a note to     	the
RFC editor to replace it with the RFC number allocated, this will show in
the IANA registry with the write pointer.

  2.2 In the restriction on usage I assume that the document is using RTP
framing and is not specified for other types, if this is the 	case it
should say so. E.g. "   This media type depends on RTP framing, and hence is
only defined for transfer via RTP [RFC3550].  	Transport within other
framing  protocols is not defined at this time"
  2.3 You need to add the "additional information" that is in the template
probably saying none

3. In section 5.2.1 it looks like you require symmetry between the send and
received stream. Is this your requirement. 

4. In section 5.2.1 can the answerer send a response with no FEC at all even
if not offered.

5. In section 5.2.2 Can the receiver just ignore the fec stream, the text
has a MUST use the configuration that is offered which can hint that it MUST
use FEC.

6. In section 7 you discuss an exchange using grouping. Is it also allowed
to use FEC withoutout grouping using the video or audio media types like
section 14 in RFC 5109


Thanks
Roni Even
 




-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Ali C.
Begen (abegen)
Sent: Sunday, April 19, 2009 12:04 AM
To: fecframe@ietf.org
Cc: avt@ietf.org
Subject: [AVT] FW: New Version Notification for
draft-ietf-fecframe-interleaved-fec-scheme-03

This update addresses all the comments received in the WGLC period initated
on April 1st. If there are further comments, please post them soon as we
would like to move forward with this draft.
http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interleaved-fec-sche
me-03.txt

BR,
-acbegen 

-----Original Message-----
From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
Sent: Saturday, April 18, 2009 1:58 PM
To: Ali C. Begen (abegen)
Subject: New Version Notification for
draft-ietf-fecframe-interleaved-fec-scheme-03 


A new version of I-D, draft-ietf-fecframe-interleaved-fec-scheme-03.txt has
been successfuly submitted by Ali Begen and posted to the IETF repository.

Filename:	 draft-ietf-fecframe-interleaved-fec-scheme
Revision:	 03
Title:		 RTP Payload Format for 1-D Interleaved Parity FEC
Creation_date:	 2009-04-18
WG ID:		 fecframe
Number_of_pages: 31

Abstract:
This document defines a new RTP payload format for the Forward Error
Correction (FEC) that is generated by the 1-D interleaved parity code
from a source media encapsulated in RTP.  The 1-D interleaved parity
code is a systematic code, where a number of repair symbols are
generated from a set of source symbols and sent in a repair flow
separate from the source flow that carries the source symbols.  The
1-D interleaved parity code offers a good protection against bursty
packet losses at a cost of decent complexity.  The new payload format
defined in this document is used (with some exceptions) as a part of
the DVB Application-layer FEC specification.
 



The IETF Secretariat.


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www.ietf.org/mailman/listinfo/avt


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Date: Fri, 24 Apr 2009 08:12:46 -0700
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Zou ZiXuan" <tendyntu@huawei.com>, "Bill Ver Steeg (versteb)" <versteb@cisco.com>, <avt@ietf.org>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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Hi Zou,

This is a very good question. Let me take a stab here.

Per RFC 3550, each SSRC space has its own clock rate and seq numbering. =
We include the SSRC in the RAMS request messages.

There are different flavors of your scneario. Lets assume that we have =
two RTP streams (audio and video).


Case 1: Different RTP sessions are transmitted over different mcast =
groups.

Here, we run a different independent RAMS session for each RTP stream =
that we want to rapidly acquire (they can run in parallel). Here one =
needs to pay attention to how to use the shared bandwidth among =
different bursts, though.

Case 2: Different RTP sessions are transmitted over the same mcast group =
(with different dest ports).

Again, we run a different independent RAMS for each RTP stream =
(different request for each, etc). But, one should keep in mind that the =
join times for the all RAMS sessions need to be common as there is only =
one multicast.

Case 3: One RTP session (same mcast group, same dest port) carrying both =
audio and video (this is SSRC multiplexing).

Similar to case 2, join time needs to be coordinated among different =
RAMS sessions running for each RTP stream. Since we signal SSRC in the =
request message and we have independent RAMS sessions, using different =
clock rates is not an issue.

Case 4: One RTP session, one SSRC with payload type multiplexing. This =
is not recommended for many reasons. On top of that, different payloads =
with different clock rates cannot be multiplexed in this fashion. So, I =
don't think this will apply to our problem.

HTH,
-acbegen


> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Zou ZiXuan
> Sent: Friday, April 24, 2009 3:03 AM
> To: Bill Ver Steeg (versteb); avt@ietf.org
> Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTP Sessions (RAMS)" available
>=20
> Hi,
>=20
>          The burst duration (8 octets) in RAMS-I message is=20
> in a single RTP ticks. And I note that from draft-01 on, the=20
> solution is payload-independent (is my understanding=20
> correct?). So, for non-MEPG2-TS payload, for example,=20
> H.264-NAL where audio and video content are encapsulated=20
> separately in two RTP streams with different ticks, does the=20
> duration indicated by this single ticks still work? Which=20
> stream does this tick belong to?
>=20
> =20
>=20
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Bill Ver Steeg (versteb)
> Sent: Friday, April 17, 2009 12:39 AM
> To: avt@ietf.org
> Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> RTP Sessions (RAMS)" available
>=20
> =20
>=20
> There is a new draft of the "Rapid Acquisition of Multicast=20
> RTP Sessions (RAMS)" draft available at=20
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
ynchronization-for-rtp-03.txt <http://www.ietf.org/internet-> =
drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
>=20
> =20
>=20
> We have incorporated the changes from the technical breakout=20
> session in San Francisco. The major changes in this version=20
> of the draft include=20
>=20
> 1- Changing the document title to avoid confusion with other=20
> ongoing "synchronization" drafts
>=20
> 2- changing the message names to reflect the title change
>=20
> 3- clarification of the RTCP message semantics, including=20
> changes to the "Request" and "Inform" messages
>=20
> 4- additional description/motivation for the various message=20
> flows has been added
>=20
> 5- RTP/RTCP muxing has been added
>=20
> =20
>=20
> =20
>=20
> We hope to make this a Working Group item, and will change=20
> the name of the draft to avoid conflicts with other=20
> "synchronization" drafts at that time.
>=20
> =20
>=20
> Bill VerSteeg
>=20
> =20
>=20
> =20
>=20
>=20

From marshall.eubanks@gmail.com  Fri Apr 24 08:45:18 2009
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Date: Fri, 24 Apr 2009 11:46:31 -0400
Message-ID: <dcad22d80904240846m3d8fa9d3maebd27cb0ce5b47a@mail.gmail.com>
From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: IETF AVT WG <avt@ietf.org>
Content-Type: multipart/alternative; boundary=001485f90f44887fa904684ee66b
Subject: [AVT] Fwd: [Fecframe] Working Group Last Call on draft-ietf-fecframe-config-signaling
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I apologize for not forwarding this to AVT, but I think it may be of
interest here.
Regards
Marshall

---------- Forwarded message ----------
From: Marshall Eubanks <tme@americafree.tv>
Date: Fri, Apr 24, 2009 at 11:44 AM
Subject: Fwd: [Fecframe] Working Group Last Call on
draft-ietf-fecframe-config-signaling
To: "marshall.eubanks@gmail.com Eubanks" <marshall.eubanks@gmail.com>




Begin forwarded message:

 From: Marshall Eubanks <tme@multicasttech.com>
> Date: April 18, 2009 2:04:14 PM EDT
> To: fecframe@ietf.org
> Subject: [Fecframe] Working Group Last Call on
> draft-ietf-fecframe-config-signaling
>
> I think that this document is also ready, so I would like to issue
> a working group last call on draft-ietf-fecframe-config-signaling-01. This
> document can be
> found at
>
> http://tools.ietf.org/id/draft-ietf-fecframe-config-signaling-01.txt
>
> I encourage everyone to read this document and comment during the last call
> period.
>
> This last call will end on Monday, May 4th, 2009.
>
> Regards
> Marshall Eubanks
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe
>
>
Regards
Marshall Eubanks
CEO / AmericaFree.TV

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I apologize for not forwarding this to AVT, but I think it may be of intere=
st here.=A0<div><br></div><div>Regards</div><div>Marshall<br><br><div class=
=3D"gmail_quote">---------- Forwarded message ----------<br>From: <b class=
=3D"gmail_sendername">Marshall Eubanks</b> <span dir=3D"ltr">&lt;<a href=3D=
"mailto:tme@americafree.tv">tme@americafree.tv</a>&gt;</span><br>
Date: Fri, Apr 24, 2009 at 11:44 AM<br>Subject: Fwd: [Fecframe] Working Gro=
up Last Call on draft-ietf-fecframe-config-signaling<br>To: &quot;<a href=
=3D"mailto:marshall.eubanks@gmail.com">marshall.eubanks@gmail.com</a> Euban=
ks&quot; &lt;<a href=3D"mailto:marshall.eubanks@gmail.com">marshall.eubanks=
@gmail.com</a>&gt;<br>
<br><br><br>
<br>
Begin forwarded message:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
From: Marshall Eubanks &lt;<a href=3D"mailto:tme@multicasttech.com" target=
=3D"_blank">tme@multicasttech.com</a>&gt;<br>
Date: April 18, 2009 2:04:14 PM EDT<br>
To: <a href=3D"mailto:fecframe@ietf.org" target=3D"_blank">fecframe@ietf.or=
g</a><br>
Subject: [Fecframe] Working Group Last Call on draft-ietf-fecframe-config-s=
ignaling<br>
<br>
I think that this document is also ready, so I would like to issue<br>
a working group last call on draft-ietf-fecframe-config-signaling-01. This =
document can be<br>
found at<br>
<br>
<a href=3D"http://tools.ietf.org/id/draft-ietf-fecframe-config-signaling-01=
.txt" target=3D"_blank">http://tools.ietf.org/id/draft-ietf-fecframe-config=
-signaling-01.txt</a><br>
<br>
I encourage everyone to read this document and comment during the last call=
 period.<br>
<br>
This last call will end on Monday, May 4th, 2009.<br>
<br>
Regards<br>
Marshall Eubanks<br>
_______________________________________________<br>
Fecframe mailing list<br>
<a href=3D"mailto:Fecframe@ietf.org" target=3D"_blank">Fecframe@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/fecframe" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/fecframe</a><br>
<br>
</blockquote>
<br>
Regards<br><font color=3D"#888888">
Marshall Eubanks<br>
CEO / AmericaFree.TV<br>
<br>
</font></div><br></div>

--001485f90f44887fa904684ee66b--

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Date: Fri, 24 Apr 2009 11:47:20 -0400
Message-ID: <dcad22d80904240847s55a75c96o8e175273f1f7e7a0@mail.gmail.com>
From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: IETF AVT WG <avt@ietf.org>
Content-Type: multipart/alternative; boundary=001485f7c5a07ea52d04684ee992
Subject: [AVT] Fwd: [Fecframe] Working Group Last Call on draft-ietf-fecframe-framework
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This should certainly be of interest here and I apologize for not forwarding
it previously.
Regards
Marshall

---------- Forwarded message ----------
From: Marshall Eubanks <tme@americafree.tv>
Date: Fri, Apr 24, 2009 at 11:43 AM
Subject: Fwd: [Fecframe] Working Group Last Call on
draft-ietf-fecframe-framework
To: "marshall.eubanks@gmail.com Eubanks" <marshall.eubanks@gmail.com>




Begin forwarded message:

 From: Marshall Eubanks <tme@multicasttech.com>
> Date: April 18, 2009 1:58:27 PM EDT
> To: fecframe@ietf.org
> Subject: [Fecframe] Working Group Last Call on
> draft-ietf-fecframe-framework
>
> Based on the discussion at the FECFRAME meeting in San Francisco, I would
> like to
> issue a working group last call on draft-ietf-fecframe-framework-03. This
> document can be
> found at
>
> http://www.ietf.org/internet-drafts/draft-ietf-fecframe-framework-03.txt
>
> This is an essential document for the Fecframe work, so I would like to
> encourage everyone to
> read it again and comment.
>
> This last call will end on Monday, May 4th, 2009.
>
> Regards
> Marshall Eubanks
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe
>
>
Regards
Marshall Eubanks
CEO / AmericaFree.TV

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This should certainly be of interest here and I apologize for not forwardin=
g it previously.<div><br></div><div>Regards</div><div>Marshall<br><br><div =
class=3D"gmail_quote">---------- Forwarded message ----------<br>From: <b c=
lass=3D"gmail_sendername">Marshall Eubanks</b> <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:tme@americafree.tv">tme@americafree.tv</a>&gt;</span><br>
Date: Fri, Apr 24, 2009 at 11:43 AM<br>Subject: Fwd: [Fecframe] Working Gro=
up Last Call on draft-ietf-fecframe-framework<br>To: &quot;<a href=3D"mailt=
o:marshall.eubanks@gmail.com">marshall.eubanks@gmail.com</a> Eubanks&quot; =
&lt;<a href=3D"mailto:marshall.eubanks@gmail.com">marshall.eubanks@gmail.co=
m</a>&gt;<br>
<br><br><br>
<br>
Begin forwarded message:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
From: Marshall Eubanks &lt;<a href=3D"mailto:tme@multicasttech.com" target=
=3D"_blank">tme@multicasttech.com</a>&gt;<br>
Date: April 18, 2009 1:58:27 PM EDT<br>
To: <a href=3D"mailto:fecframe@ietf.org" target=3D"_blank">fecframe@ietf.or=
g</a><br>
Subject: [Fecframe] Working Group Last Call on draft-ietf-fecframe-framewor=
k<br>
<br>
Based on the discussion at the FECFRAME meeting in San Francisco, I would l=
ike to<br>
issue a working group last call on draft-ietf-fecframe-framework-03. This d=
ocument can be<br>
found at<br>
<br>
<a href=3D"http://www.ietf.org/internet-drafts/draft-ietf-fecframe-framewor=
k-03.txt" target=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-=
fecframe-framework-03.txt</a><br>
<br>
This is an essential document for the Fecframe work, so I would like to enc=
ourage everyone to<br>
read it again and comment.<br>
<br>
This last call will end on Monday, May 4th, 2009.<br>
<br>
Regards<br>
Marshall Eubanks<br>
_______________________________________________<br>
Fecframe mailing list<br>
<a href=3D"mailto:Fecframe@ietf.org" target=3D"_blank">Fecframe@ietf.org</a=
><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/fecframe" target=3D"_blank=
">https://www.ietf.org/mailman/listinfo/fecframe</a><br>
<br>
</blockquote>
<br>
Regards<br><font color=3D"#888888">
Marshall Eubanks<br>
CEO / AmericaFree.TV<br>
<br>
</font></div><br></div>

--001485f7c5a07ea52d04684ee992--

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Hi Roni,=20

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Roni Even
> Sent: Friday, April 24, 2009 4:24 AM
> To: avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Hi,
>=20
> I think that the current draft does not give a description of=20
> a system that works since there is no text explaining how the=20

What do you mean by "a system that works"?=20

> RS knows the unicast transport address on the RR to where to=20
> send the stream.=20

Once RS receives the request packet from an RR, RS knows its address. =
Ports are defined in the SDP. If you are asking about "NAT" issues, we =
have a section for it, and we plan to complete it as we move forward. It =
is not as critical as the other parts for now.
=20
> If you mandate the use of RTP/RTCP mux it should say so=20
> otherwise the RAMS-R should have an optional parameter that=20
> supplies this information and a flag for using RTP/RTCP mux.

IMO, we cannot mandate muxing as it is not required to make RAMS work. =
If muxing is supported, the SDP should reflect it.

BR,
-acbegen
=20
> Thanks
>=20
> Roni Even
>=20
> =20
>=20
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Bill Ver Steeg (versteb)
> Sent: Thursday, April 16, 2009 7:39 PM
> To: avt@ietf.org
> Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> RTP Sessions (RAMS)" available
>=20
> =20
>=20
> There is a new draft of the "Rapid Acquisition of Multicast=20
> RTP Sessions (RAMS)" draft available at=20
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
ynchronization-for-rtp-03.txt <http://www.ietf.org/internet-> =
drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
>=20
> =20
>=20
> We have incorporated the changes from the technical breakout=20
> session in San Francisco. The major changes in this version=20
> of the draft include=20
>=20
> 1- Changing the document title to avoid confusion with other=20
> ongoing "synchronization" drafts
>=20
> 2- changing the message names to reflect the title change
>=20
> 3- clarification of the RTCP message semantics, including=20
> changes to the "Request" and "Inform" messages
>=20
> 4- additional description/motivation for the various message=20
> flows has been added
>=20
> 5- RTP/RTCP muxing has been added
>=20
> =20
>=20
> =20
>=20
> We hope to make this a Working Group item, and will change=20
> the name of the draft to avoid conflicts with other=20
> "synchronization" drafts at that time.
>=20
> =20
>=20
> Bill VerSteeg
>=20
> =20
>=20
> =20
>=20
>=20

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Thanks for the review. See inline.=20

> 1. In section 1 you reference RFC2733, I think it was=20
> obsolete by RFC 5109 ,
> do you need this downref?

Yes, this has been already discussed within fecframe and with Magnus. We =
give an informative reference as we are using an extension of the header =
RFC 2733 had.
=20
> 2. In the registration section 5.1, the comments applies to=20
> all subtypes: =20
>   2.1  You have for security considerations and published=20
> specification
> "this document" it is better to write RFC xxxx and have a=20
> note to     	the
> RFC editor to replace it with the RFC number allocated, this=20
> will show in
> the IANA registry with the write pointer.

Will do.
=20
>   2.2 In the restriction on usage I assume that the document=20
> is using RTP
> framing and is not specified for other types, if this is the 	case it
> should say so. E.g. "   This media type depends on RTP=20
> framing, and hence is
> only defined for transfer via RTP [RFC3550].  =09
> Transport within other
> framing  protocols is not defined at this time"

Will do.

>   2.3 You need to add the "additional information" that is in=20
> the template
> probably saying none

I believe I already have "additional information: none" for each =
subtype.

> 3. In section 5.2.1 it looks like you require symmetry=20
> between the send and
> received stream. Is this your requirement.=20

I am not sure what "symmetry" means here. Could you explain?
=20
> 4. In section 5.2.1 can the answerer send a response with no=20
> FEC at all even
> if not offered.

If the answerer does not want FEC, yes, it can drop FEC from the offer. =
Let me make that clear in the text.

=20
> 5. In section 5.2.2 Can the receiver just ignore the fec=20
> stream, the text
> has a MUST use the configuration that is offered which can=20
> hint that it MUST
> use FEC.

The receiver may decide to receive or not receive the FEC stream. If it =
decides to receive, it MUST use the configuration that is provided for =
the session. I guess the text is clear about this point.

=20
> 6. In section 7 you discuss an exchange using grouping. Is it=20
> also allowed
> to use FEC withoutout grouping using the video or audio media=20
> types like
> section 14 in RFC 5109

Per FEC framework, we require the FEC to be in a separate flow, which =
requires the source and FEC streams to be on different RTP sessions. =
Thus, we require grouping (RFC 4756 or 4756bis)

BR,
-acbegen
=20
>=20
> Thanks
> Roni Even
> =20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Ali C.
> Begen (abegen)
> Sent: Sunday, April 19, 2009 12:04 AM
> To: fecframe@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] FW: New Version Notification for
> draft-ietf-fecframe-interleaved-fec-scheme-03
>=20
> This update addresses all the comments received in the WGLC=20
> period initated
> on April 1st. If there are further comments, please post them=20
> soon as we
> would like to move forward with this draft.
> http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interl
eaved-fec-sche
> me-03.txt
>=20
> BR,
> -acbegen=20
>=20
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]=20
> Sent: Saturday, April 18, 2009 1:58 PM
> To: Ali C. Begen (abegen)
> Subject: New Version Notification for
> draft-ietf-fecframe-interleaved-fec-scheme-03=20
>=20
>=20
> A new version of I-D,=20
> draft-ietf-fecframe-interleaved-fec-scheme-03.txt has
> been successfuly submitted by Ali Begen and posted to the=20
> IETF repository.
>=20
> Filename:	 draft-ietf-fecframe-interleaved-fec-scheme
> Revision:	 03
> Title:		 RTP Payload Format for 1-D Interleaved=20
> Parity FEC
> Creation_date:	 2009-04-18
> WG ID:		 fecframe
> Number_of_pages: 31
>=20
> Abstract:
> This document defines a new RTP payload format for the Forward Error
> Correction (FEC) that is generated by the 1-D interleaved parity code
> from a source media encapsulated in RTP.  The 1-D interleaved parity
> code is a systematic code, where a number of repair symbols are
> generated from a set of source symbols and sent in a repair flow
> separate from the source flow that carries the source symbols.  The
> 1-D interleaved parity code offers a good protection against bursty
> packet losses at a cost of decent complexity.  The new payload format
> defined in this document is used (with some exceptions) as a part of
> the DVB Application-layer FEC specification.
> =20
>=20
>=20
>=20
> The IETF Secretariat.
>=20
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20

From ron.even.tlv@gmail.com  Fri Apr 24 11:44:44 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com>
Date: Fri, 24 Apr 2009 21:44:01 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
I will try to explain in simple way

When the RS receives the RAMS-R it need to start sending an RTP stream to
the RR. 
In order to send a unicast packet, the RS needs to know a transport address
on the RR to where the RTP stream will be sent. The current draft does not
say how the RS knows this address. There is no SDP from RR to RS like you
mention in your response.
This is why I say that the current draft does not specify a solution that
can be implemented as a working solution
Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Friday, April 24, 2009 7:38 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Hi Roni, 

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Roni Even
> Sent: Friday, April 24, 2009 4:24 AM
> To: avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Hi,
> 
> I think that the current draft does not give a description of 
> a system that works since there is no text explaining how the 

What do you mean by "a system that works"? 

> RS knows the unicast transport address on the RR to where to 
> send the stream. 

Once RS receives the request packet from an RR, RS knows its address. Ports
are defined in the SDP. If you are asking about "NAT" issues, we have a
section for it, and we plan to complete it as we move forward. It is not as
critical as the other parts for now.
 
> If you mandate the use of RTP/RTCP mux it should say so 
> otherwise the RAMS-R should have an optional parameter that 
> supplies this information and a flag for using RTP/RTCP mux.

IMO, we cannot mandate muxing as it is not required to make RAMS work. If
muxing is supported, the SDP should reflect it.

BR,
-acbegen
 
> Thanks
> 
> Roni Even
> 
>  
> 
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Bill Ver Steeg (versteb)
> Sent: Thursday, April 16, 2009 7:39 PM
> To: avt@ietf.org
> Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> RTP Sessions (RAMS)" available
> 
>  
> 
> There is a new draft of the "Rapid Acquisition of Multicast 
> RTP Sessions (RAMS)" draft available at 
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> 
>  
> 
> We have incorporated the changes from the technical breakout 
> session in San Francisco. The major changes in this version 
> of the draft include 
> 
> 1- Changing the document title to avoid confusion with other 
> ongoing "synchronization" drafts
> 
> 2- changing the message names to reflect the title change
> 
> 3- clarification of the RTCP message semantics, including 
> changes to the "Request" and "Inform" messages
> 
> 4- additional description/motivation for the various message 
> flows has been added
> 
> 5- RTP/RTCP muxing has been added
> 
>  
> 
>  
> 
> We hope to make this a Working Group item, and will change 
> the name of the draft to avoid conflicts with other 
> "synchronization" drafts at that time.
> 
>  
> 
> Bill VerSteeg
> 
>  
> 
>  
> 
> 


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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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Roni-
=20
Yes, the document needs updating in this respect. We have two broad
options which need to be described in detail.
=20
The method that I have been assuming does use muxing. The RS sends the
unicast burst back to the UDP port number on the RR that sent the
RAMS-R. This allows transparent operation through NAT. There is a single
UDP port, and the transaction is initiated by the "client", so it is NAT
friendly.
=20
There are quite a few ways we could handle the case in which muxing is
not used. We could add a parameter to the RAMS-R, use another RTCP
packet in the same compound packet, define the port numbers as a
relationship to the opened port (not attractive), or define something in
the SDP (also not attractive). I personally like the "RAMS-R optional
field" approach, but I think this should be a topic that we consider in
detail once we have a WG item.
=20
=20
=20
=20
bvs
=20

________________________________

From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
Sent: Friday, April 24, 2009 7:24 AM
To: avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP
Sessions (RAMS)" available



Hi,

I think that the current draft does not give a description of a system
that works since there is no text explaining how the RS knows the
unicast transport address on the RR to where to send the stream.=20

If you mandate the use of RTP/RTCP mux it should say so otherwise the
RAMS-R should have an optional parameter that supplies this information
and a flag for using RTP/RTCP mux.

Thanks

Roni Even

=20

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Bill Ver Steeg (versteb)
Sent: Thursday, April 16, 2009 7:39 PM
To: avt@ietf.org
Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

=20

There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniza
tion-for-rtp-03.txt
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniz
ation-for-rtp-03.txt>=20

=20

We have incorporated the changes from the technical breakout session in
San Francisco. The major changes in this version of the draft include=20

1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts

2- changing the message names to reflect the title change

3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages

4- additional description/motivation for the various message flows has
been added

5- RTP/RTCP muxing has been added

=20

=20

We hope to make this a Working Group item, and will change the name of
the draft to avoid conflicts with other "synchronization" drafts at that
time.

=20

Bill VerSteeg

=20

=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Roni-</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Yes, the document needs updating in this =
respect. We have=20
two broad options which need to be described in =
detail.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The method that I have been assuming does use =
muxing.=20
The&nbsp;RS sends the unicast burst back to the UDP port number on the =
RR that=20
sent the RAMS-R. This allows transparent operation through NAT. There is =
a=20
single UDP port, and the transaction is initiated by the "client", so it =
is NAT=20
friendly.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>There are quite a few ways we could handle the =
case in=20
which muxing is not used. We could add a parameter to the RAMS-R, use =
another=20
RTCP packet in the same compound packet, define the port numbers as a=20
relationship to the opened port (not attractive), or define something in =
the SDP=20
(also not attractive). I personally like the "RAMS-R optional field" =
approach,=20
but I think this should be a topic that we consider in detail once we =
have a WG=20
item.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D734475517-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Roni Even =
[mailto:ron.even.tlv@gmail.com]=20
<BR><B>Sent:</B> Friday, April 24, 2009 7:24 AM<BR><B>To:</B>=20
avt@ietf.org<BR><B>Cc:</B> Bill Ver Steeg (versteb)<BR><B>Subject:</B> =
RE: [AVT]=20
New draft of "Rapid Acquisition of Multicast RTP Sessions (RAMS)"=20
available<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Hi,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">I=20
think that the current draft does not give a description of a system =
that works=20
since there is no text explaining how the RS knows the unicast transport =
address=20
on the RR to where to send the stream. <o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">If=20
you mandate the use of RTP/RTCP mux it should say so otherwise the =
RAMS-R should=20
have an optional parameter that supplies this information and a flag for =
using=20
RTP/RTCP mux.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Thanks<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Roni=20
Even<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
#b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: =
medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">=20
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <B>On Behalf Of =
</B>Bill Ver=20
Steeg (versteb)<BR><B>Sent:</B> Thursday, April 16, 2009 7:39 =
PM<BR><B>To:</B>=20
avt@ietf.org<BR><B>Subject:</B> [AVT] New draft of "Rapid Acquisition of =

Multicast RTP Sessions (RAMS)" =
available<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">There is a =
new draft=20
of the "Rapid Acquisition of Multicast RTP Sessions (RAMS)" draft =
available at=20
<A=20
href=3D"http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-sync=
hronization-for-rtp-03.txt"><SPAN=20
style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New =
Roman','serif'">http://www.ietf.org/internet-drafts/draft-versteeg-avt-ra=
pid-synchronization-for-rtp-03.txt</SPAN></A></SPAN><o:p></o:p></P></DIV>=

<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">We=20
have&nbsp;incorporated the changes from the technical breakout session =
in San=20
Francisco. The major changes in this version of the draft include=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">1- Changing =
the=20
document title to avoid confusion with other ongoing "synchronization"=20
drafts</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">2- changing =
the=20
message names to reflect the title change</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">3- =
clarification of=20
the RTCP message semantics, including&nbsp;changes to the&nbsp;"Request" =
and=20
"Inform" messages</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">4- =
additional=20
description/motivation for the various message flows has been=20
added</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">5- RTP/RTCP =
muxing=20
has been added</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">We hope to =
make this=20
a Working Group item, and will change the name of the draft to avoid =
conflicts=20
with other "synchronization" drafts at that =
time.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Bill=20
VerSteeg</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV></DIV></BODY></HTML>

------_=_NextPart_001_01C9C50D.87FA4A67--

From abegen@cisco.com  Fri Apr 24 11:59:37 2009
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com>
From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Oh, I see. The burst goes to the port that we would normally send the =
retransmissions. For example, consider the SDP from the draft:

        m=3Dvideo 41000 RTP/AVPF 98
        i=3DPrimary Multicast Stream #2
        c=3DIN IP4 224.1.1.2/255
        a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
        a=3Drtpmap:98 MP2T/90000
        a=3Drtcp:41001 IN IP4 192.0.2.1
        a=3Drtcp-fb:98 nack
        a=3Drtcp-fb:98 nack ssli
        a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
        a=3Dmid:3
        m=3Dvideo 41002 RTP/AVPF 99
        i=3DUnicast Retransmission Stream #2 (Ret. and Rapid Acq. =
Support)
        c=3DIN IP4 192.0.2.1
        a=3Drecvonly
        a=3Drtpmap:99 rtx/90000
        a=3Drtcp:41003
        a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
        a=3Dmid:4

In this case, the burst will go to port 41002 on the RR. The RAMS-I =
message(s), which is an RTCP feedback message, will go to port 41003.

HTH,
-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Friday, April 24, 2009 11:44 AM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Ali,
> I will try to explain in simple way
>=20
> When the RS receives the RAMS-R it need to start sending an=20
> RTP stream to
> the RR.=20
> In order to send a unicast packet, the RS needs to know a=20
> transport address
> on the RR to where the RTP stream will be sent. The current=20
> draft does not
> say how the RS knows this address. There is no SDP from RR to=20
> RS like you
> mention in your response.
> This is why I say that the current draft does not specify a=20
> solution that
> can be implemented as a working solution
> Roni
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Friday, April 24, 2009 7:38 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> Hi Roni,=20
>=20
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > Behalf Of Roni Even
> > Sent: Friday, April 24, 2009 4:24 AM
> > To: avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Hi,
> >=20
> > I think that the current draft does not give a description of=20
> > a system that works since there is no text explaining how the=20
>=20
> What do you mean by "a system that works"?=20
>=20
> > RS knows the unicast transport address on the RR to where to=20
> > send the stream.=20
>=20
> Once RS receives the request packet from an RR, RS knows its=20
> address. Ports
> are defined in the SDP. If you are asking about "NAT" issues,=20
> we have a
> section for it, and we plan to complete it as we move=20
> forward. It is not as
> critical as the other parts for now.
> =20
> > If you mandate the use of RTP/RTCP mux it should say so=20
> > otherwise the RAMS-R should have an optional parameter that=20
> > supplies this information and a flag for using RTP/RTCP mux.
>=20
> IMO, we cannot mandate muxing as it is not required to make=20
> RAMS work. If
> muxing is supported, the SDP should reflect it.
>=20
> BR,
> -acbegen
> =20
> > Thanks
> >=20
> > Roni Even
> >=20
> > =20
> >=20
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > Behalf Of Bill Ver Steeg (versteb)
> > Sent: Thursday, April 16, 2009 7:39 PM
> > To: avt@ietf.org
> > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > RTP Sessions (RAMS)" available
> >=20
> > =20
> >=20
> > There is a new draft of the "Rapid Acquisition of Multicast=20
> > RTP Sessions (RAMS)" draft available at=20
> > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> >=20
> > =20
> >=20
> > We have incorporated the changes from the technical breakout=20
> > session in San Francisco. The major changes in this version=20
> > of the draft include=20
> >=20
> > 1- Changing the document title to avoid confusion with other=20
> > ongoing "synchronization" drafts
> >=20
> > 2- changing the message names to reflect the title change
> >=20
> > 3- clarification of the RTCP message semantics, including=20
> > changes to the "Request" and "Inform" messages
> >=20
> > 4- additional description/motivation for the various message=20
> > flows has been added
> >=20
> > 5- RTP/RTCP muxing has been added
> >=20
> > =20
> >=20
> > =20
> >=20
> > We hope to make this a Working Group item, and will change=20
> > the name of the draft to avoid conflicts with other=20
> > "synchronization" drafts at that time.
> >=20
> > =20
> >=20
> > Bill VerSteeg
> >=20
> > =20
> >=20
> > =20
> >=20
> >=20
>=20
>=20

From ron.even.tlv@gmail.com  Fri Apr 24 12:05:24 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <fecframe@ietf.org>
References: <04CAD96D4C5A3D48B1919248A8FE0D54091BAA88@xmb-sjc-215.amer.cisco.com> <49f1cd82.15185e0a.0fc3.ffffb9e7@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80EF@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C80EF@xmb-sjc-215.amer.cisco.com>
Date: Fri, 24 Apr 2009 22:03:13 +0300
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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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Ali,
Inline
Roni

>   2.3 You need to add the "additional information" that is in 
> the template
> probably saying none

I believe I already have "additional information: none" for each subtype.

RE- The order was different so I missed it

> 3. In section 5.2.1 it looks like you require symmetry 
> between the send and
> received stream. Is this your requirement. 

I am not sure what "symmetry" means here. Could you explain?

RE - Symmetry means that the offerer and answerer MUST use the same mode
since this exchange results in a two streams, one from each direction.
 

 
> 5. In section 5.2.2 Can the receiver just ignore the fec 
> stream, the text
> has a MUST use the configuration that is offered which can 
> hint that it MUST
> use FEC.

The receiver may decide to receive or not receive the FEC stream. If it
decides to receive, it MUST use the configuration that is provided for the
session. I guess the text is clear about this point.

RE- I think it has to do with my assumption that you can multiplex the
streams like RFC 5109

 
> 6. In section 7 you discuss an exchange using grouping. Is it 
> also allowed
> to use FEC withoutout grouping using the video or audio media 
> types like
> section 14 in RFC 5109

Per FEC framework, we require the FEC to be in a separate flow, which
requires the source and FEC streams to be on different RTP sessions. Thus,
we require grouping (RFC 4756 or 4756bis)

RE- I will have to look at that draft, this is different from RFC 5109 and
RFC 2733

BR,
-acbegen
 
> 
> Thanks
> Roni Even
>  
> 
> 
> 
> 
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Ali C.
> Begen (abegen)
> Sent: Sunday, April 19, 2009 12:04 AM
> To: fecframe@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] FW: New Version Notification for
> draft-ietf-fecframe-interleaved-fec-scheme-03
> 
> This update addresses all the comments received in the WGLC 
> period initated
> on April 1st. If there are further comments, please post them 
> soon as we
> would like to move forward with this draft.
> http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interl
eaved-fec-sche
> me-03.txt
> 
> BR,
> -acbegen 
> 
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
> Sent: Saturday, April 18, 2009 1:58 PM
> To: Ali C. Begen (abegen)
> Subject: New Version Notification for
> draft-ietf-fecframe-interleaved-fec-scheme-03 
> 
> 
> A new version of I-D, 
> draft-ietf-fecframe-interleaved-fec-scheme-03.txt has
> been successfuly submitted by Ali Begen and posted to the 
> IETF repository.
> 
> Filename:	 draft-ietf-fecframe-interleaved-fec-scheme
> Revision:	 03
> Title:		 RTP Payload Format for 1-D Interleaved 
> Parity FEC
> Creation_date:	 2009-04-18
> WG ID:		 fecframe
> Number_of_pages: 31
> 
> Abstract:
> This document defines a new RTP payload format for the Forward Error
> Correction (FEC) that is generated by the 1-D interleaved parity code
> from a source media encapsulated in RTP.  The 1-D interleaved parity
> code is a systematic code, where a number of repair symbols are
> generated from a set of source symbols and sent in a repair flow
> separate from the source flow that carries the source symbols.  The
> 1-D interleaved parity code offers a good protection against bursty
> packet losses at a cost of decent complexity.  The new payload format
> defined in this document is used (with some exceptions) as a part of
> the DVB Application-layer FEC specification.
>  
> 
> 
> 
> The IETF Secretariat.
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
> 
> 


From ron.even.tlv@gmail.com  Fri Apr 24 12:19:48 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <81B9B88E2F9D574AA5328A85547CDE9101153491@xmb-rtp-21d.amer.cisco.com>
In-Reply-To: <81B9B88E2F9D574AA5328A85547CDE9101153491@xmb-rtp-21d.amer.cisco.com>
Date: Fri, 24 Apr 2009 22:19:04 +0300
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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Bill,

 

Since I have two hats here then as a WG chair, my intention is to ask the
list to start this topic as a WG item

 

Taking my WG chair off, my expectation is that a WG document will describe a
solution that can be implemented while the current text does not specify how
to make it work. I do not care if you write that RTP/RTCP mux MUST be used
by the RS and supported by the RR and update the SDP in the example to
specify that this is case.  

 

My personal preference is to have the RS indicate in the SDP that it wants
to use multiplexing and to add a flag to RAMS-R that will indicate if
RTP/RTCP mux is supported by RR and can be used. Only if the RR cannot
support the multiplexing it should indicate it using the flag and add to the
RAMS-R the transport address to be used (Note that it can be v4 or v6).

SDP cannot be used since there is no SDP from RR to RS.

 

Roni Even

 

From: Bill Ver Steeg (versteb) [mailto:versteb@cisco.com] 
Sent: Friday, April 24, 2009 9:50 PM
To: Roni Even; avt@ietf.org
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

 

Roni-

 

Yes, the document needs updating in this respect. We have two broad options
which need to be described in detail.

 

The method that I have been assuming does use muxing. The RS sends the
unicast burst back to the UDP port number on the RR that sent the RAMS-R.
This allows transparent operation through NAT. There is a single UDP port,
and the transaction is initiated by the "client", so it is NAT friendly.

 

There are quite a few ways we could handle the case in which muxing is not
used. We could add a parameter to the RAMS-R, use another RTCP packet in the
same compound packet, define the port numbers as a relationship to the
opened port (not attractive), or define something in the SDP (also not
attractive). I personally like the "RAMS-R optional field" approach, but I
think this should be a topic that we consider in detail once we have a WG
item.

 

 

 

 

bvs

 

 

  _____  

From: Roni Even [mailto:ron.even.tlv@gmail.com] 
Sent: Friday, April 24, 2009 7:24 AM
To: avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

Hi,

I think that the current draft does not give a description of a system that
works since there is no text explaining how the RS knows the unicast
transport address on the RR to where to send the stream. 

If you mandate the use of RTP/RTCP mux it should say so otherwise the RAMS-R
should have an optional parameter that supplies this information and a flag
for using RTP/RTCP mux.

Thanks

Roni Even

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Bill
Ver Steeg (versteb)
Sent: Thursday, April 16, 2009 7:39 PM
To: avt@ietf.org
Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

 

There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronizatio
n-for-rtp-03.txt>
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchronization
-for-rtp-03.txt

 

We have incorporated the changes from the technical breakout session in San
Francisco. The major changes in this version of the draft include 

1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts

2- changing the message names to reflect the title change

3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages

4- additional description/motivation for the various message flows has been
added

5- RTP/RTCP muxing has been added

 

 

We hope to make this a Working Group item, and will change the name of the
draft to avoid conflicts with other "synchronization" drafts at that time.

 

Bill VerSteeg

 

 


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<body lang=3DEN-US link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Bill,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Since I have two hats here then as a WG chair, my =
intention is
to ask the list to start this topic as a WG item<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Taking my WG chair off, my expectation is that a WG =
document will
describe a solution that can be implemented while the current text does =
not specify
how to make it work. I do not care if you write that RTP/RTCP mux MUST =
be used by
the RS and supported by the RR and update the SDP in the example to =
specify
that this is case. &nbsp;<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>My personal preference is to have the RS indicate in the =
SDP
that it wants to use multiplexing and to add a flag to RAMS-R that will
indicate if RTP/RTCP mux is supported by RR and can be used. Only if the =
RR cannot
support the multiplexing it should indicate it using the flag and add to =
the
RAMS-R the transport address to be used (Note that it can be v4 or =
v6).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>SDP cannot be used since there is no SDP from RR to =
RS.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Bill Ver =
Steeg
(versteb) [mailto:versteb@cisco.com] <br>
<b>Sent:</b> Friday, April 24, 2009 9:50 PM<br>
<b>To:</b> Roni Even; avt@ietf.org<br>
<b>Subject:</b> RE: [AVT] New draft of &quot;Rapid Acquisition of =
Multicast RTP
Sessions (RAMS)&quot; available<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Roni-</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>Yes, the document needs updating in this respect. We have =
two broad
options which need to be described in detail.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>The method that I have been assuming does use muxing. =
The&nbsp;RS
sends the unicast burst back to the UDP port number on the RR that sent =
the
RAMS-R. This allows transparent operation through NAT. There is a single =
UDP
port, and the transaction is initiated by the &quot;client&quot;, so it =
is NAT
friendly.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>There are quite a few ways we could handle the case in which =
muxing
is not used. We could add a parameter to the RAMS-R, use another RTCP =
packet in
the same compound packet, define the port numbers as a relationship to =
the
opened port (not attractive), or define something in the SDP (also not
attractive). I personally like the &quot;RAMS-R optional field&quot; =
approach,
but I think this should be a topic that we consider in detail once we =
have a WG
item.</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";
color:blue'>bvs</span><o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'>

<hr size=3D2 width=3D"100%" align=3Dcenter>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'>From:</span></b><span =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Roni Even =
[mailto:ron.even.tlv@gmail.com] <br>
<b>Sent:</b> Friday, April 24, 2009 7:24 AM<br>
<b>To:</b> avt@ietf.org<br>
<b>Cc:</b> Bill Ver Steeg (versteb)<br>
<b>Subject:</b> RE: [AVT] New draft of &quot;Rapid Acquisition of =
Multicast RTP
Sessions (RAMS)&quot; available</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think that the current draft does not give a =
description of a
system that works since there is no text explaining how the RS knows the
unicast transport address on the RR to where to send the stream. =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>If you mandate the use of RTP/RTCP mux it should say so
otherwise the RAMS-R should have an optional parameter that supplies =
this
information and a flag for using RTP/RTCP mux.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of =
</b>Bill Ver
Steeg (versteb)<br>
<b>Sent:</b> Thursday, April 16, 2009 7:39 PM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> [AVT] New draft of &quot;Rapid Acquisition of Multicast =
RTP
Sessions (RAMS)&quot; available<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>There
is a new draft of the &quot;Rapid Acquisition of Multicast RTP Sessions
(RAMS)&quot; draft available at <a
href=3D"http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-sync=
hronization-for-rtp-03.txt"><span
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>http://www.ietf.org/internet-drafts/draft-versteeg-avt-ra=
pid-synchronization-for-rtp-03.txt</span></a></span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We
have&nbsp;incorporated the changes from the technical breakout session =
in San
Francisco. The major changes in this version of the draft include =
</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>1-
Changing the document title to avoid confusion with other ongoing
&quot;synchronization&quot; drafts</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>2-
changing the message names to reflect the title =
change</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>3-
clarification of the RTCP message semantics, including&nbsp;changes to
the&nbsp;&quot;Request&quot; and &quot;Inform&quot; =
messages</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>4-
additional description/motivation for the various message flows has been =
added</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>5-
RTP/RTCP muxing has been added</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>We
hope to make this a Working Group item, and will change the name of the =
draft
to avoid conflicts with other &quot;synchronization&quot; drafts at that =
time.</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Bill
VerSteeg</span><o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

<div>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

</div>

</div>

</body>

</html>

------=_NextPart_000_00E4_01C9C52A.B5D2D000--


From ron.even.tlv@gmail.com  Fri Apr 24 12:23:28 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com>
Date: Fri, 24 Apr 2009 22:22:37 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
I think you get it wrong, this SDP is from the RS and not the RR so the RS
cannot specify to which address it will send it can only specify to which
address it can receive RTP stream. In this SDP the relevant information is
that the request for retransmission will be sent by the RR to port 41003
Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Friday, April 24, 2009 10:01 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Oh, I see. The burst goes to the port that we would normally send the
retransmissions. For example, consider the SDP from the draft:

        m=video 41000 RTP/AVPF 98
        i=Primary Multicast Stream #2
        c=IN IP4 224.1.1.2/255
        a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
        a=rtpmap:98 MP2T/90000
        a=rtcp:41001 IN IP4 192.0.2.1
        a=rtcp-fb:98 nack
        a=rtcp-fb:98 nack ssli
        a=ssrc:123321 cname:iptv-ch32@rams.example.com
        a=mid:3
        m=video 41002 RTP/AVPF 99
        i=Unicast Retransmission Stream #2 (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1
        a=recvonly
        a=rtpmap:99 rtx/90000
        a=rtcp:41003
        a=fmtp:99 apt=98; rtx-time=5000
        a=mid:4

In this case, the burst will go to port 41002 on the RR. The RAMS-I
message(s), which is an RTCP feedback message, will go to port 41003.

HTH,
-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Friday, April 24, 2009 11:44 AM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> I will try to explain in simple way
> 
> When the RS receives the RAMS-R it need to start sending an 
> RTP stream to
> the RR. 
> In order to send a unicast packet, the RS needs to know a 
> transport address
> on the RR to where the RTP stream will be sent. The current 
> draft does not
> say how the RS knows this address. There is no SDP from RR to 
> RS like you
> mention in your response.
> This is why I say that the current draft does not specify a 
> solution that
> can be implemented as a working solution
> Roni
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Friday, April 24, 2009 7:38 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> Hi Roni, 
> 
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > Behalf Of Roni Even
> > Sent: Friday, April 24, 2009 4:24 AM
> > To: avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions (RAMS)" available
> > 
> > Hi,
> > 
> > I think that the current draft does not give a description of 
> > a system that works since there is no text explaining how the 
> 
> What do you mean by "a system that works"? 
> 
> > RS knows the unicast transport address on the RR to where to 
> > send the stream. 
> 
> Once RS receives the request packet from an RR, RS knows its 
> address. Ports
> are defined in the SDP. If you are asking about "NAT" issues, 
> we have a
> section for it, and we plan to complete it as we move 
> forward. It is not as
> critical as the other parts for now.
>  
> > If you mandate the use of RTP/RTCP mux it should say so 
> > otherwise the RAMS-R should have an optional parameter that 
> > supplies this information and a flag for using RTP/RTCP mux.
> 
> IMO, we cannot mandate muxing as it is not required to make 
> RAMS work. If
> muxing is supported, the SDP should reflect it.
> 
> BR,
> -acbegen
>  
> > Thanks
> > 
> > Roni Even
> > 
> >  
> > 
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > Behalf Of Bill Ver Steeg (versteb)
> > Sent: Thursday, April 16, 2009 7:39 PM
> > To: avt@ietf.org
> > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > RTP Sessions (RAMS)" available
> > 
> >  
> > 
> > There is a new draft of the "Rapid Acquisition of Multicast 
> > RTP Sessions (RAMS)" draft available at 
> > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > 
> >  
> > 
> > We have incorporated the changes from the technical breakout 
> > session in San Francisco. The major changes in this version 
> > of the draft include 
> > 
> > 1- Changing the document title to avoid confusion with other 
> > ongoing "synchronization" drafts
> > 
> > 2- changing the message names to reflect the title change
> > 
> > 3- clarification of the RTCP message semantics, including 
> > changes to the "Request" and "Inform" messages
> > 
> > 4- additional description/motivation for the various message 
> > flows has been added
> > 
> > 5- RTP/RTCP muxing has been added
> > 
> >  
> > 
> >  
> > 
> > We hope to make this a Working Group item, and will change 
> > the name of the draft to avoid conflicts with other 
> > "synchronization" drafts at that time.
> > 
> >  
> > 
> > Bill VerSteeg
> > 
> >  
> > 
> >  
> > 
> > 
> 
> 


From ron.even.tlv@gmail.com  Fri Apr 24 12:32:23 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <fecframe@ietf.org>
References: <04CAD96D4C5A3D48B1919248A8FE0D54091BAA88@xmb-sjc-215.amer.cisco.com> <49f1cd82.15185e0a.0fc3.ffffb9e7@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80EF@xmb-sjc-215.amer.cisco.com>
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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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Ali,
I am a bit confused about your answer, I assume that FEC framework you refer
to draft-ietf-fecframe-framework-03, at least I cold not see any reference
in the draft to a document. 
In section 4 of the above draft it says 

"
In the case that RTP is used for the repair flows, the source and repair
   data may be multiplexed using RTP onto a single UDP flow and must
   consequently be demultiplexed at the receiver."


Roni


> 6. In section 7 you discuss an exchange using grouping. Is it 
> also allowed
> to use FEC withoutout grouping using the video or audio media 
> types like
> section 14 in RFC 5109

Per FEC framework, we require the FEC to be in a separate flow, which
requires the source and FEC streams to be on different RTP sessions. Thus,
we require grouping (RFC 4756 or 4756bis)





In the case that RTP is used for the repair flows, the source and repair
   data may be multiplexed using RTP onto a single UDP flow and must
   consequently be demultiplexed at the receiver.


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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This is a part of an example SDP sent to RS and RR in a SAP =
announcement, for example. So, the SDP describes what both parties =
should do (RR cannot say that he wants to receive this multicast on its =
favorite port). The individual SDPs sent to RR or RS may include other =
portions of descriptions that will contain specific information.=20

-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Friday, April 24, 2009 12:23 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Ali,
> I think you get it wrong, this SDP is from the RS and not the=20
> RR so the RS
> cannot specify to which address it will send it can only=20
> specify to which
> address it can receive RTP stream. In this SDP the relevant=20
> information is
> that the request for retransmission will be sent by the RR to=20
> port 41003
> Roni
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Friday, April 24, 2009 10:01 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> Oh, I see. The burst goes to the port that we would normally send the
> retransmissions. For example, consider the SDP from the draft:
>=20
>         m=3Dvideo 41000 RTP/AVPF 98
>         i=3DPrimary Multicast Stream #2
>         c=3DIN IP4 224.1.1.2/255
>         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
>         a=3Drtpmap:98 MP2T/90000
>         a=3Drtcp:41001 IN IP4 192.0.2.1
>         a=3Drtcp-fb:98 nack
>         a=3Drtcp-fb:98 nack ssli
>         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
>         a=3Dmid:3
>         m=3Dvideo 41002 RTP/AVPF 99
>         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> Acq. Support)
>         c=3DIN IP4 192.0.2.1
>         a=3Drecvonly
>         a=3Drtpmap:99 rtx/90000
>         a=3Drtcp:41003
>         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
>         a=3Dmid:4
>=20
> In this case, the burst will go to port 41002 on the RR. The RAMS-I
> message(s), which is an RTCP feedback message, will go to port 41003.
>=20
> HTH,
> -acbegen
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > Sent: Friday, April 24, 2009 11:44 AM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Ali,
> > I will try to explain in simple way
> >=20
> > When the RS receives the RAMS-R it need to start sending an=20
> > RTP stream to
> > the RR.=20
> > In order to send a unicast packet, the RS needs to know a=20
> > transport address
> > on the RR to where the RTP stream will be sent. The current=20
> > draft does not
> > say how the RS knows this address. There is no SDP from RR to=20
> > RS like you
> > mention in your response.
> > This is why I say that the current draft does not specify a=20
> > solution that
> > can be implemented as a working solution
> > Roni
> >=20
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Friday, April 24, 2009 7:38 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > Hi Roni,=20
> >=20
> > > -----Original Message-----
> > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > Behalf Of Roni Even
> > > Sent: Friday, April 24, 2009 4:24 AM
> > > To: avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions (RAMS)" available
> > >=20
> > > Hi,
> > >=20
> > > I think that the current draft does not give a description of=20
> > > a system that works since there is no text explaining how the=20
> >=20
> > What do you mean by "a system that works"?=20
> >=20
> > > RS knows the unicast transport address on the RR to where to=20
> > > send the stream.=20
> >=20
> > Once RS receives the request packet from an RR, RS knows its=20
> > address. Ports
> > are defined in the SDP. If you are asking about "NAT" issues,=20
> > we have a
> > section for it, and we plan to complete it as we move=20
> > forward. It is not as
> > critical as the other parts for now.
> > =20
> > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > otherwise the RAMS-R should have an optional parameter that=20
> > > supplies this information and a flag for using RTP/RTCP mux.
> >=20
> > IMO, we cannot mandate muxing as it is not required to make=20
> > RAMS work. If
> > muxing is supported, the SDP should reflect it.
> >=20
> > BR,
> > -acbegen
> > =20
> > > Thanks
> > >=20
> > > Roni Even
> > >=20
> > > =20
> > >=20
> > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > Behalf Of Bill Ver Steeg (versteb)
> > > Sent: Thursday, April 16, 2009 7:39 PM
> > > To: avt@ietf.org
> > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > > RTP Sessions (RAMS)" available
> > >=20
> > > =20
> > >=20
> > > There is a new draft of the "Rapid Acquisition of Multicast=20
> > > RTP Sessions (RAMS)" draft available at=20
> > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > >=20
> > > =20
> > >=20
> > > We have incorporated the changes from the technical breakout=20
> > > session in San Francisco. The major changes in this version=20
> > > of the draft include=20
> > >=20
> > > 1- Changing the document title to avoid confusion with other=20
> > > ongoing "synchronization" drafts
> > >=20
> > > 2- changing the message names to reflect the title change
> > >=20
> > > 3- clarification of the RTCP message semantics, including=20
> > > changes to the "Request" and "Inform" messages
> > >=20
> > > 4- additional description/motivation for the various message=20
> > > flows has been added
> > >=20
> > > 5- RTP/RTCP muxing has been added
> > >=20
> > > =20
> > >=20
> > > =20
> > >=20
> > > We hope to make this a Working Group item, and will change=20
> > > the name of the draft to avoid conflicts with other=20
> > > "synchronization" drafts at that time.
> > >=20
> > > =20
> > >=20
> > > Bill VerSteeg
> > >=20
> > > =20
> > >=20
> > > =20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Roni-
=20
These are reasonable approaches, which I need to consider in detail. I
have (mostly) been thinking about NAT-friendly solutions, which leads me
back to muxing. I need to consider the ramifications of NAT on the
methods that signal IP addresses and port numbers as part of the
run-time for acceleration. The intent is to make acquisition faster, and
additional complexity at session instantiation should be avoided (IMHO).
=20
More to come, once I get my head around the various options.
=20
bvs
=20

________________________________

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Roni Even
Sent: Friday, April 24, 2009 3:19 PM
To: Bill Ver Steeg (versteb); avt@ietf.org
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast
RTPSessions (RAMS)" available



Bill,

=20

Since I have two hats here then as a WG chair, my intention is to ask
the list to start this topic as a WG item

=20

Taking my WG chair off, my expectation is that a WG document will
describe a solution that can be implemented while the current text does
not specify how to make it work. I do not care if you write that
RTP/RTCP mux MUST be used by the RS and supported by the RR and update
the SDP in the example to specify that this is case. =20

=20

My personal preference is to have the RS indicate in the SDP that it
wants to use multiplexing and to add a flag to RAMS-R that will indicate
if RTP/RTCP mux is supported by RR and can be used. Only if the RR
cannot support the multiplexing it should indicate it using the flag and
add to the RAMS-R the transport address to be used (Note that it can be
v4 or v6).

SDP cannot be used since there is no SDP from RR to RS.

=20

Roni Even

=20

From: Bill Ver Steeg (versteb) [mailto:versteb@cisco.com]=20
Sent: Friday, April 24, 2009 9:50 PM
To: Roni Even; avt@ietf.org
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP
Sessions (RAMS)" available

=20

Roni-

=20

Yes, the document needs updating in this respect. We have two broad
options which need to be described in detail.

=20

The method that I have been assuming does use muxing. The RS sends the
unicast burst back to the UDP port number on the RR that sent the
RAMS-R. This allows transparent operation through NAT. There is a single
UDP port, and the transaction is initiated by the "client", so it is NAT
friendly.

=20

There are quite a few ways we could handle the case in which muxing is
not used. We could add a parameter to the RAMS-R, use another RTCP
packet in the same compound packet, define the port numbers as a
relationship to the opened port (not attractive), or define something in
the SDP (also not attractive). I personally like the "RAMS-R optional
field" approach, but I think this should be a topic that we consider in
detail once we have a WG item.

=20

=20

=20

=20

bvs

=20

=20

________________________________

From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
Sent: Friday, April 24, 2009 7:24 AM
To: avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP
Sessions (RAMS)" available

Hi,

I think that the current draft does not give a description of a system
that works since there is no text explaining how the RS knows the
unicast transport address on the RR to where to send the stream.=20

If you mandate the use of RTP/RTCP mux it should say so otherwise the
RAMS-R should have an optional parameter that supplies this information
and a flag for using RTP/RTCP mux.

Thanks

Roni Even

=20

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of
Bill Ver Steeg (versteb)
Sent: Thursday, April 16, 2009 7:39 PM
To: avt@ietf.org
Subject: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" available

=20

There is a new draft of the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft available at
http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniza
tion-for-rtp-03.txt
<http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-synchroniz
ation-for-rtp-03.txt>=20

=20

We have incorporated the changes from the technical breakout session in
San Francisco. The major changes in this version of the draft include=20

1- Changing the document title to avoid confusion with other ongoing
"synchronization" drafts

2- changing the message names to reflect the title change

3- clarification of the RTCP message semantics, including changes to the
"Request" and "Inform" messages

4- additional description/motivation for the various message flows has
been added

5- RTP/RTCP muxing has been added

=20

=20

We hope to make this a Working Group item, and will change the name of
the draft to avoid conflicts with other "synchronization" drafts at that
time.

=20

Bill VerSteeg

=20

=20


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<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>These are reasonable approaches, which I need =
to consider=20
in detail. I have (mostly) been thinking about NAT-friendly solutions, =
which=20
leads me back to muxing. I need to consider the ramifications of NAT on =
the=20
methods that signal IP addresses and port numbers as part of the =
run-time for=20
acceleration. The intent is to make acquisition faster, and additional=20
complexity at session instantiation should be avoided=20
(IMHO).</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>More to come, once I get my head around the =
various=20
options.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>bvs</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D785083319-24042009><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> avt-bounces@ietf.org=20
[mailto:avt-bounces@ietf.org] <B>On Behalf Of </B>Roni =
Even<BR><B>Sent:</B>=20
Friday, April 24, 2009 3:19 PM<BR><B>To:</B> Bill Ver Steeg (versteb);=20
avt@ietf.org<BR><B>Subject:</B> Re: [AVT] New draft of "Rapid =
Acquisition of=20
Multicast RTPSessions (RAMS)" available<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV class=3DSection1>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Bill,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Since=20
I have two hats here then as a WG chair, my intention is to ask the list =
to=20
start this topic as a WG item<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Taking=20
my WG chair off, my expectation is that a WG document will describe a =
solution=20
that can be implemented while the current text does not specify how to =
make it=20
work. I do not care if you write that RTP/RTCP mux MUST be used by the =
RS and=20
supported by the RR and update the SDP in the example to specify that =
this is=20
case. &nbsp;<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">My=20
personal preference is to have the RS indicate in the SDP that it wants =
to use=20
multiplexing and to add a flag to RAMS-R that will indicate if RTP/RTCP =
mux is=20
supported by RR and can be used. Only if the RR cannot support the =
multiplexing=20
it should indicate it using the flag and add to the RAMS-R the transport =
address=20
to be used (Note that it can be v4 or v6).<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">SDP=20
cannot be used since there is no SDP from RR to =
RS.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Roni=20
Even<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
#b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: =
medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Bill Ver =
Steeg=20
(versteb) [mailto:versteb@cisco.com] <BR><B>Sent:</B> Friday, April 24, =
2009=20
9:50 PM<BR><B>To:</B> Roni Even; avt@ietf.org<BR><B>Subject:</B> RE: =
[AVT] New=20
draft of "Rapid Acquisition of Multicast RTP Sessions (RAMS)"=20
available<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
'Arial','sans-serif'">Roni-</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
'Arial','sans-serif'">Yes, the=20
document needs updating in this respect. We have two broad options which =
need to=20
be described in detail.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
'Arial','sans-serif'">The=20
method that I have been assuming does use muxing. The&nbsp;RS sends the =
unicast=20
burst back to the UDP port number on the RR that sent the RAMS-R. This =
allows=20
transparent operation through NAT. There is a single UDP port, and the=20
transaction is initiated by the "client", so it is NAT=20
friendly.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
'Arial','sans-serif'">There=20
are quite a few ways we could handle the case in which muxing is not =
used. We=20
could add a parameter to the RAMS-R, use another RTCP packet in the same =

compound packet, define the port numbers as a relationship to the opened =
port=20
(not attractive), or define something in the SDP (also not attractive). =
I=20
personally like the "RAMS-R optional field" approach, but I think this =
should be=20
a topic that we consider in detail once we have a WG =
item.</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; COLOR: blue; FONT-FAMILY: =
'Arial','sans-serif'">bvs</SPAN><o:p></o:p></P>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV class=3DMsoNormal style=3D"TEXT-ALIGN: center" align=3Dcenter>
<HR align=3Dcenter width=3D"100%" SIZE=3D2>
</DIV>
<P class=3DMsoNormal style=3D"MARGIN-BOTTOM: 12pt"><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'"> Roni Even =

[mailto:ron.even.tlv@gmail.com] <BR><B>Sent:</B> Friday, April 24, 2009 =
7:24=20
AM<BR><B>To:</B> avt@ietf.org<BR><B>Cc:</B> Bill Ver Steeg=20
(versteb)<BR><B>Subject:</B> RE: [AVT] New draft of "Rapid Acquisition =
of=20
Multicast RTP Sessions (RAMS)" available</SPAN><o:p></o:p></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Hi,<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">I=20
think that the current draft does not give a description of a system =
that works=20
since there is no text explaining how the RS knows the unicast transport =
address=20
on the RR to where to send the stream. <o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">If=20
you mandate the use of RTP/RTCP mux it should say so otherwise the =
RAMS-R should=20
have an optional parameter that supplies this information and a flag for =
using=20
RTP/RTCP mux.<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Thanks<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'">Roni=20
Even<o:p></o:p></SPAN></P>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 11pt; COLOR: #1f497d; FONT-FAMILY: =
'Calibri','sans-serif'"><o:p>&nbsp;</o:p></SPAN></P>
<DIV>
<DIV=20
style=3D"BORDER-RIGHT: medium none; PADDING-RIGHT: 0in; BORDER-TOP: =
#b5c4df 1pt solid; PADDING-LEFT: 0in; PADDING-BOTTOM: 0in; BORDER-LEFT: =
medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<P class=3DMsoNormal><B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: =
'Tahoma','sans-serif'">From:</SPAN></B><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Tahoma','sans-serif'">=20
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <B>On Behalf Of =
</B>Bill Ver=20
Steeg (versteb)<BR><B>Sent:</B> Thursday, April 16, 2009 7:39 =
PM<BR><B>To:</B>=20
avt@ietf.org<BR><B>Subject:</B> [AVT] New draft of "Rapid Acquisition of =

Multicast RTP Sessions (RAMS)" =
available<o:p></o:p></SPAN></P></DIV></DIV>
<P class=3DMsoNormal><o:p>&nbsp;</o:p></P>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">There is a =
new draft=20
of the "Rapid Acquisition of Multicast RTP Sessions (RAMS)" draft =
available at=20
<A=20
href=3D"http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-sync=
hronization-for-rtp-03.txt"><SPAN=20
style=3D"FONT-SIZE: 12pt; FONT-FAMILY: 'Times New =
Roman','serif'">http://www.ietf.org/internet-drafts/draft-versteeg-avt-ra=
pid-synchronization-for-rtp-03.txt</SPAN></A></SPAN><o:p></o:p></P></DIV>=

<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">We=20
have&nbsp;incorporated the changes from the technical breakout session =
in San=20
Francisco. The major changes in this version of the draft include=20
</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">1- Changing =
the=20
document title to avoid confusion with other ongoing "synchronization"=20
drafts</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">2- changing =
the=20
message names to reflect the title change</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">3- =
clarification of=20
the RTCP message semantics, including&nbsp;changes to the&nbsp;"Request" =
and=20
"Inform" messages</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">4- =
additional=20
description/motivation for the various message flows has been=20
added</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">5- RTP/RTCP =
muxing=20
has been added</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">We hope to =
make this=20
a Working Group item, and will change the name of the draft to avoid =
conflicts=20
with other "synchronization" drafts at that =
time.</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal><SPAN=20
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: 'Arial','sans-serif'">Bill=20
VerSteeg</SPAN><o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV>
<DIV>
<P class=3DMsoNormal>&nbsp;<o:p></o:p></P></DIV></DIV></BODY></HTML>

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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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> > 6. In section 7 you discuss an exchange using grouping. Is it=20
> > also allowed
> > to use FEC withoutout grouping using the video or audio media=20
> > types like
> > section 14 in RFC 5109
>=20
> Per FEC framework, we require the FEC to be in a separate flow, which
> requires the source and FEC streams to be on different RTP=20
> sessions. Thus,
> we require grouping (RFC 4756 or 4756bis)
>=20
> RE- I will have to look at that draft, this is different from=20
> RFC 5109 and
> RFC 2733

Actually, we might be supporting SSRC multiplexing in the fec framework =
(see my email sent today to fecframe/mmusic about this topic), since =
there are some use cases for it. It does bring up other issues, though. =
I would be glad to hear your comments.

There are also some issues with RFC 5109 regarding SSRC multiplexing as =
you mentioned in one of your earlier emails that may need to be =
addressed.

BR,
-acbegen

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Thread-Topic: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <fecframe@ietf.org>
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Subject: Re: [AVT] FW: New Version Notification for	draft-ietf-fecframe-interleaved-fec-scheme-03
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Yes, that is the fecframework draft, and I raised exactly the same point =
this morning in the fecframe/mmusic list when I noticed that text in the =
draft. Please use that thread for that discussion.=20

-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Friday, April 24, 2009 12:32 PM
> To: Ali C. Begen (abegen); fecframe@ietf.org
> Cc: avt@ietf.org
> Subject: RE: [AVT] FW: New Version Notification for=20
> draft-ietf-fecframe-interleaved-fec-scheme-03
>=20
> Ali,
> I am a bit confused about your answer, I assume that FEC=20
> framework you refer
> to draft-ietf-fecframe-framework-03, at least I cold not see=20
> any reference
> in the draft to a document.=20
> In section 4 of the above draft it says=20
>=20
> "
> In the case that RTP is used for the repair flows, the source=20
> and repair
>    data may be multiplexed using RTP onto a single UDP flow and must
>    consequently be demultiplexed at the receiver."
>=20
>=20
> Roni
>=20
>=20
> > 6. In section 7 you discuss an exchange using grouping. Is it=20
> > also allowed
> > to use FEC withoutout grouping using the video or audio media=20
> > types like
> > section 14 in RFC 5109
>=20
> Per FEC framework, we require the FEC to be in a separate flow, which
> requires the source and FEC streams to be on different RTP=20
> sessions. Thus,
> we require grouping (RFC 4756 or 4756bis)
>=20
>=20
>=20
>=20
>=20
> In the case that RTP is used for the repair flows, the source=20
> and repair
>    data may be multiplexed using RTP onto a single UDP flow and must
>    consequently be demultiplexed at the receiver.
>=20
>=20

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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 Format for Adaptive TRansform Acoustic Coding (ATRAC) Family
	Author(s)	: J. Matsumoto, M. Hatanaka
	Filename	: draft-ietf-avt-rtp-atrac-family-22.txt
	Pages		: 35
	Date		: 2009-4-24
	
This document describes an RTP payload format for efficient and
   flexible transporting of audio data encoded with the Adaptive
   TRansform Audio Coding (ATRAC) family of codecs. Recent enhancements
   to the ATRAC family of codecs support high quality audio coding with
   multiple channels.  The RTP payload format as presented in this
   document also includes support for data fragmentation, elementary
   redundancy measures, and a variation on scalable streaming.

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From petithug@acm.org  Fri Apr 24 16:30:06 2009
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Subject: [AVT] SDP attribute in rtcpssm-18
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Section 10.1 of draft-ietf-avt-rtcpssm-18 states that '[a] new
session level attribute MUST be used [...]: "rtcp-unicast"'  but the
 IANA considerations section registers a media level only att-field
named "unicast-rtcp".  So which one should be used, and is it a
media or a session level attribute?

Thanks.

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

From tendyntu@huawei.com  Fri Apr 24 18:32:51 2009
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Date: Sat, 25 Apr 2009 09:23:41 +0800
From: Zou ZiXuan <tendyntu@huawei.com>
In-reply-to: <04CAD96D4C5A3D48B1919248A8FE0D54092C8066@xmb-sjc-215.amer.cisco.com>
To: "'Ali C. Begen (abegen)'" <abegen@cisco.com>, "'Bill Ver Steeg (versteb)'" <versteb@cisco.com>, avt@ietf.org
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions	(RAMS)" available
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Hi, Ali,
	Thanks for your explanation. 

> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]
> Sent: Friday, April 24, 2009 11:13 PM
> To: Zou ZiXuan; Bill Ver Steeg (versteb); avt@ietf.org
> Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTP
Sessions
> (RAMS)" available
> 
> Hi Zou,
> 
> This is a very good question. Let me take a stab here.
> 
> Per RFC 3550, each SSRC space has its own clock rate and seq
numbering.
> We include the SSRC in the RAMS request messages.
> 
> There are different flavors of your scneario. Lets assume that we have
two
> RTP streams (audio and video).
> 
> 
> Case 1: Different RTP sessions are transmitted over different mcast
groups.
> 
> Here, we run a different independent RAMS session for each RTP stream
> that we want to rapidly acquire (they can run in parallel). Here one
needs to
> pay attention to how to use the shared bandwidth among different
bursts,
> though.
> 
> Case 2: Different RTP sessions are transmitted over the same mcast
group
> (with different dest ports).
> 
> Again, we run a different independent RAMS for each RTP stream
(different
> request for each, etc). But, one should keep in mind that the join
times for the
> all RAMS sessions need to be common as there is only one multicast.
> 
> Case 3: One RTP session (same mcast group, same dest port) carrying
both
> audio and video (this is SSRC multiplexing).
> 
> Similar to case 2, join time needs to be coordinated among different
RAMS
> sessions running for each RTP stream. Since we signal SSRC in the
request
> message and we have independent RAMS sessions, using different clock
> rates is not an issue.
> 
> Case 4: One RTP session, one SSRC with payload type multiplexing. This
is
> not recommended for many reasons. On top of that, different payloads
with
> different clock rates cannot be multiplexed in this fashion. So, I
don't think this
> will apply to our problem.
> 
> HTH,
> -acbegen
> 
> 
> > -----Original Message-----
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
> > Behalf Of Zou ZiXuan
> > Sent: Friday, April 24, 2009 3:03 AM
> > To: Bill Ver Steeg (versteb); avt@ietf.org
> > Subject: Re: [AVT] New draft of "Rapid Acquisition of
> > Multicast RTP Sessions (RAMS)" available
> >
> > Hi,
> >
> >          The burst duration (8 octets) in RAMS-I message is
> > in a single RTP ticks. And I note that from draft-01 on, the
> > solution is payload-independent (is my understanding
> > correct?). So, for non-MEPG2-TS payload, for example,
> > H.264-NAL where audio and video content are encapsulated
> > separately in two RTP streams with different ticks, does the
> > duration indicated by this single ticks still work? Which
> > stream does this tick belong to?
> >
> >
> >
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On
> > Behalf Of Bill Ver Steeg (versteb)
> > Sent: Friday, April 17, 2009 12:39 AM
> > To: avt@ietf.org
> > Subject: [AVT] New draft of "Rapid Acquisition of Multicast
> > RTP Sessions (RAMS)" available
> >
> >
> >
> > There is a new draft of the "Rapid Acquisition of Multicast
> > RTP Sessions (RAMS)" draft available at
> > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>
> >
> >
> >
> > We have incorporated the changes from the technical breakout
> > session in San Francisco. The major changes in this version
> > of the draft include
> >
> > 1- Changing the document title to avoid confusion with other
> > ongoing "synchronization" drafts
> >
> > 2- changing the message names to reflect the title change
> >
> > 3- clarification of the RTCP message semantics, including
> > changes to the "Request" and "Inform" messages
> >
> > 4- additional description/motivation for the various message
> > flows has been added
> >
> > 5- RTP/RTCP muxing has been added
> >
> >
> >
> >
> >
> > We hope to make this a Working Group item, and will change
> > the name of the draft to avoid conflicts with other
> > "synchronization" drafts at that time.
> >
> >
> >
> > Bill VerSteeg
> >
> >
> >
> >
> >
> >


From ron.even.tlv@gmail.com  Sat Apr 25 04:09:09 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com>
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Date: Sat, 25 Apr 2009 14:08:27 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
The example SDP is syntactically correct but it does not say that the
rtp/rtcp mux will be used and it does not give the information to where the
unicast stream will be sent when the RR sends a RAMS-R. 


I am not sure why the unicast stream m-line has a port number with an
attribute of recvonly. What is the use case for that. The only information
that the RR will need is the RTCP port on the RS to where to send the RAMS-R
message.
Roni
-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Friday, April 24, 2009 10:37 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

This is a part of an example SDP sent to RS and RR in a SAP announcement,
for example. So, the SDP describes what both parties should do (RR cannot
say that he wants to receive this multicast on its favorite port). The
individual SDPs sent to RR or RS may include other portions of descriptions
that will contain specific information. 

-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Friday, April 24, 2009 12:23 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> I think you get it wrong, this SDP is from the RS and not the 
> RR so the RS
> cannot specify to which address it will send it can only 
> specify to which
> address it can receive RTP stream. In this SDP the relevant 
> information is
> that the request for retransmission will be sent by the RR to 
> port 41003
> Roni
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Friday, April 24, 2009 10:01 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> Oh, I see. The burst goes to the port that we would normally send the
> retransmissions. For example, consider the SDP from the draft:
> 
>         m=video 41000 RTP/AVPF 98
>         i=Primary Multicast Stream #2
>         c=IN IP4 224.1.1.2/255
>         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
>         a=rtpmap:98 MP2T/90000
>         a=rtcp:41001 IN IP4 192.0.2.1
>         a=rtcp-fb:98 nack
>         a=rtcp-fb:98 nack ssli
>         a=ssrc:123321 cname:iptv-ch32@rams.example.com
>         a=mid:3
>         m=video 41002 RTP/AVPF 99
>         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> Acq. Support)
>         c=IN IP4 192.0.2.1
>         a=recvonly
>         a=rtpmap:99 rtx/90000
>         a=rtcp:41003
>         a=fmtp:99 apt=98; rtx-time=5000
>         a=mid:4
> 
> In this case, the burst will go to port 41002 on the RR. The RAMS-I
> message(s), which is an RTCP feedback message, will go to port 41003.
> 
> HTH,
> -acbegen
> 
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > Sent: Friday, April 24, 2009 11:44 AM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions (RAMS)" available
> > 
> > Ali,
> > I will try to explain in simple way
> > 
> > When the RS receives the RAMS-R it need to start sending an 
> > RTP stream to
> > the RR. 
> > In order to send a unicast packet, the RS needs to know a 
> > transport address
> > on the RR to where the RTP stream will be sent. The current 
> > draft does not
> > say how the RS knows this address. There is no SDP from RR to 
> > RS like you
> > mention in your response.
> > This is why I say that the current draft does not specify a 
> > solution that
> > can be implemented as a working solution
> > Roni
> > 
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Friday, April 24, 2009 7:38 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > Hi Roni, 
> > 
> > > -----Original Message-----
> > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > Behalf Of Roni Even
> > > Sent: Friday, April 24, 2009 4:24 AM
> > > To: avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Hi,
> > > 
> > > I think that the current draft does not give a description of 
> > > a system that works since there is no text explaining how the 
> > 
> > What do you mean by "a system that works"? 
> > 
> > > RS knows the unicast transport address on the RR to where to 
> > > send the stream. 
> > 
> > Once RS receives the request packet from an RR, RS knows its 
> > address. Ports
> > are defined in the SDP. If you are asking about "NAT" issues, 
> > we have a
> > section for it, and we plan to complete it as we move 
> > forward. It is not as
> > critical as the other parts for now.
> >  
> > > If you mandate the use of RTP/RTCP mux it should say so 
> > > otherwise the RAMS-R should have an optional parameter that 
> > > supplies this information and a flag for using RTP/RTCP mux.
> > 
> > IMO, we cannot mandate muxing as it is not required to make 
> > RAMS work. If
> > muxing is supported, the SDP should reflect it.
> > 
> > BR,
> > -acbegen
> >  
> > > Thanks
> > > 
> > > Roni Even
> > > 
> > >  
> > > 
> > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > Behalf Of Bill Ver Steeg (versteb)
> > > Sent: Thursday, April 16, 2009 7:39 PM
> > > To: avt@ietf.org
> > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > RTP Sessions (RAMS)" available
> > > 
> > >  
> > > 
> > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > RTP Sessions (RAMS)" draft available at 
> > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > 
> > >  
> > > 
> > > We have incorporated the changes from the technical breakout 
> > > session in San Francisco. The major changes in this version 
> > > of the draft include 
> > > 
> > > 1- Changing the document title to avoid confusion with other 
> > > ongoing "synchronization" drafts
> > > 
> > > 2- changing the message names to reflect the title change
> > > 
> > > 3- clarification of the RTCP message semantics, including 
> > > changes to the "Request" and "Inform" messages
> > > 
> > > 4- additional description/motivation for the various message 
> > > flows has been added
> > > 
> > > 5- RTP/RTCP muxing has been added
> > > 
> > >  
> > > 
> > >  
> > > 
> > > We hope to make this a Working Group item, and will change 
> > > the name of the draft to avoid conflicts with other 
> > > "synchronization" drafts at that time.
> > > 
> > >  
> > > 
> > > Bill VerSteeg
> > > 
> > >  
> > > 
> > >  
> > > 
> > > 
> > 
> > 
> 
> 


From ingemar.s.johansson@ericsson.com  Sat Apr 25 05:24:32 2009
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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From: "Ingemar Johansson S" <ingemar.s.johansson@ericsson.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, <avt@ietf.org>
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Cc: Ingemar Johansson S <ingemar.s.johansson@ericsson.com>, "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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Hi

Sorry for the delay, see my comments inline

Regards
Ingemar=20

> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: den 21 april 2009 20:28
> To: Ingemar Johansson S; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTP Sessions(RAMS)" available
>=20
> Hi Ingemar,=20
>=20
> > Hi
> >=20
> > I have read through the updated draft and have a few comments and=20
> > requests for your consideration.
>=20
> Thanks for the review.
> =20
> > In general (as I hinted before) I would like this draft to not just=20
> > discuss the unicast version of RAMS, instead I would like=20
> it to also=20
> > incorporate the multicast version into the framework.
>=20
> While I see the common goal for these two methods, I believe=20
> the one you proposed is an additional enhancement on the=20
> unicast approach. Regarding whether they need to be combined=20
> or not, I have some comments below.
>=20
> > I have looked into this and I would say that support for=20
> the multicast=20
> > alternative as described in=20
> > http://tools.ietf.org/id/draft-johansson-avt-mcast-based-rams-00.txt
> > is fairly straightforward to add to the existing framework, I will=20
> > describe in brief below the key additions that I see today.
>=20
> I would not say it would be "fairly straighforward." To make=20
> the multicast bursting effective, many parameters need to be=20
> adjusted properly and optimized. Ow, it can easily degrade=20
> the performance even at the expense of additional complexity.
>=20
> A concern I have which is not addressed in your document is=20
> that to make the proposal make sense, Td needs to be small=20
> enough. But, to ensure that all receivers join the burst=20
> mcast session before the burst starts, Td cannot be very=20
> small and RS should not admit last-minute RAMS requests for=20
> that multicast burst (they need to wait for the next batch).=20
> This is because if some receivers miss the first few packets=20
> of the burst, their acquisition will fail. Also, how many=20
> clients can the retransmission server aggregate in a single=20
> burst batch in a practical scenario? The eligible requests=20
> should be received within +/- a few hundereds of=20
> milliseconds. How practical is this, I wonder.
The neat thing is that during low load conditions Td can be made very =
low such that in practice only one user is served by each request. This =
can be implemented with either unicast or multicast bursts that only =
serve one STB. As he load increases Td increases and more users that =
send a requests will be served by one multicast channel. As Td is =
limited some requests will be too late but they can too be collected and =
served by additional multicast feeds. Then depending on the distribuion =
the tail may be handled with unicast bursts. This is more or less an =
optimization issue on the servers.
The idea is not that the unicast and multicast alternatives should =
compete with one another, rather they should complement each other, due =
to the highly dynamic nature of channel switching behavior in a =
population both alternatives are useful. I do believe that the multicast =
solution can help to reduce deplyment cost.=20
I have tried to look for statistics on channel switching behavior and of =
what I have seen sofar the "peak-to-average" ratio of channel swithing =
density can be more than 5. This is seen in a master thesis with the =
title "TPTV traffic measuring & modeling" by Geng Yu (have not found it =
published anywhere yet).=20

> The selection of a common burst bitrate is also critical. If=20
> everybody supports a different burst bitrate and there is a=20
> large variation, multicast bursting will perform poorly for=20
> the well-connected clients. One solution could be to include=20
> the receivers with an available bw that is smaller than R1=20
> and larger than R2 in the multicast pool. The very fast and=20
> very slow receivers should be still served via unicast.
Sounds reasonable.

>=20
> My personal view is that the multicast bursting proposal will=20
> be a nice solution when the issues above (and possible=20
> others) are addressed, however, I think it should be a=20
> separate draft that normatively references the unicast draft.=20
> This way, we can keep the unicast draft more focused.
Not sure that I agree here, it all depends on how the signaling is =
standardized. For instance an STB likely needs to declare its =
unicast/multicast preference or capability in the RAMS-R as the SDP's =
are likely only declarative.=20

>=20
> > The title of the draft would then become "Rapid Acquisition of=20
> > Multicast RTP Sessions". I can offer to add the necessary=20
> parts in the=20
> > document that describes the multicast based version of RAMS.
> >=20
> > Packet structure:
> > In the draft, three FMT's are defined for the RAMS R,I and=20
> T messages.=20
> > I wonder if it would be better to use one just one FMT for this=20
> > "family" of messages and use a subtype to define R, I, T=20
> (or any other=20
> > message). This was also brought up during the break-out=20
> session in SF.
>=20
> This was indeed a good recommendation and we already took=20
> that into account, but we did not want to modify the packet=20
> formats too much to avoid any confusion in this version. If=20
> you look at the open items section, it is already there.
Hmm... Sorry, did not see it.

>=20
> > Also I would consider a version field as well as there is a chance=20
> > that things need to be changed in the future and TLV extensions may=20
> > not solve all backwards compatibility issues in a satisfactory way=20
> > (changing version number should not be too easy though, TLV=20
> elements=20
> > should be the first choice).
>=20
> In a later version, the meanings of the mandatory fields=20
> would not change and TLVs are optional anyway. So, why do you=20
> think new TLVs would not give you what you want? Not that I=20
> am against using a version field, but I would rather see a=20
> need for it.
I would appreciate the comments from others here, my argument sofar is =
that the cost is quite small but I agree that there are ways to solve =
future issues like using a completely new FMT. Problem here is maybe =
that it is difficult to foresee the future need.

>=20
> > With this in mind the first 32 bits may look like:
> >  0                   1                   2                   3
> >  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 =20
> > +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> >  | ver |subtype| flags   .... |reserved                      =20
> >    |         =20
> >  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
> > In this example 3 bits are used for the version number and=20
> 5 bits are=20
> > used for the subtype but there may be a better distribution=20
> here. The=20
> > number of bits for the flags and reserved field depends on=20
> the subtype=20
> > The subtypes would be encoded as RAMS-R =3D 0 RAMS-I =3D 1 RAMS-T =
=3D 2
> >=20
> >=20
> > Figure 2:=20
> > A flow diagram that describes the multicast version to be added.
> >=20
> > Figure 3:=20
> > A description that describes the multicast version to be added.
> >=20
> > Section 6.4: Failure cases
> > An alternative to RAMS-T in the multicast version of RAMS=20
> is the use=20
> > of IGMP-Leave (details about this are needed)
> >=20
> > Section 7.1 RAMS Request
> > Here the 32 bit header is added (with subtype=3D0) and possibly the=20
> > preference and or support of the multicast and the unicast=20
> version of=20
> > RAMS can be defined in the flags field.
> > The latter is likely needed as the SDP's are likely only=20
> declarative.
> >
> > Section 7.2 RAMS Information
> > Here the 32 bit header is added (with subtype=3D1) and the MSN,S and =

> > Response field is put in the flags field.
> > An additional TLV element is added that includes a=20
> multicast address=20
> > and possibly other information such as expected delay until the=20
> > multicast RAMS starts. This field is mandatory if the multicast=20
> > address for RAMS is described in this way.
> >=20
> > Section 7.2 RAMS Termination
> > Here the 32 bit header is added (with subtype=3D2) Note that=20
> RAMS-T may=20
> > not be needed in the multicast version of RAMS (T.B.D)
>=20
> These could be all addressed if/when we decide to combine the=20
> mcast bursting support.
OK


>=20
> Thanks again for the detail review.
>=20
> BR,
> -acbegen
> =20
> > Comments to my comments are welcome
> > Regards
> > Ingemar
> > *******************************************
> > Ingemar Johansson
> > Senior Research Engineer, IETF "nethead"=20
> > EAB/TVP - Multimedia Technologies
> > Ericsson Research Ericsson AB
> > Box 920 S-971 28 Lule=E5, Sweden
> > Tel: +46 (0)10 7143042
> > ECN: 852-43042
> > ECC: 852-19042
> > Mobile: +46 (0)730 783289
> > Visit http://labs.ericsson.com !
> > *******************************************
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
> >=20
>=20

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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com>
From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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> Ali,
> The example SDP is syntactically correct but it does not say that the
> rtp/rtcp mux will be used and it does not give the=20
> information to where the
> unicast stream will be sent when the RR sends a RAMS-R.=20

To my knowledge, the first line in the following SDP tells the RRs on =
which port they will receive the retransmission/burst packets.

        m=3Dvideo 41002 RTP/AVPF 99
        i=3DUnicast Retransmission Stream #2 (Ret. and Rapid Acq. =
Support)
        c=3DIN IP4 192.0.2.1
        a=3Drtpmap:99 rtx/90000
        a=3Drtcp:41003
        a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
        a=3Dmid:4

There is a typo, you are right. That "a=3Drecvonly" line should only =
exist in the SDP sent to RRs. In the SDP sent to RS, we should rather =
have "a=3Dsendonly". I will make this correction in the next version.

The feedback target for the SSM session and the RTCP port for the =
retransmission session are also defined in the SDP.

Hope this clarifies.

BR,
-acbegen
=20
>=20
> I am not sure why the unicast stream m-line has a port number with an
> attribute of recvonly. What is the use case for that. The=20
> only information
> that the RR will need is the RTCP port on the RS to where to=20
> send the RAMS-R
> message.
> Roni
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Friday, April 24, 2009 10:37 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> This is a part of an example SDP sent to RS and RR in a SAP=20
> announcement,
> for example. So, the SDP describes what both parties should=20
> do (RR cannot
> say that he wants to receive this multicast on its favorite port). The
> individual SDPs sent to RR or RS may include other portions=20
> of descriptions
> that will contain specific information.=20
>=20
> -acbegen
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > Sent: Friday, April 24, 2009 12:23 PM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Ali,
> > I think you get it wrong, this SDP is from the RS and not the=20
> > RR so the RS
> > cannot specify to which address it will send it can only=20
> > specify to which
> > address it can receive RTP stream. In this SDP the relevant=20
> > information is
> > that the request for retransmission will be sent by the RR to=20
> > port 41003
> > Roni
> >=20
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Friday, April 24, 2009 10:01 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > Oh, I see. The burst goes to the port that we would=20
> normally send the
> > retransmissions. For example, consider the SDP from the draft:
> >=20
> >         m=3Dvideo 41000 RTP/AVPF 98
> >         i=3DPrimary Multicast Stream #2
> >         c=3DIN IP4 224.1.1.2/255
> >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> >         a=3Drtpmap:98 MP2T/90000
> >         a=3Drtcp:41001 IN IP4 192.0.2.1
> >         a=3Drtcp-fb:98 nack
> >         a=3Drtcp-fb:98 nack ssli
> >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> >         a=3Dmid:3
> >         m=3Dvideo 41002 RTP/AVPF 99
> >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > Acq. Support)
> >         c=3DIN IP4 192.0.2.1
> >         a=3Drecvonly
> >         a=3Drtpmap:99 rtx/90000
> >         a=3Drtcp:41003
> >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> >         a=3Dmid:4
> >=20
> > In this case, the burst will go to port 41002 on the RR. The RAMS-I
> > message(s), which is an RTCP feedback message, will go to=20
> port 41003.
> >=20
> > HTH,
> > -acbegen
> >=20
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > Sent: Friday, April 24, 2009 11:44 AM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions (RAMS)" available
> > >=20
> > > Ali,
> > > I will try to explain in simple way
> > >=20
> > > When the RS receives the RAMS-R it need to start sending an=20
> > > RTP stream to
> > > the RR.=20
> > > In order to send a unicast packet, the RS needs to know a=20
> > > transport address
> > > on the RR to where the RTP stream will be sent. The current=20
> > > draft does not
> > > say how the RS knows this address. There is no SDP from RR to=20
> > > RS like you
> > > mention in your response.
> > > This is why I say that the current draft does not specify a=20
> > > solution that
> > > can be implemented as a working solution
> > > Roni
> > >=20
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > Sent: Friday, April 24, 2009 7:38 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions
> > > (RAMS)" available
> > >=20
> > > Hi Roni,=20
> > >=20
> > > > -----Original Message-----
> > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > Behalf Of Roni Even
> > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > To: avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions (RAMS)" available
> > > >=20
> > > > Hi,
> > > >=20
> > > > I think that the current draft does not give a description of=20
> > > > a system that works since there is no text explaining how the=20
> > >=20
> > > What do you mean by "a system that works"?=20
> > >=20
> > > > RS knows the unicast transport address on the RR to where to=20
> > > > send the stream.=20
> > >=20
> > > Once RS receives the request packet from an RR, RS knows its=20
> > > address. Ports
> > > are defined in the SDP. If you are asking about "NAT" issues,=20
> > > we have a
> > > section for it, and we plan to complete it as we move=20
> > > forward. It is not as
> > > critical as the other parts for now.
> > > =20
> > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > otherwise the RAMS-R should have an optional parameter that=20
> > > > supplies this information and a flag for using RTP/RTCP mux.
> > >=20
> > > IMO, we cannot mandate muxing as it is not required to make=20
> > > RAMS work. If
> > > muxing is supported, the SDP should reflect it.
> > >=20
> > > BR,
> > > -acbegen
> > > =20
> > > > Thanks
> > > >=20
> > > > Roni Even
> > > >=20
> > > > =20
> > > >=20
> > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > Behalf Of Bill Ver Steeg (versteb)
> > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > To: avt@ietf.org
> > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > > > RTP Sessions (RAMS)" available
> > > >=20
> > > > =20
> > > >=20
> > > > There is a new draft of the "Rapid Acquisition of Multicast=20
> > > > RTP Sessions (RAMS)" draft available at=20
> > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > >=20
> > > > =20
> > > >=20
> > > > We have incorporated the changes from the technical breakout=20
> > > > session in San Francisco. The major changes in this version=20
> > > > of the draft include=20
> > > >=20
> > > > 1- Changing the document title to avoid confusion with other=20
> > > > ongoing "synchronization" drafts
> > > >=20
> > > > 2- changing the message names to reflect the title change
> > > >=20
> > > > 3- clarification of the RTCP message semantics, including=20
> > > > changes to the "Request" and "Inform" messages
> > > >=20
> > > > 4- additional description/motivation for the various message=20
> > > > flows has been added
> > > >=20
> > > > 5- RTP/RTCP muxing has been added
> > > >=20
> > > > =20
> > > >=20
> > > > =20
> > > >=20
> > > > We hope to make this a Working Group item, and will change=20
> > > > the name of the draft to avoid conflicts with other=20
> > > > "synchronization" drafts at that time.
> > > >=20
> > > > =20
> > > >=20
> > > > Bill VerSteeg
> > > >=20
> > > > =20
> > > >=20
> > > > =20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

From ron.even.tlv@gmail.com  Sat Apr 25 13:12:06 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com>
Date: Sat, 25 Apr 2009 23:11:28 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
Can you please write three addresses from this strange SDP.

1. The address and port of multicast

2. The address and port of the RS where the RTCP FB should go to

3. The address and port on the RR where the unicast stream should be sent to

Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Saturday, April 25, 2009 11:02 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

> Ali,
> The example SDP is syntactically correct but it does not say that the
> rtp/rtcp mux will be used and it does not give the 
> information to where the
> unicast stream will be sent when the RR sends a RAMS-R. 

To my knowledge, the first line in the following SDP tells the RRs on which
port they will receive the retransmission/burst packets.

        m=video 41002 RTP/AVPF 99
        i=Unicast Retransmission Stream #2 (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1
        a=rtpmap:99 rtx/90000
        a=rtcp:41003
        a=fmtp:99 apt=98; rtx-time=5000
        a=mid:4

There is a typo, you are right. That "a=recvonly" line should only exist in
the SDP sent to RRs. In the SDP sent to RS, we should rather have
"a=sendonly". I will make this correction in the next version.

The feedback target for the SSM session and the RTCP port for the
retransmission session are also defined in the SDP.

Hope this clarifies.

BR,
-acbegen
 
> 
> I am not sure why the unicast stream m-line has a port number with an
> attribute of recvonly. What is the use case for that. The 
> only information
> that the RR will need is the RTCP port on the RS to where to 
> send the RAMS-R
> message.
> Roni
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Friday, April 24, 2009 10:37 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> This is a part of an example SDP sent to RS and RR in a SAP 
> announcement,
> for example. So, the SDP describes what both parties should 
> do (RR cannot
> say that he wants to receive this multicast on its favorite port). The
> individual SDPs sent to RR or RS may include other portions 
> of descriptions
> that will contain specific information. 
> 
> -acbegen
> 
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > Sent: Friday, April 24, 2009 12:23 PM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions (RAMS)" available
> > 
> > Ali,
> > I think you get it wrong, this SDP is from the RS and not the 
> > RR so the RS
> > cannot specify to which address it will send it can only 
> > specify to which
> > address it can receive RTP stream. In this SDP the relevant 
> > information is
> > that the request for retransmission will be sent by the RR to 
> > port 41003
> > Roni
> > 
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Friday, April 24, 2009 10:01 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > Oh, I see. The burst goes to the port that we would 
> normally send the
> > retransmissions. For example, consider the SDP from the draft:
> > 
> >         m=video 41000 RTP/AVPF 98
> >         i=Primary Multicast Stream #2
> >         c=IN IP4 224.1.1.2/255
> >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> >         a=rtpmap:98 MP2T/90000
> >         a=rtcp:41001 IN IP4 192.0.2.1
> >         a=rtcp-fb:98 nack
> >         a=rtcp-fb:98 nack ssli
> >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> >         a=mid:3
> >         m=video 41002 RTP/AVPF 99
> >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > Acq. Support)
> >         c=IN IP4 192.0.2.1
> >         a=recvonly
> >         a=rtpmap:99 rtx/90000
> >         a=rtcp:41003
> >         a=fmtp:99 apt=98; rtx-time=5000
> >         a=mid:4
> > 
> > In this case, the burst will go to port 41002 on the RR. The RAMS-I
> > message(s), which is an RTCP feedback message, will go to 
> port 41003.
> > 
> > HTH,
> > -acbegen
> > 
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > Sent: Friday, April 24, 2009 11:44 AM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Ali,
> > > I will try to explain in simple way
> > > 
> > > When the RS receives the RAMS-R it need to start sending an 
> > > RTP stream to
> > > the RR. 
> > > In order to send a unicast packet, the RS needs to know a 
> > > transport address
> > > on the RR to where the RTP stream will be sent. The current 
> > > draft does not
> > > say how the RS knows this address. There is no SDP from RR to 
> > > RS like you
> > > mention in your response.
> > > This is why I say that the current draft does not specify a 
> > > solution that
> > > can be implemented as a working solution
> > > Roni
> > > 
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Friday, April 24, 2009 7:38 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > Hi Roni, 
> > > 
> > > > -----Original Message-----
> > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > Behalf Of Roni Even
> > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > To: avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions (RAMS)" available
> > > > 
> > > > Hi,
> > > > 
> > > > I think that the current draft does not give a description of 
> > > > a system that works since there is no text explaining how the 
> > > 
> > > What do you mean by "a system that works"? 
> > > 
> > > > RS knows the unicast transport address on the RR to where to 
> > > > send the stream. 
> > > 
> > > Once RS receives the request packet from an RR, RS knows its 
> > > address. Ports
> > > are defined in the SDP. If you are asking about "NAT" issues, 
> > > we have a
> > > section for it, and we plan to complete it as we move 
> > > forward. It is not as
> > > critical as the other parts for now.
> > >  
> > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > otherwise the RAMS-R should have an optional parameter that 
> > > > supplies this information and a flag for using RTP/RTCP mux.
> > > 
> > > IMO, we cannot mandate muxing as it is not required to make 
> > > RAMS work. If
> > > muxing is supported, the SDP should reflect it.
> > > 
> > > BR,
> > > -acbegen
> > >  
> > > > Thanks
> > > > 
> > > > Roni Even
> > > > 
> > > >  
> > > > 
> > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > Behalf Of Bill Ver Steeg (versteb)
> > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > To: avt@ietf.org
> > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > > RTP Sessions (RAMS)" available
> > > > 
> > > >  
> > > > 
> > > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > > RTP Sessions (RAMS)" draft available at 
> > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > 
> > > >  
> > > > 
> > > > We have incorporated the changes from the technical breakout 
> > > > session in San Francisco. The major changes in this version 
> > > > of the draft include 
> > > > 
> > > > 1- Changing the document title to avoid confusion with other 
> > > > ongoing "synchronization" drafts
> > > > 
> > > > 2- changing the message names to reflect the title change
> > > > 
> > > > 3- clarification of the RTCP message semantics, including 
> > > > changes to the "Request" and "Inform" messages
> > > > 
> > > > 4- additional description/motivation for the various message 
> > > > flows has been added
> > > > 
> > > > 5- RTP/RTCP muxing has been added
> > > > 
> > > >  
> > > > 
> > > >  
> > > > 
> > > > We hope to make this a Working Group item, and will change 
> > > > the name of the draft to avoid conflicts with other 
> > > > "synchronization" drafts at that time.
> > > > 
> > > >  
> > > > 
> > > > Bill VerSteeg
> > > > 
> > > >  
> > > > 
> > > >  
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


From abegen@cisco.com  Sat Apr 25 13:29:04 2009
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Here is the SDP.

        m=3Dvideo 41000 RTP/AVPF 98
        i=3DPrimary Multicast Stream #2
        c=3DIN IP4 224.1.1.2/255
        a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2

Source (192.0.2.2) sends the rtp packets to the multicast group =
(224.1.1.2) with a dest port 41000.

        a=3Drtpmap:98 MP2T/90000
        a=3Drtcp:41001 IN IP4 192.0.2.1

The feedback target (RS) address for this SSM session is 192.0.2.1 and =
its port is 41001. This is the address/port where RR sends the RAMS-R, =
or receiver reports for the SSM session.

        a=3Drtcp-fb:98 nack
        a=3Drtcp-fb:98 nack ssli
        a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
        a=3Dmid:3
        m=3Dvideo 41002 RTP/AVPF 99

The retransmission packets go to port 41002. This is the port RRs listen =
to for retransmission and RAMS.

        i=3DUnicast Retransmission Stream #2 (Ret. and Rapid Acq. =
Support)
        c=3DIN IP4 192.0.2.1

This is where the retransmission packets come from, same as the feedback =
target (in this example).

        a=3Drtpmap:99 rtx/90000
        a=3Drtcp:41003

This is where the RTCP packets for the retransmission session go. For =
example, RAMS-I goes to this port on RRs. RAMS-T goes to this port on =
RS.

        a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
        a=3Dmid:4


-acbegen
=20

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Saturday, April 25, 2009 1:11 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Ali,
> Can you please write three addresses from this strange SDP.
>=20
> 1. The address and port of multicast
>=20
> 2. The address and port of the RS where the RTCP FB should go to
>=20
> 3. The address and port on the RR where the unicast stream=20
> should be sent to
>=20
> Roni
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Saturday, April 25, 2009 11:02 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> > Ali,
> > The example SDP is syntactically correct but it does not=20
> say that the
> > rtp/rtcp mux will be used and it does not give the=20
> > information to where the
> > unicast stream will be sent when the RR sends a RAMS-R.=20
>=20
> To my knowledge, the first line in the following SDP tells=20
> the RRs on which
> port they will receive the retransmission/burst packets.
>=20
>         m=3Dvideo 41002 RTP/AVPF 99
>         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> Acq. Support)
>         c=3DIN IP4 192.0.2.1
>         a=3Drtpmap:99 rtx/90000
>         a=3Drtcp:41003
>         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
>         a=3Dmid:4
>=20
> There is a typo, you are right. That "a=3Drecvonly" line should=20
> only exist in
> the SDP sent to RRs. In the SDP sent to RS, we should rather have
> "a=3Dsendonly". I will make this correction in the next version.
>=20
> The feedback target for the SSM session and the RTCP port for the
> retransmission session are also defined in the SDP.
>=20
> Hope this clarifies.
>=20
> BR,
> -acbegen
> =20
> >=20
> > I am not sure why the unicast stream m-line has a port=20
> number with an
> > attribute of recvonly. What is the use case for that. The=20
> > only information
> > that the RR will need is the RTCP port on the RS to where to=20
> > send the RAMS-R
> > message.
> > Roni
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Friday, April 24, 2009 10:37 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > This is a part of an example SDP sent to RS and RR in a SAP=20
> > announcement,
> > for example. So, the SDP describes what both parties should=20
> > do (RR cannot
> > say that he wants to receive this multicast on its favorite=20
> port). The
> > individual SDPs sent to RR or RS may include other portions=20
> > of descriptions
> > that will contain specific information.=20
> >=20
> > -acbegen
> >=20
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > Sent: Friday, April 24, 2009 12:23 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions (RAMS)" available
> > >=20
> > > Ali,
> > > I think you get it wrong, this SDP is from the RS and not the=20
> > > RR so the RS
> > > cannot specify to which address it will send it can only=20
> > > specify to which
> > > address it can receive RTP stream. In this SDP the relevant=20
> > > information is
> > > that the request for retransmission will be sent by the RR to=20
> > > port 41003
> > > Roni
> > >=20
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > Sent: Friday, April 24, 2009 10:01 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions
> > > (RAMS)" available
> > >=20
> > > Oh, I see. The burst goes to the port that we would=20
> > normally send the
> > > retransmissions. For example, consider the SDP from the draft:
> > >=20
> > >         m=3Dvideo 41000 RTP/AVPF 98
> > >         i=3DPrimary Multicast Stream #2
> > >         c=3DIN IP4 224.1.1.2/255
> > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > >         a=3Drtpmap:98 MP2T/90000
> > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > >         a=3Drtcp-fb:98 nack
> > >         a=3Drtcp-fb:98 nack ssli
> > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > >         a=3Dmid:3
> > >         m=3Dvideo 41002 RTP/AVPF 99
> > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > Acq. Support)
> > >         c=3DIN IP4 192.0.2.1
> > >         a=3Drecvonly
> > >         a=3Drtpmap:99 rtx/90000
> > >         a=3Drtcp:41003
> > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > >         a=3Dmid:4
> > >=20
> > > In this case, the burst will go to port 41002 on the RR.=20
> The RAMS-I
> > > message(s), which is an RTCP feedback message, will go to=20
> > port 41003.
> > >=20
> > > HTH,
> > > -acbegen
> > >=20
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions (RAMS)" available
> > > >=20
> > > > Ali,
> > > > I will try to explain in simple way
> > > >=20
> > > > When the RS receives the RAMS-R it need to start sending an=20
> > > > RTP stream to
> > > > the RR.=20
> > > > In order to send a unicast packet, the RS needs to know a=20
> > > > transport address
> > > > on the RR to where the RTP stream will be sent. The current=20
> > > > draft does not
> > > > say how the RS knows this address. There is no SDP from RR to=20
> > > > RS like you
> > > > mention in your response.
> > > > This is why I say that the current draft does not specify a=20
> > > > solution that
> > > > can be implemented as a working solution
> > > > Roni
> > > >=20
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > >=20
> > > > Hi Roni,=20
> > > >=20
> > > > > -----Original Message-----
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > Behalf Of Roni Even
> > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > To: avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions (RAMS)" available
> > > > >=20
> > > > > Hi,
> > > > >=20
> > > > > I think that the current draft does not give a description of=20
> > > > > a system that works since there is no text explaining how the=20
> > > >=20
> > > > What do you mean by "a system that works"?=20
> > > >=20
> > > > > RS knows the unicast transport address on the RR to where to=20
> > > > > send the stream.=20
> > > >=20
> > > > Once RS receives the request packet from an RR, RS knows its=20
> > > > address. Ports
> > > > are defined in the SDP. If you are asking about "NAT" issues,=20
> > > > we have a
> > > > section for it, and we plan to complete it as we move=20
> > > > forward. It is not as
> > > > critical as the other parts for now.
> > > > =20
> > > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > > otherwise the RAMS-R should have an optional parameter that=20
> > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > >=20
> > > > IMO, we cannot mandate muxing as it is not required to make=20
> > > > RAMS work. If
> > > > muxing is supported, the SDP should reflect it.
> > > >=20
> > > > BR,
> > > > -acbegen
> > > > =20
> > > > > Thanks
> > > > >=20
> > > > > Roni Even
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > To: avt@ietf.org
> > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > > > > RTP Sessions (RAMS)" available
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > There is a new draft of the "Rapid Acquisition of Multicast=20
> > > > > RTP Sessions (RAMS)" draft available at=20
> > > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > We have incorporated the changes from the technical breakout=20
> > > > > session in San Francisco. The major changes in this version=20
> > > > > of the draft include=20
> > > > >=20
> > > > > 1- Changing the document title to avoid confusion with other=20
> > > > > ongoing "synchronization" drafts
> > > > >=20
> > > > > 2- changing the message names to reflect the title change
> > > > >=20
> > > > > 3- clarification of the RTCP message semantics, including=20
> > > > > changes to the "Request" and "Inform" messages
> > > > >=20
> > > > > 4- additional description/motivation for the various message=20
> > > > > flows has been added
> > > > >=20
> > > > > 5- RTP/RTCP muxing has been added
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > We hope to make this a Working Group item, and will change=20
> > > > > the name of the draft to avoid conflicts with other=20
> > > > > "synchronization" drafts at that time.
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > Bill VerSteeg
> > > > >=20
> > > > > =20
> > > > >=20
> > > > > =20
> > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

From abegen@cisco.com  Sat Apr 25 13:49:20 2009
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Ingemar Johansson S" <ingemar.s.johansson@ericsson.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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> > I would not say it would be "fairly straighforward." To make=20
> > the multicast bursting effective, many parameters need to be=20
> > adjusted properly and optimized. Ow, it can easily degrade=20
> > the performance even at the expense of additional complexity.
> >=20
> > A concern I have which is not addressed in your document is=20
> > that to make the proposal make sense, Td needs to be small=20
> > enough. But, to ensure that all receivers join the burst=20
> > mcast session before the burst starts, Td cannot be very=20
> > small and RS should not admit last-minute RAMS requests for=20
> > that multicast burst (they need to wait for the next batch).=20
> > This is because if some receivers miss the first few packets=20
> > of the burst, their acquisition will fail. Also, how many=20
> > clients can the retransmission server aggregate in a single=20
> > burst batch in a practical scenario? The eligible requests=20
> > should be received within +/- a few hundereds of=20
> > milliseconds. How practical is this, I wonder.
> The neat thing is that during low load conditions Td can be=20
> made very low such that in practice only one user is served=20
> by each request. This can be implemented with either unicast=20
> or multicast bursts that only serve one STB. As he load=20
> increases Td increases and more users that send a requests=20
> will be served by one multicast channel. As Td is limited=20
> some requests will be too late but they can too be collected=20
> and served by additional multicast feeds. Then depending on=20
> the distribuion the tail may be handled with unicast bursts.=20
> This is more or less an optimization issue on the servers.

Agree.

> The idea is not that the unicast and multicast alternatives=20
> should compete with one another, rather they should=20
> complement each other, due to the highly dynamic nature of=20
> channel switching behavior in a population both alternatives=20
> are useful. I do believe that the multicast solution can help=20
> to reduce deplyment cost.=20

It can. However, remember that in a highly-loaded environment, if RS =
cannot provide (unicast) acceleration, it can deny the request. For the =
sake of doing multicast bursting (to save bandwidth), if RRs experience =
a slower acquisition (due to a large Td, for example) compared to a =
simple join, that is not good.

> I have tried to look for statistics on channel switching=20
> behavior and of what I have seen sofar the "peak-to-average"=20
> ratio of channel swithing density can be more than 5. This is=20
> seen in a master thesis with the title "TPTV traffic=20
> measuring & modeling" by Geng Yu (have not found it published=20
> anywhere yet).=20

I don't think I have that study, either. But, there is a recent study in =
IMC 2008. Here is the link.
http://portal.acm.org/ft_gateway.cfm?id=3D1452529&type=3Dpdf&coll=3Dporta=
l&dl=3DACM

The results are in general more or less similar to what we found in one =
of our own studies. The figure you will find interesting is Figure 14.=20

For the most popular channel, the entire data set shows that on average =
14 viewers tune into that channel per second (Figure 14a). Of course, =
there are occasions where a lot more viewers may tune at approximately =
the same time (within the same second), but this is for the top-ranked =
channel and the number 14 includes all viewers in the data set, which is =
about 250K (This is what I got from the text). If we consider only a =
subset of these viewers will be served by the same RS, the concurrency =
we will get in a single server will be far smaller, impying that =
multicast burst will not be needed most of the time.

Different studies may suggest different numbers based on the regional =
characteristics. So, I don't think we should read too much into this =
study. If there are other studies, please post them here. So, we can =
make a better judgment.

> > The selection of a common burst bitrate is also critical. If=20
> > everybody supports a different burst bitrate and there is a=20
> > large variation, multicast bursting will perform poorly for=20
> > the well-connected clients. One solution could be to include=20
> > the receivers with an available bw that is smaller than R1=20
> > and larger than R2 in the multicast pool. The very fast and=20
> > very slow receivers should be still served via unicast.
> Sounds reasonable.
>=20
> >=20
> > My personal view is that the multicast bursting proposal will=20
> > be a nice solution when the issues above (and possible=20
> > others) are addressed, however, I think it should be a=20
> > separate draft that normatively references the unicast draft.=20
> > This way, we can keep the unicast draft more focused.
> Not sure that I agree here, it all depends on how the=20
> signaling is standardized. For instance an STB likely needs=20
> to declare its unicast/multicast preference or capability in=20
> the RAMS-R as the SDP's are likely only declarative.=20

There can be a TLV in the request to do this, no?=20

BR,
-acbegen

From ron.even.tlv@gmail.com  Sat Apr 25 14:24:34 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com> <49f36ee2.0437560a.11df.1eef@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com>
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
See inline
Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Saturday, April 25, 2009 11:30 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Here is the SDP.

        m=video 41000 RTP/AVPF 98
        i=Primary Multicast Stream #2
        c=IN IP4 224.1.1.2/255
        a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2

Source (192.0.2.2) sends the rtp packets to the multicast group (224.1.1.2)
with a dest port 41000.

        a=rtpmap:98 MP2T/90000
        a=rtcp:41001 IN IP4 192.0.2.1

The feedback target (RS) address for this SSM session is 192.0.2.1 and its
port is 41001. This is the address/port where RR sends the RAMS-R, or
receiver reports for the SSM session.

        a=rtcp-fb:98 nack
        a=rtcp-fb:98 nack ssli
        a=ssrc:123321 cname:iptv-ch32@rams.example.com
        a=mid:3
        m=video 41002 RTP/AVPF 99

The retransmission packets go to port 41002. This is the port RRs listen to
for retransmission and RAMS.

Roni E: This implies that the multicast source defines the port that all
clients MUST use to receive the unicast information from the Burst source.  

        i=Unicast Retransmission Stream #2 (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1

This is where the retransmission packets come from, same as the feedback
target (in this example).

Roni E: If this is the address of the feedback target it belongs to the
above m-line (m=video 41002 RTP/AVPF 99) that according to you gives the
port on the RR and not on the RS. The problem is that according to SDP the
combination of the c= and port on the m-line provides the transport address
to where a stream will be sent so according to SDP this m-line specify an
address and port on the Feedback Target and not on the RR and only to this
address RTP payload type 99 will be sent. I am positive that this is not
what you want.

        a=rtpmap:99 rtx/90000
        a=rtcp:41003

This is where the RTCP packets for the retransmission session go. For
example, RAMS-I goes to this port on RRs. RAMS-T goes to this port on RS.

Roni E: So now you assume that the RR must use the same port as the FT for
this communication. I could not find any place that one m-line gives the
address for both sides. This RTCP port is on 192.0.2.1 so it will be used
for RAMS-R but not for RAMS-I.

        a=fmtp:99 apt=98; rtx-time=5000
        a=mid:4


-acbegen
 

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Saturday, April 25, 2009 1:11 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> Can you please write three addresses from this strange SDP.
> 
> 1. The address and port of multicast
> 
> 2. The address and port of the RS where the RTCP FB should go to
> 
> 3. The address and port on the RR where the unicast stream 
> should be sent to
> 
> Roni
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Saturday, April 25, 2009 11:02 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> > Ali,
> > The example SDP is syntactically correct but it does not 
> say that the
> > rtp/rtcp mux will be used and it does not give the 
> > information to where the
> > unicast stream will be sent when the RR sends a RAMS-R. 
> 
> To my knowledge, the first line in the following SDP tells 
> the RRs on which
> port they will receive the retransmission/burst packets.
> 
>         m=video 41002 RTP/AVPF 99
>         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> Acq. Support)
>         c=IN IP4 192.0.2.1
>         a=rtpmap:99 rtx/90000
>         a=rtcp:41003
>         a=fmtp:99 apt=98; rtx-time=5000
>         a=mid:4
> 
> There is a typo, you are right. That "a=recvonly" line should 
> only exist in
> the SDP sent to RRs. In the SDP sent to RS, we should rather have
> "a=sendonly". I will make this correction in the next version.
> 
> The feedback target for the SSM session and the RTCP port for the
> retransmission session are also defined in the SDP.
> 
> Hope this clarifies.
> 
> BR,
> -acbegen
>  
> > 
> > I am not sure why the unicast stream m-line has a port 
> number with an
> > attribute of recvonly. What is the use case for that. The 
> > only information
> > that the RR will need is the RTCP port on the RS to where to 
> > send the RAMS-R
> > message.
> > Roni
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Friday, April 24, 2009 10:37 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > This is a part of an example SDP sent to RS and RR in a SAP 
> > announcement,
> > for example. So, the SDP describes what both parties should 
> > do (RR cannot
> > say that he wants to receive this multicast on its favorite 
> port). The
> > individual SDPs sent to RR or RS may include other portions 
> > of descriptions
> > that will contain specific information. 
> > 
> > -acbegen
> > 
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > Sent: Friday, April 24, 2009 12:23 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Ali,
> > > I think you get it wrong, this SDP is from the RS and not the 
> > > RR so the RS
> > > cannot specify to which address it will send it can only 
> > > specify to which
> > > address it can receive RTP stream. In this SDP the relevant 
> > > information is
> > > that the request for retransmission will be sent by the RR to 
> > > port 41003
> > > Roni
> > > 
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Friday, April 24, 2009 10:01 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > Oh, I see. The burst goes to the port that we would 
> > normally send the
> > > retransmissions. For example, consider the SDP from the draft:
> > > 
> > >         m=video 41000 RTP/AVPF 98
> > >         i=Primary Multicast Stream #2
> > >         c=IN IP4 224.1.1.2/255
> > >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > >         a=rtpmap:98 MP2T/90000
> > >         a=rtcp:41001 IN IP4 192.0.2.1
> > >         a=rtcp-fb:98 nack
> > >         a=rtcp-fb:98 nack ssli
> > >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> > >         a=mid:3
> > >         m=video 41002 RTP/AVPF 99
> > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > Acq. Support)
> > >         c=IN IP4 192.0.2.1
> > >         a=recvonly
> > >         a=rtpmap:99 rtx/90000
> > >         a=rtcp:41003
> > >         a=fmtp:99 apt=98; rtx-time=5000
> > >         a=mid:4
> > > 
> > > In this case, the burst will go to port 41002 on the RR. 
> The RAMS-I
> > > message(s), which is an RTCP feedback message, will go to 
> > port 41003.
> > > 
> > > HTH,
> > > -acbegen
> > > 
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions (RAMS)" available
> > > > 
> > > > Ali,
> > > > I will try to explain in simple way
> > > > 
> > > > When the RS receives the RAMS-R it need to start sending an 
> > > > RTP stream to
> > > > the RR. 
> > > > In order to send a unicast packet, the RS needs to know a 
> > > > transport address
> > > > on the RR to where the RTP stream will be sent. The current 
> > > > draft does not
> > > > say how the RS knows this address. There is no SDP from RR to 
> > > > RS like you
> > > > mention in your response.
> > > > This is why I say that the current draft does not specify a 
> > > > solution that
> > > > can be implemented as a working solution
> > > > Roni
> > > > 
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > > 
> > > > Hi Roni, 
> > > > 
> > > > > -----Original Message-----
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Roni Even
> > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > To: avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions (RAMS)" available
> > > > > 
> > > > > Hi,
> > > > > 
> > > > > I think that the current draft does not give a description of 
> > > > > a system that works since there is no text explaining how the 
> > > > 
> > > > What do you mean by "a system that works"? 
> > > > 
> > > > > RS knows the unicast transport address on the RR to where to 
> > > > > send the stream. 
> > > > 
> > > > Once RS receives the request packet from an RR, RS knows its 
> > > > address. Ports
> > > > are defined in the SDP. If you are asking about "NAT" issues, 
> > > > we have a
> > > > section for it, and we plan to complete it as we move 
> > > > forward. It is not as
> > > > critical as the other parts for now.
> > > >  
> > > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > > otherwise the RAMS-R should have an optional parameter that 
> > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > 
> > > > IMO, we cannot mandate muxing as it is not required to make 
> > > > RAMS work. If
> > > > muxing is supported, the SDP should reflect it.
> > > > 
> > > > BR,
> > > > -acbegen
> > > >  
> > > > > Thanks
> > > > > 
> > > > > Roni Even
> > > > > 
> > > > >  
> > > > > 
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > To: avt@ietf.org
> > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" available
> > > > > 
> > > > >  
> > > > > 
> > > > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" draft available at 
> > > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > > 
> > > > >  
> > > > > 
> > > > > We have incorporated the changes from the technical breakout 
> > > > > session in San Francisco. The major changes in this version 
> > > > > of the draft include 
> > > > > 
> > > > > 1- Changing the document title to avoid confusion with other 
> > > > > ongoing "synchronization" drafts
> > > > > 
> > > > > 2- changing the message names to reflect the title change
> > > > > 
> > > > > 3- clarification of the RTCP message semantics, including 
> > > > > changes to the "Request" and "Inform" messages
> > > > > 
> > > > > 4- additional description/motivation for the various message 
> > > > > flows has been added
> > > > > 
> > > > > 5- RTP/RTCP muxing has been added
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > We hope to make this a Working Group item, and will change 
> > > > > the name of the draft to avoid conflicts with other 
> > > > > "synchronization" drafts at that time.
> > > > > 
> > > > >  
> > > > > 
> > > > > Bill VerSteeg
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > 
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


From ron.even.tlv@gmail.com  Sat Apr 25 14:35:28 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com> <49f36ee2.0437560a.11df.1eef@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com>
Date: Sun, 26 Apr 2009 00:34:51 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
Another comment
I thought that RAMS-R and RAMS-T go to the same port but in the example you
say that RAMS-R go to port 41001 and RAMS-T go to port 41003

Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Saturday, April 25, 2009 11:30 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Here is the SDP.

        m=video 41000 RTP/AVPF 98
        i=Primary Multicast Stream #2
        c=IN IP4 224.1.1.2/255
        a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2

Source (192.0.2.2) sends the rtp packets to the multicast group (224.1.1.2)
with a dest port 41000.

        a=rtpmap:98 MP2T/90000
        a=rtcp:41001 IN IP4 192.0.2.1

The feedback target (RS) address for this SSM session is 192.0.2.1 and its
port is 41001. This is the address/port where RR sends the RAMS-R, or
receiver reports for the SSM session.

        a=rtcp-fb:98 nack
        a=rtcp-fb:98 nack ssli
        a=ssrc:123321 cname:iptv-ch32@rams.example.com
        a=mid:3
        m=video 41002 RTP/AVPF 99

The retransmission packets go to port 41002. This is the port RRs listen to
for retransmission and RAMS.

        i=Unicast Retransmission Stream #2 (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1

This is where the retransmission packets come from, same as the feedback
target (in this example).

        a=rtpmap:99 rtx/90000
        a=rtcp:41003

This is where the RTCP packets for the retransmission session go. For
example, RAMS-I goes to this port on RRs. RAMS-T goes to this port on RS.

        a=fmtp:99 apt=98; rtx-time=5000
        a=mid:4


-acbegen
 

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Saturday, April 25, 2009 1:11 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> Can you please write three addresses from this strange SDP.
> 
> 1. The address and port of multicast
> 
> 2. The address and port of the RS where the RTCP FB should go to
> 
> 3. The address and port on the RR where the unicast stream 
> should be sent to
> 
> Roni
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Saturday, April 25, 2009 11:02 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> > Ali,
> > The example SDP is syntactically correct but it does not 
> say that the
> > rtp/rtcp mux will be used and it does not give the 
> > information to where the
> > unicast stream will be sent when the RR sends a RAMS-R. 
> 
> To my knowledge, the first line in the following SDP tells 
> the RRs on which
> port they will receive the retransmission/burst packets.
> 
>         m=video 41002 RTP/AVPF 99
>         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> Acq. Support)
>         c=IN IP4 192.0.2.1
>         a=rtpmap:99 rtx/90000
>         a=rtcp:41003
>         a=fmtp:99 apt=98; rtx-time=5000
>         a=mid:4
> 
> There is a typo, you are right. That "a=recvonly" line should 
> only exist in
> the SDP sent to RRs. In the SDP sent to RS, we should rather have
> "a=sendonly". I will make this correction in the next version.
> 
> The feedback target for the SSM session and the RTCP port for the
> retransmission session are also defined in the SDP.
> 
> Hope this clarifies.
> 
> BR,
> -acbegen
>  
> > 
> > I am not sure why the unicast stream m-line has a port 
> number with an
> > attribute of recvonly. What is the use case for that. The 
> > only information
> > that the RR will need is the RTCP port on the RS to where to 
> > send the RAMS-R
> > message.
> > Roni
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Friday, April 24, 2009 10:37 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > This is a part of an example SDP sent to RS and RR in a SAP 
> > announcement,
> > for example. So, the SDP describes what both parties should 
> > do (RR cannot
> > say that he wants to receive this multicast on its favorite 
> port). The
> > individual SDPs sent to RR or RS may include other portions 
> > of descriptions
> > that will contain specific information. 
> > 
> > -acbegen
> > 
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > Sent: Friday, April 24, 2009 12:23 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Ali,
> > > I think you get it wrong, this SDP is from the RS and not the 
> > > RR so the RS
> > > cannot specify to which address it will send it can only 
> > > specify to which
> > > address it can receive RTP stream. In this SDP the relevant 
> > > information is
> > > that the request for retransmission will be sent by the RR to 
> > > port 41003
> > > Roni
> > > 
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Friday, April 24, 2009 10:01 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > Oh, I see. The burst goes to the port that we would 
> > normally send the
> > > retransmissions. For example, consider the SDP from the draft:
> > > 
> > >         m=video 41000 RTP/AVPF 98
> > >         i=Primary Multicast Stream #2
> > >         c=IN IP4 224.1.1.2/255
> > >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > >         a=rtpmap:98 MP2T/90000
> > >         a=rtcp:41001 IN IP4 192.0.2.1
> > >         a=rtcp-fb:98 nack
> > >         a=rtcp-fb:98 nack ssli
> > >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> > >         a=mid:3
> > >         m=video 41002 RTP/AVPF 99
> > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > Acq. Support)
> > >         c=IN IP4 192.0.2.1
> > >         a=recvonly
> > >         a=rtpmap:99 rtx/90000
> > >         a=rtcp:41003
> > >         a=fmtp:99 apt=98; rtx-time=5000
> > >         a=mid:4
> > > 
> > > In this case, the burst will go to port 41002 on the RR. 
> The RAMS-I
> > > message(s), which is an RTCP feedback message, will go to 
> > port 41003.
> > > 
> > > HTH,
> > > -acbegen
> > > 
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions (RAMS)" available
> > > > 
> > > > Ali,
> > > > I will try to explain in simple way
> > > > 
> > > > When the RS receives the RAMS-R it need to start sending an 
> > > > RTP stream to
> > > > the RR. 
> > > > In order to send a unicast packet, the RS needs to know a 
> > > > transport address
> > > > on the RR to where the RTP stream will be sent. The current 
> > > > draft does not
> > > > say how the RS knows this address. There is no SDP from RR to 
> > > > RS like you
> > > > mention in your response.
> > > > This is why I say that the current draft does not specify a 
> > > > solution that
> > > > can be implemented as a working solution
> > > > Roni
> > > > 
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > > 
> > > > Hi Roni, 
> > > > 
> > > > > -----Original Message-----
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Roni Even
> > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > To: avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions (RAMS)" available
> > > > > 
> > > > > Hi,
> > > > > 
> > > > > I think that the current draft does not give a description of 
> > > > > a system that works since there is no text explaining how the 
> > > > 
> > > > What do you mean by "a system that works"? 
> > > > 
> > > > > RS knows the unicast transport address on the RR to where to 
> > > > > send the stream. 
> > > > 
> > > > Once RS receives the request packet from an RR, RS knows its 
> > > > address. Ports
> > > > are defined in the SDP. If you are asking about "NAT" issues, 
> > > > we have a
> > > > section for it, and we plan to complete it as we move 
> > > > forward. It is not as
> > > > critical as the other parts for now.
> > > >  
> > > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > > otherwise the RAMS-R should have an optional parameter that 
> > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > 
> > > > IMO, we cannot mandate muxing as it is not required to make 
> > > > RAMS work. If
> > > > muxing is supported, the SDP should reflect it.
> > > > 
> > > > BR,
> > > > -acbegen
> > > >  
> > > > > Thanks
> > > > > 
> > > > > Roni Even
> > > > > 
> > > > >  
> > > > > 
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > To: avt@ietf.org
> > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" available
> > > > > 
> > > > >  
> > > > > 
> > > > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" draft available at 
> > > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > > 
> > > > >  
> > > > > 
> > > > > We have incorporated the changes from the technical breakout 
> > > > > session in San Francisco. The major changes in this version 
> > > > > of the draft include 
> > > > > 
> > > > > 1- Changing the document title to avoid confusion with other 
> > > > > ongoing "synchronization" drafts
> > > > > 
> > > > > 2- changing the message names to reflect the title change
> > > > > 
> > > > > 3- clarification of the RTCP message semantics, including 
> > > > > changes to the "Request" and "Inform" messages
> > > > > 
> > > > > 4- additional description/motivation for the various message 
> > > > > flows has been added
> > > > > 
> > > > > 5- RTP/RTCP muxing has been added
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > We hope to make this a Working Group item, and will change 
> > > > > the name of the draft to avoid conflicts with other 
> > > > > "synchronization" drafts at that time.
> > > > > 
> > > > >  
> > > > > 
> > > > > Bill VerSteeg
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > 
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


From ron.even.tlv@gmail.com  Sat Apr 25 14:58:22 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com> <49f36ee2.0437560a.11df.1eef@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com>
Date: Sun, 26 Apr 2009 00:57:46 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
I looked at RFC 4570 and I assume that port 41001 is the port for the
unicast RTCP reports from the receivers and according to section 3.2.1 of
that RFC you also should have a RTCP-unicast specification. This is for the
multicast receiver reports.

Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Saturday, April 25, 2009 11:30 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Here is the SDP.

        m=video 41000 RTP/AVPF 98
        i=Primary Multicast Stream #2
        c=IN IP4 224.1.1.2/255
        a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2

Source (192.0.2.2) sends the rtp packets to the multicast group (224.1.1.2)
with a dest port 41000.

        a=rtpmap:98 MP2T/90000
        a=rtcp:41001 IN IP4 192.0.2.1

The feedback target (RS) address for this SSM session is 192.0.2.1 and its
port is 41001. This is the address/port where RR sends the RAMS-R, or
receiver reports for the SSM session.

        a=rtcp-fb:98 nack
        a=rtcp-fb:98 nack ssli
        a=ssrc:123321 cname:iptv-ch32@rams.example.com
        a=mid:3
        m=video 41002 RTP/AVPF 99

The retransmission packets go to port 41002. This is the port RRs listen to
for retransmission and RAMS.

        i=Unicast Retransmission Stream #2 (Ret. and Rapid Acq. Support)
        c=IN IP4 192.0.2.1

This is where the retransmission packets come from, same as the feedback
target (in this example).

        a=rtpmap:99 rtx/90000
        a=rtcp:41003

This is where the RTCP packets for the retransmission session go. For
example, RAMS-I goes to this port on RRs. RAMS-T goes to this port on RS.

        a=fmtp:99 apt=98; rtx-time=5000
        a=mid:4


-acbegen
 

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Saturday, April 25, 2009 1:11 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> Can you please write three addresses from this strange SDP.
> 
> 1. The address and port of multicast
> 
> 2. The address and port of the RS where the RTCP FB should go to
> 
> 3. The address and port on the RR where the unicast stream 
> should be sent to
> 
> Roni
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Saturday, April 25, 2009 11:02 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> > Ali,
> > The example SDP is syntactically correct but it does not 
> say that the
> > rtp/rtcp mux will be used and it does not give the 
> > information to where the
> > unicast stream will be sent when the RR sends a RAMS-R. 
> 
> To my knowledge, the first line in the following SDP tells 
> the RRs on which
> port they will receive the retransmission/burst packets.
> 
>         m=video 41002 RTP/AVPF 99
>         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> Acq. Support)
>         c=IN IP4 192.0.2.1
>         a=rtpmap:99 rtx/90000
>         a=rtcp:41003
>         a=fmtp:99 apt=98; rtx-time=5000
>         a=mid:4
> 
> There is a typo, you are right. That "a=recvonly" line should 
> only exist in
> the SDP sent to RRs. In the SDP sent to RS, we should rather have
> "a=sendonly". I will make this correction in the next version.
> 
> The feedback target for the SSM session and the RTCP port for the
> retransmission session are also defined in the SDP.
> 
> Hope this clarifies.
> 
> BR,
> -acbegen
>  
> > 
> > I am not sure why the unicast stream m-line has a port 
> number with an
> > attribute of recvonly. What is the use case for that. The 
> > only information
> > that the RR will need is the RTCP port on the RS to where to 
> > send the RAMS-R
> > message.
> > Roni
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Friday, April 24, 2009 10:37 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > This is a part of an example SDP sent to RS and RR in a SAP 
> > announcement,
> > for example. So, the SDP describes what both parties should 
> > do (RR cannot
> > say that he wants to receive this multicast on its favorite 
> port). The
> > individual SDPs sent to RR or RS may include other portions 
> > of descriptions
> > that will contain specific information. 
> > 
> > -acbegen
> > 
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > Sent: Friday, April 24, 2009 12:23 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Ali,
> > > I think you get it wrong, this SDP is from the RS and not the 
> > > RR so the RS
> > > cannot specify to which address it will send it can only 
> > > specify to which
> > > address it can receive RTP stream. In this SDP the relevant 
> > > information is
> > > that the request for retransmission will be sent by the RR to 
> > > port 41003
> > > Roni
> > > 
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Friday, April 24, 2009 10:01 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > Oh, I see. The burst goes to the port that we would 
> > normally send the
> > > retransmissions. For example, consider the SDP from the draft:
> > > 
> > >         m=video 41000 RTP/AVPF 98
> > >         i=Primary Multicast Stream #2
> > >         c=IN IP4 224.1.1.2/255
> > >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > >         a=rtpmap:98 MP2T/90000
> > >         a=rtcp:41001 IN IP4 192.0.2.1
> > >         a=rtcp-fb:98 nack
> > >         a=rtcp-fb:98 nack ssli
> > >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> > >         a=mid:3
> > >         m=video 41002 RTP/AVPF 99
> > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > Acq. Support)
> > >         c=IN IP4 192.0.2.1
> > >         a=recvonly
> > >         a=rtpmap:99 rtx/90000
> > >         a=rtcp:41003
> > >         a=fmtp:99 apt=98; rtx-time=5000
> > >         a=mid:4
> > > 
> > > In this case, the burst will go to port 41002 on the RR. 
> The RAMS-I
> > > message(s), which is an RTCP feedback message, will go to 
> > port 41003.
> > > 
> > > HTH,
> > > -acbegen
> > > 
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions (RAMS)" available
> > > > 
> > > > Ali,
> > > > I will try to explain in simple way
> > > > 
> > > > When the RS receives the RAMS-R it need to start sending an 
> > > > RTP stream to
> > > > the RR. 
> > > > In order to send a unicast packet, the RS needs to know a 
> > > > transport address
> > > > on the RR to where the RTP stream will be sent. The current 
> > > > draft does not
> > > > say how the RS knows this address. There is no SDP from RR to 
> > > > RS like you
> > > > mention in your response.
> > > > This is why I say that the current draft does not specify a 
> > > > solution that
> > > > can be implemented as a working solution
> > > > Roni
> > > > 
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > > 
> > > > Hi Roni, 
> > > > 
> > > > > -----Original Message-----
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Roni Even
> > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > To: avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions (RAMS)" available
> > > > > 
> > > > > Hi,
> > > > > 
> > > > > I think that the current draft does not give a description of 
> > > > > a system that works since there is no text explaining how the 
> > > > 
> > > > What do you mean by "a system that works"? 
> > > > 
> > > > > RS knows the unicast transport address on the RR to where to 
> > > > > send the stream. 
> > > > 
> > > > Once RS receives the request packet from an RR, RS knows its 
> > > > address. Ports
> > > > are defined in the SDP. If you are asking about "NAT" issues, 
> > > > we have a
> > > > section for it, and we plan to complete it as we move 
> > > > forward. It is not as
> > > > critical as the other parts for now.
> > > >  
> > > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > > otherwise the RAMS-R should have an optional parameter that 
> > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > 
> > > > IMO, we cannot mandate muxing as it is not required to make 
> > > > RAMS work. If
> > > > muxing is supported, the SDP should reflect it.
> > > > 
> > > > BR,
> > > > -acbegen
> > > >  
> > > > > Thanks
> > > > > 
> > > > > Roni Even
> > > > > 
> > > > >  
> > > > > 
> > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > To: avt@ietf.org
> > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" available
> > > > > 
> > > > >  
> > > > > 
> > > > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > > > RTP Sessions (RAMS)" draft available at 
> > > > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > > 
> > > > >  
> > > > > 
> > > > > We have incorporated the changes from the technical breakout 
> > > > > session in San Francisco. The major changes in this version 
> > > > > of the draft include 
> > > > > 
> > > > > 1- Changing the document title to avoid confusion with other 
> > > > > ongoing "synchronization" drafts
> > > > > 
> > > > > 2- changing the message names to reflect the title change
> > > > > 
> > > > > 3- clarification of the RTCP message semantics, including 
> > > > > changes to the "Request" and "Inform" messages
> > > > > 
> > > > > 4- additional description/motivation for the various message 
> > > > > flows has been added
> > > > > 
> > > > > 5- RTP/RTCP muxing has been added
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > We hope to make this a Working Group item, and will change 
> > > > > the name of the draft to avoid conflicts with other 
> > > > > "synchronization" drafts at that time.
> > > > > 
> > > > >  
> > > > > 
> > > > > Bill VerSteeg
> > > > > 
> > > > >  
> > > > > 
> > > > >  
> > > > > 
> > > > > 
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


From abegen@cisco.com  Sat Apr 25 15:24:56 2009
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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Inline.

>         m=3Dvideo 41000 RTP/AVPF 98
>         i=3DPrimary Multicast Stream #2
>         c=3DIN IP4 224.1.1.2/255
>         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
>=20
> Source (192.0.2.2) sends the rtp packets to the multicast=20
> group (224.1.1.2)
> with a dest port 41000.
>=20
>         a=3Drtpmap:98 MP2T/90000
>         a=3Drtcp:41001 IN IP4 192.0.2.1
>=20
> The feedback target (RS) address for this SSM session is=20
> 192.0.2.1 and its
> port is 41001. This is the address/port where RR sends the RAMS-R, or
> receiver reports for the SSM session.
>=20
>         a=3Drtcp-fb:98 nack
>         a=3Drtcp-fb:98 nack ssli
>         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
>         a=3Dmid:3
>         m=3Dvideo 41002 RTP/AVPF 99
>=20
> The retransmission packets go to port 41002. This is the port=20
> RRs listen to
> for retransmission and RAMS.
>=20
> Roni E: This implies that the multicast source defines the=20
> port that all
> clients MUST use to receive the unicast information from the=20
> Burst source. =20

Yes, that is what it implies (One nit is that I would call it unicast =
RTP packets rather than information since we may use another port for =
RTCP packets).
=20
>         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> Acq. Support)
>         c=3DIN IP4 192.0.2.1
>=20
> This is where the retransmission packets come from, same as=20
> the feedback
> target (in this example).
>=20
> Roni E: If this is the address of the feedback target it=20
> belongs to the
> above m-line (m=3Dvideo 41002 RTP/AVPF 99) that according to=20
> you gives the
> port on the RR and not on the RS. The problem is that=20

This line actually specifies the sender for the unicast retransmission =
session. In our case, it just happens to be the feedback target for the =
primary SSM session, too. I recall some discussion about using a =
retransmission source which is different than the feedback target for =
the SSM. In that case, the address would be different, I guess.

> according to SDP the
> combination of the c=3D and port on the m-line provides the=20
> transport address
> to where a stream will be sent so according to SDP this=20

The port on the m line is the port where the retranmission packets are =
sent. However, c line denotes the source for these packets. RFC 4566 =
page 13 says that "... If the session is not multicast, then the =
connection address contains the unicast IP address of the expected data =
source or data relay or data sink as determined by additional attribute =
fields..."

Am I missing something?

> m-line specify an
> address and port on the Feedback Target and not on the RR and=20
> only to this
> address RTP payload type 99 will be sent. I am positive that=20
> this is not
> what you want.

There can be other payload types we might need to send on this =
retransmission session in the future. In that case, we will add them.
=20
>         a=3Drtpmap:99 rtx/90000
>         a=3Drtcp:41003
>=20
> This is where the RTCP packets for the retransmission session go. For
> example, RAMS-I goes to this port on RRs. RAMS-T goes to this=20
> port on RS.
>=20
> Roni E: So now you assume that the RR must use the same port=20
> as the FT for
> this communication. I could not find any place that one=20
> m-line gives the

Not the m line, but the "a=3Drtcp" line. Is this wrong?

> address for both sides. This RTCP port is on 192.0.2.1 so it=20
> will be used
> for RAMS-R but not for RAMS-I.

RAMS-R should go to the feedback target address and port for the primary =
SSM session since it is a feedback message for the SSM session. But, =
RAMS-T is a feedback message for the retransmission session, thus, it =
should go to the RTCP port of the retransmission session (41003). I hope =
this also clarifies your question in the next email. If you think I got =
the ports wrong, let me know.

-acbegen
=20
>         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
>         a=3Dmid:4
>=20
>=20
> -acbegen
> =20
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > Sent: Saturday, April 25, 2009 1:11 PM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Ali,
> > Can you please write three addresses from this strange SDP.
> >=20
> > 1. The address and port of multicast
> >=20
> > 2. The address and port of the RS where the RTCP FB should go to
> >=20
> > 3. The address and port on the RR where the unicast stream=20
> > should be sent to
> >=20
> > Roni
> >=20
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Saturday, April 25, 2009 11:02 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > > Ali,
> > > The example SDP is syntactically correct but it does not=20
> > say that the
> > > rtp/rtcp mux will be used and it does not give the=20
> > > information to where the
> > > unicast stream will be sent when the RR sends a RAMS-R.=20
> >=20
> > To my knowledge, the first line in the following SDP tells=20
> > the RRs on which
> > port they will receive the retransmission/burst packets.
> >=20
> >         m=3Dvideo 41002 RTP/AVPF 99
> >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > Acq. Support)
> >         c=3DIN IP4 192.0.2.1
> >         a=3Drtpmap:99 rtx/90000
> >         a=3Drtcp:41003
> >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> >         a=3Dmid:4
> >=20
> > There is a typo, you are right. That "a=3Drecvonly" line should=20
> > only exist in
> > the SDP sent to RRs. In the SDP sent to RS, we should rather have
> > "a=3Dsendonly". I will make this correction in the next version.
> >=20
> > The feedback target for the SSM session and the RTCP port for the
> > retransmission session are also defined in the SDP.
> >=20
> > Hope this clarifies.
> >=20
> > BR,
> > -acbegen
> > =20
> > >=20
> > > I am not sure why the unicast stream m-line has a port=20
> > number with an
> > > attribute of recvonly. What is the use case for that. The=20
> > > only information
> > > that the RR will need is the RTCP port on the RS to where to=20
> > > send the RAMS-R
> > > message.
> > > Roni
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > Sent: Friday, April 24, 2009 10:37 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions
> > > (RAMS)" available
> > >=20
> > > This is a part of an example SDP sent to RS and RR in a SAP=20
> > > announcement,
> > > for example. So, the SDP describes what both parties should=20
> > > do (RR cannot
> > > say that he wants to receive this multicast on its favorite=20
> > port). The
> > > individual SDPs sent to RR or RS may include other portions=20
> > > of descriptions
> > > that will contain specific information.=20
> > >=20
> > > -acbegen
> > >=20
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions (RAMS)" available
> > > >=20
> > > > Ali,
> > > > I think you get it wrong, this SDP is from the RS and not the=20
> > > > RR so the RS
> > > > cannot specify to which address it will send it can only=20
> > > > specify to which
> > > > address it can receive RTP stream. In this SDP the relevant=20
> > > > information is
> > > > that the request for retransmission will be sent by the RR to=20
> > > > port 41003
> > > > Roni
> > > >=20
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > >=20
> > > > Oh, I see. The burst goes to the port that we would=20
> > > normally send the
> > > > retransmissions. For example, consider the SDP from the draft:
> > > >=20
> > > >         m=3Dvideo 41000 RTP/AVPF 98
> > > >         i=3DPrimary Multicast Stream #2
> > > >         c=3DIN IP4 224.1.1.2/255
> > > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > >         a=3Drtpmap:98 MP2T/90000
> > > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > > >         a=3Drtcp-fb:98 nack
> > > >         a=3Drtcp-fb:98 nack ssli
> > > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > > >         a=3Dmid:3
> > > >         m=3Dvideo 41002 RTP/AVPF 99
> > > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > > Acq. Support)
> > > >         c=3DIN IP4 192.0.2.1
> > > >         a=3Drecvonly
> > > >         a=3Drtpmap:99 rtx/90000
> > > >         a=3Drtcp:41003
> > > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > > >         a=3Dmid:4
> > > >=20
> > > > In this case, the burst will go to port 41002 on the RR.=20
> > The RAMS-I
> > > > message(s), which is an RTCP feedback message, will go to=20
> > > port 41003.
> > > >=20
> > > > HTH,
> > > > -acbegen
> > > >=20
> > > > > -----Original Message-----
> > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions (RAMS)" available
> > > > >=20
> > > > > Ali,
> > > > > I will try to explain in simple way
> > > > >=20
> > > > > When the RS receives the RAMS-R it need to start sending an=20
> > > > > RTP stream to
> > > > > the RR.=20
> > > > > In order to send a unicast packet, the RS needs to know a=20
> > > > > transport address
> > > > > on the RR to where the RTP stream will be sent. The current=20
> > > > > draft does not
> > > > > say how the RS knows this address. There is no SDP from RR to=20
> > > > > RS like you
> > > > > mention in your response.
> > > > > This is why I say that the current draft does not specify a=20
> > > > > solution that
> > > > > can be implemented as a working solution
> > > > > Roni
> > > > >=20
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > >=20
> > > > > Hi Roni,=20
> > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > > Behalf Of Roni Even
> > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > To: avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > >=20
> > > > > > Hi,
> > > > > >=20
> > > > > > I think that the current draft does not give a=20
> description of=20
> > > > > > a system that works since there is no text=20
> explaining how the=20
> > > > >=20
> > > > > What do you mean by "a system that works"?=20
> > > > >=20
> > > > > > RS knows the unicast transport address on the RR to=20
> where to=20
> > > > > > send the stream.=20
> > > > >=20
> > > > > Once RS receives the request packet from an RR, RS knows its=20
> > > > > address. Ports
> > > > > are defined in the SDP. If you are asking about "NAT" issues,=20
> > > > > we have a
> > > > > section for it, and we plan to complete it as we move=20
> > > > > forward. It is not as
> > > > > critical as the other parts for now.
> > > > > =20
> > > > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > > > otherwise the RAMS-R should have an optional parameter that=20
> > > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > >=20
> > > > > IMO, we cannot mandate muxing as it is not required to make=20
> > > > > RAMS work. If
> > > > > muxing is supported, the SDP should reflect it.
> > > > >=20
> > > > > BR,
> > > > > -acbegen
> > > > > =20
> > > > > > Thanks
> > > > > >=20
> > > > > > Roni Even
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > To: avt@ietf.org
> > > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > > > > > RTP Sessions (RAMS)" available
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > There is a new draft of the "Rapid Acquisition of Multicast=20
> > > > > > RTP Sessions (RAMS)" draft available at=20
> > > > > >=20
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > >=20
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > We have incorporated the changes from the technical=20
> breakout=20
> > > > > > session in San Francisco. The major changes in this version=20
> > > > > > of the draft include=20
> > > > > >=20
> > > > > > 1- Changing the document title to avoid confusion=20
> with other=20
> > > > > > ongoing "synchronization" drafts
> > > > > >=20
> > > > > > 2- changing the message names to reflect the title change
> > > > > >=20
> > > > > > 3- clarification of the RTCP message semantics, including=20
> > > > > > changes to the "Request" and "Inform" messages
> > > > > >=20
> > > > > > 4- additional description/motivation for the=20
> various message=20
> > > > > > flows has been added
> > > > > >=20
> > > > > > 5- RTP/RTCP muxing has been added
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > We hope to make this a Working Group item, and will change=20
> > > > > > the name of the draft to avoid conflicts with other=20
> > > > > > "synchronization" drafts at that time.
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > Bill VerSteeg
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > >=20
> > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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=20

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Saturday, April 25, 2009 2:58 PM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Ali,
> I looked at RFC 4570 and I assume that port 41001 is the port for the
> unicast RTCP reports from the receivers and according to

Yes, indeed.
=20
> section 3.2.1 of
> that RFC you also should have a RTCP-unicast specification.=20
> This is for the
> multicast receiver reports.

Correct. We do have that line in our draft at the top right after the =
grouping lines:

	a=3Drtcp-unicast:rsi

BR,
-acbegen
=20
> Roni
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Saturday, April 25, 2009 11:30 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> Here is the SDP.
>=20
>         m=3Dvideo 41000 RTP/AVPF 98
>         i=3DPrimary Multicast Stream #2
>         c=3DIN IP4 224.1.1.2/255
>         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
>=20
> Source (192.0.2.2) sends the rtp packets to the multicast=20
> group (224.1.1.2)
> with a dest port 41000.
>=20
>         a=3Drtpmap:98 MP2T/90000
>         a=3Drtcp:41001 IN IP4 192.0.2.1
>=20
> The feedback target (RS) address for this SSM session is=20
> 192.0.2.1 and its
> port is 41001. This is the address/port where RR sends the RAMS-R, or
> receiver reports for the SSM session.
>=20
>         a=3Drtcp-fb:98 nack
>         a=3Drtcp-fb:98 nack ssli
>         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
>         a=3Dmid:3
>         m=3Dvideo 41002 RTP/AVPF 99
>=20
> The retransmission packets go to port 41002. This is the port=20
> RRs listen to
> for retransmission and RAMS.
>=20
>         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> Acq. Support)
>         c=3DIN IP4 192.0.2.1
>=20
> This is where the retransmission packets come from, same as=20
> the feedback
> target (in this example).
>=20
>         a=3Drtpmap:99 rtx/90000
>         a=3Drtcp:41003
>=20
> This is where the RTCP packets for the retransmission session go. For
> example, RAMS-I goes to this port on RRs. RAMS-T goes to this=20
> port on RS.
>=20
>         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
>         a=3Dmid:4
>=20
>=20
> -acbegen
> =20
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > Sent: Saturday, April 25, 2009 1:11 PM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Ali,
> > Can you please write three addresses from this strange SDP.
> >=20
> > 1. The address and port of multicast
> >=20
> > 2. The address and port of the RS where the RTCP FB should go to
> >=20
> > 3. The address and port on the RR where the unicast stream=20
> > should be sent to
> >=20
> > Roni
> >=20
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Saturday, April 25, 2009 11:02 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > > Ali,
> > > The example SDP is syntactically correct but it does not=20
> > say that the
> > > rtp/rtcp mux will be used and it does not give the=20
> > > information to where the
> > > unicast stream will be sent when the RR sends a RAMS-R.=20
> >=20
> > To my knowledge, the first line in the following SDP tells=20
> > the RRs on which
> > port they will receive the retransmission/burst packets.
> >=20
> >         m=3Dvideo 41002 RTP/AVPF 99
> >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > Acq. Support)
> >         c=3DIN IP4 192.0.2.1
> >         a=3Drtpmap:99 rtx/90000
> >         a=3Drtcp:41003
> >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> >         a=3Dmid:4
> >=20
> > There is a typo, you are right. That "a=3Drecvonly" line should=20
> > only exist in
> > the SDP sent to RRs. In the SDP sent to RS, we should rather have
> > "a=3Dsendonly". I will make this correction in the next version.
> >=20
> > The feedback target for the SSM session and the RTCP port for the
> > retransmission session are also defined in the SDP.
> >=20
> > Hope this clarifies.
> >=20
> > BR,
> > -acbegen
> > =20
> > >=20
> > > I am not sure why the unicast stream m-line has a port=20
> > number with an
> > > attribute of recvonly. What is the use case for that. The=20
> > > only information
> > > that the RR will need is the RTCP port on the RS to where to=20
> > > send the RAMS-R
> > > message.
> > > Roni
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > Sent: Friday, April 24, 2009 10:37 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions
> > > (RAMS)" available
> > >=20
> > > This is a part of an example SDP sent to RS and RR in a SAP=20
> > > announcement,
> > > for example. So, the SDP describes what both parties should=20
> > > do (RR cannot
> > > say that he wants to receive this multicast on its favorite=20
> > port). The
> > > individual SDPs sent to RR or RS may include other portions=20
> > > of descriptions
> > > that will contain specific information.=20
> > >=20
> > > -acbegen
> > >=20
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions (RAMS)" available
> > > >=20
> > > > Ali,
> > > > I think you get it wrong, this SDP is from the RS and not the=20
> > > > RR so the RS
> > > > cannot specify to which address it will send it can only=20
> > > > specify to which
> > > > address it can receive RTP stream. In this SDP the relevant=20
> > > > information is
> > > > that the request for retransmission will be sent by the RR to=20
> > > > port 41003
> > > > Roni
> > > >=20
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > >=20
> > > > Oh, I see. The burst goes to the port that we would=20
> > > normally send the
> > > > retransmissions. For example, consider the SDP from the draft:
> > > >=20
> > > >         m=3Dvideo 41000 RTP/AVPF 98
> > > >         i=3DPrimary Multicast Stream #2
> > > >         c=3DIN IP4 224.1.1.2/255
> > > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > >         a=3Drtpmap:98 MP2T/90000
> > > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > > >         a=3Drtcp-fb:98 nack
> > > >         a=3Drtcp-fb:98 nack ssli
> > > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > > >         a=3Dmid:3
> > > >         m=3Dvideo 41002 RTP/AVPF 99
> > > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > > Acq. Support)
> > > >         c=3DIN IP4 192.0.2.1
> > > >         a=3Drecvonly
> > > >         a=3Drtpmap:99 rtx/90000
> > > >         a=3Drtcp:41003
> > > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > > >         a=3Dmid:4
> > > >=20
> > > > In this case, the burst will go to port 41002 on the RR.=20
> > The RAMS-I
> > > > message(s), which is an RTCP feedback message, will go to=20
> > > port 41003.
> > > >=20
> > > > HTH,
> > > > -acbegen
> > > >=20
> > > > > -----Original Message-----
> > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions (RAMS)" available
> > > > >=20
> > > > > Ali,
> > > > > I will try to explain in simple way
> > > > >=20
> > > > > When the RS receives the RAMS-R it need to start sending an=20
> > > > > RTP stream to
> > > > > the RR.=20
> > > > > In order to send a unicast packet, the RS needs to know a=20
> > > > > transport address
> > > > > on the RR to where the RTP stream will be sent. The current=20
> > > > > draft does not
> > > > > say how the RS knows this address. There is no SDP from RR to=20
> > > > > RS like you
> > > > > mention in your response.
> > > > > This is why I say that the current draft does not specify a=20
> > > > > solution that
> > > > > can be implemented as a working solution
> > > > > Roni
> > > > >=20
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > >=20
> > > > > Hi Roni,=20
> > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > > Behalf Of Roni Even
> > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > To: avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > >=20
> > > > > > Hi,
> > > > > >=20
> > > > > > I think that the current draft does not give a=20
> description of=20
> > > > > > a system that works since there is no text=20
> explaining how the=20
> > > > >=20
> > > > > What do you mean by "a system that works"?=20
> > > > >=20
> > > > > > RS knows the unicast transport address on the RR to=20
> where to=20
> > > > > > send the stream.=20
> > > > >=20
> > > > > Once RS receives the request packet from an RR, RS knows its=20
> > > > > address. Ports
> > > > > are defined in the SDP. If you are asking about "NAT" issues,=20
> > > > > we have a
> > > > > section for it, and we plan to complete it as we move=20
> > > > > forward. It is not as
> > > > > critical as the other parts for now.
> > > > > =20
> > > > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > > > otherwise the RAMS-R should have an optional parameter that=20
> > > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > >=20
> > > > > IMO, we cannot mandate muxing as it is not required to make=20
> > > > > RAMS work. If
> > > > > muxing is supported, the SDP should reflect it.
> > > > >=20
> > > > > BR,
> > > > > -acbegen
> > > > > =20
> > > > > > Thanks
> > > > > >=20
> > > > > > Roni Even
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > To: avt@ietf.org
> > > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast=20
> > > > > > RTP Sessions (RAMS)" available
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > There is a new draft of the "Rapid Acquisition of Multicast=20
> > > > > > RTP Sessions (RAMS)" draft available at=20
> > > > > >=20
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > >=20
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > We have incorporated the changes from the technical=20
> breakout=20
> > > > > > session in San Francisco. The major changes in this version=20
> > > > > > of the draft include=20
> > > > > >=20
> > > > > > 1- Changing the document title to avoid confusion=20
> with other=20
> > > > > > ongoing "synchronization" drafts
> > > > > >=20
> > > > > > 2- changing the message names to reflect the title change
> > > > > >=20
> > > > > > 3- clarification of the RTCP message semantics, including=20
> > > > > > changes to the "Request" and "Inform" messages
> > > > > >=20
> > > > > > 4- additional description/motivation for the=20
> various message=20
> > > > > > flows has been added
> > > > > >=20
> > > > > > 5- RTP/RTCP muxing has been added
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > We hope to make this a Working Group item, and will change=20
> > > > > > the name of the draft to avoid conflicts with other=20
> > > > > > "synchronization" drafts at that time.
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > Bill VerSteeg
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > > =20
> > > > > >=20
> > > > > >=20
> > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

From ron.even.tlv@gmail.com  Sun Apr 26 01:14:27 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com> <49f36ee2.0437560a.11df.1eef@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com> <49f37fde.09a1660a.2b46.ffffb694@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8408@xmb-sjc-215.amer.cisco.com>
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Date: Sun, 26 Apr 2009 11:14:07 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
I looked at RTSP
http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-20#appendix-D 



D.1.2.  Media Streams

   The "m=" field is used to enumerate the streams.  It is expected that
   all the specified streams will be rendered with appropriate
   synchronization.  If the session is over multicast, the port number
   indicated SHOULD be used for reception.  The client MAY try to
   override the destination port, through the Transport header.  The
   servers MAY allow this, the response will indicate if allowed or not.
   If the session is unicast, the port numbers are the ones RECOMMENDED
   by the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.
   The client MAY ignore this recommendation.  If the server has no
   preference, it SHOULD set the port number value to zero.


Note that the client has the choice on which port to receive the unicast
stream, this is why there should be such a parameter in RAMS-R


Also

" D.1.8.  Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream.  For on-demand unicast streams and some multicast streams,
   the destination address MAY be specified by the client via the SETUP
   request, thus overriding any specified address.  To identify streams
   without a fixed destination address, where the client is required to
   specify a destination address, the "c=" field SHOULD be set to a null
   value.  For addresses of type "IP4", this value MUST be "0.0.0.0",
   and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" (can also be
   written as "::"), i.e. the unspecified address according to RFC 4291
   [RFC4291]."


As for the c-line in the unicast stream the whole section you quoted

" If the session is multicast, the connection address will be an IP
      multicast group address.  If the session is not multicast, then
      the connection address contains the unicast IP address of the
      expected data source or data relay or data sink as determined by
      additional attribute fields.  It is not expected that unicast
      addresses will be given in a session description that is
      communicated by a multicast announcement, though this is not
      prohibited."

Now the expectation here is for the address of the stream described by the
port in the m-line, I do not see any other attribute. Also look at the above
text from RTSP. The RS does not know the IP address on the RR to where to
send the RTP unicast repair stream. If you assume it is to the IP address
from where the RTCP RAMS-R came from this is not mandated that the client
will have such implementation. Furthermore the address that the RS will see
in the received RTCP RAMS-R packet may have been change by a NAT or SBC.


Roni



-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Sunday, April 26, 2009 1:23 AM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Inline.

>         m=video 41000 RTP/AVPF 98
>         i=Primary Multicast Stream #2
>         c=IN IP4 224.1.1.2/255
>         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> 
> Source (192.0.2.2) sends the rtp packets to the multicast 
> group (224.1.1.2)
> with a dest port 41000.
> 
>         a=rtpmap:98 MP2T/90000
>         a=rtcp:41001 IN IP4 192.0.2.1
> 
> The feedback target (RS) address for this SSM session is 
> 192.0.2.1 and its
> port is 41001. This is the address/port where RR sends the RAMS-R, or
> receiver reports for the SSM session.
> 
>         a=rtcp-fb:98 nack
>         a=rtcp-fb:98 nack ssli
>         a=ssrc:123321 cname:iptv-ch32@rams.example.com
>         a=mid:3
>         m=video 41002 RTP/AVPF 99
> 
> The retransmission packets go to port 41002. This is the port 
> RRs listen to
> for retransmission and RAMS.
> 
> Roni E: This implies that the multicast source defines the 
> port that all
> clients MUST use to receive the unicast information from the 
> Burst source.  

Yes, that is what it implies (One nit is that I would call it unicast RTP
packets rather than information since we may use another port for RTCP
packets).
 
>         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> Acq. Support)
>         c=IN IP4 192.0.2.1
> 
> This is where the retransmission packets come from, same as 
> the feedback
> target (in this example).
> 
> Roni E: If this is the address of the feedback target it 
> belongs to the
> above m-line (m=video 41002 RTP/AVPF 99) that according to 
> you gives the
> port on the RR and not on the RS. The problem is that 

This line actually specifies the sender for the unicast retransmission
session. In our case, it just happens to be the feedback target for the
primary SSM session, too. I recall some discussion about using a
retransmission source which is different than the feedback target for the
SSM. In that case, the address would be different, I guess.

> according to SDP the
> combination of the c= and port on the m-line provides the 
> transport address
> to where a stream will be sent so according to SDP this 

The port on the m line is the port where the retranmission packets are sent.
However, c line denotes the source for these packets. RFC 4566 page 13 says
that "... If the session is not multicast, then the connection address
contains the unicast IP address of the expected data source or data relay or
data sink as determined by additional attribute fields..."

Am I missing something?

> m-line specify an
> address and port on the Feedback Target and not on the RR and 
> only to this
> address RTP payload type 99 will be sent. I am positive that 
> this is not
> what you want.

There can be other payload types we might need to send on this
retransmission session in the future. In that case, we will add them.
 
>         a=rtpmap:99 rtx/90000
>         a=rtcp:41003
> 
> This is where the RTCP packets for the retransmission session go. For
> example, RAMS-I goes to this port on RRs. RAMS-T goes to this 
> port on RS.
> 
> Roni E: So now you assume that the RR must use the same port 
> as the FT for
> this communication. I could not find any place that one 
> m-line gives the

Not the m line, but the "a=rtcp" line. Is this wrong?

> address for both sides. This RTCP port is on 192.0.2.1 so it 
> will be used
> for RAMS-R but not for RAMS-I.

RAMS-R should go to the feedback target address and port for the primary SSM
session since it is a feedback message for the SSM session. But, RAMS-T is a
feedback message for the retransmission session, thus, it should go to the
RTCP port of the retransmission session (41003). I hope this also clarifies
your question in the next email. If you think I got the ports wrong, let me
know.

-acbegen
 
>         a=fmtp:99 apt=98; rtx-time=5000
>         a=mid:4
> 
> 
> -acbegen
>  
> 
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > Sent: Saturday, April 25, 2009 1:11 PM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions (RAMS)" available
> > 
> > Ali,
> > Can you please write three addresses from this strange SDP.
> > 
> > 1. The address and port of multicast
> > 
> > 2. The address and port of the RS where the RTCP FB should go to
> > 
> > 3. The address and port on the RR where the unicast stream 
> > should be sent to
> > 
> > Roni
> > 
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > Sent: Saturday, April 25, 2009 11:02 PM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > Multicast RTPSessions
> > (RAMS)" available
> > 
> > > Ali,
> > > The example SDP is syntactically correct but it does not 
> > say that the
> > > rtp/rtcp mux will be used and it does not give the 
> > > information to where the
> > > unicast stream will be sent when the RR sends a RAMS-R. 
> > 
> > To my knowledge, the first line in the following SDP tells 
> > the RRs on which
> > port they will receive the retransmission/burst packets.
> > 
> >         m=video 41002 RTP/AVPF 99
> >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > Acq. Support)
> >         c=IN IP4 192.0.2.1
> >         a=rtpmap:99 rtx/90000
> >         a=rtcp:41003
> >         a=fmtp:99 apt=98; rtx-time=5000
> >         a=mid:4
> > 
> > There is a typo, you are right. That "a=recvonly" line should 
> > only exist in
> > the SDP sent to RRs. In the SDP sent to RS, we should rather have
> > "a=sendonly". I will make this correction in the next version.
> > 
> > The feedback target for the SSM session and the RTCP port for the
> > retransmission session are also defined in the SDP.
> > 
> > Hope this clarifies.
> > 
> > BR,
> > -acbegen
> >  
> > > 
> > > I am not sure why the unicast stream m-line has a port 
> > number with an
> > > attribute of recvonly. What is the use case for that. The 
> > > only information
> > > that the RR will need is the RTCP port on the RS to where to 
> > > send the RAMS-R
> > > message.
> > > Roni
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Friday, April 24, 2009 10:37 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > This is a part of an example SDP sent to RS and RR in a SAP 
> > > announcement,
> > > for example. So, the SDP describes what both parties should 
> > > do (RR cannot
> > > say that he wants to receive this multicast on its favorite 
> > port). The
> > > individual SDPs sent to RR or RS may include other portions 
> > > of descriptions
> > > that will contain specific information. 
> > > 
> > > -acbegen
> > > 
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions (RAMS)" available
> > > > 
> > > > Ali,
> > > > I think you get it wrong, this SDP is from the RS and not the 
> > > > RR so the RS
> > > > cannot specify to which address it will send it can only 
> > > > specify to which
> > > > address it can receive RTP stream. In this SDP the relevant 
> > > > information is
> > > > that the request for retransmission will be sent by the RR to 
> > > > port 41003
> > > > Roni
> > > > 
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > > 
> > > > Oh, I see. The burst goes to the port that we would 
> > > normally send the
> > > > retransmissions. For example, consider the SDP from the draft:
> > > > 
> > > >         m=video 41000 RTP/AVPF 98
> > > >         i=Primary Multicast Stream #2
> > > >         c=IN IP4 224.1.1.2/255
> > > >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > >         a=rtpmap:98 MP2T/90000
> > > >         a=rtcp:41001 IN IP4 192.0.2.1
> > > >         a=rtcp-fb:98 nack
> > > >         a=rtcp-fb:98 nack ssli
> > > >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> > > >         a=mid:3
> > > >         m=video 41002 RTP/AVPF 99
> > > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > > Acq. Support)
> > > >         c=IN IP4 192.0.2.1
> > > >         a=recvonly
> > > >         a=rtpmap:99 rtx/90000
> > > >         a=rtcp:41003
> > > >         a=fmtp:99 apt=98; rtx-time=5000
> > > >         a=mid:4
> > > > 
> > > > In this case, the burst will go to port 41002 on the RR. 
> > The RAMS-I
> > > > message(s), which is an RTCP feedback message, will go to 
> > > port 41003.
> > > > 
> > > > HTH,
> > > > -acbegen
> > > > 
> > > > > -----Original Message-----
> > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions (RAMS)" available
> > > > > 
> > > > > Ali,
> > > > > I will try to explain in simple way
> > > > > 
> > > > > When the RS receives the RAMS-R it need to start sending an 
> > > > > RTP stream to
> > > > > the RR. 
> > > > > In order to send a unicast packet, the RS needs to know a 
> > > > > transport address
> > > > > on the RR to where the RTP stream will be sent. The current 
> > > > > draft does not
> > > > > say how the RS knows this address. There is no SDP from RR to 
> > > > > RS like you
> > > > > mention in your response.
> > > > > This is why I say that the current draft does not specify a 
> > > > > solution that
> > > > > can be implemented as a working solution
> > > > > Roni
> > > > > 
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > > 
> > > > > Hi Roni, 
> > > > > 
> > > > > > -----Original Message-----
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > > Behalf Of Roni Even
> > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > To: avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > 
> > > > > > Hi,
> > > > > > 
> > > > > > I think that the current draft does not give a 
> description of 
> > > > > > a system that works since there is no text 
> explaining how the 
> > > > > 
> > > > > What do you mean by "a system that works"? 
> > > > > 
> > > > > > RS knows the unicast transport address on the RR to 
> where to 
> > > > > > send the stream. 
> > > > > 
> > > > > Once RS receives the request packet from an RR, RS knows its 
> > > > > address. Ports
> > > > > are defined in the SDP. If you are asking about "NAT" issues, 
> > > > > we have a
> > > > > section for it, and we plan to complete it as we move 
> > > > > forward. It is not as
> > > > > critical as the other parts for now.
> > > > >  
> > > > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > > > otherwise the RAMS-R should have an optional parameter that 
> > > > > > supplies this information and a flag for using RTP/RTCP mux.
> > > > > 
> > > > > IMO, we cannot mandate muxing as it is not required to make 
> > > > > RAMS work. If
> > > > > muxing is supported, the SDP should reflect it.
> > > > > 
> > > > > BR,
> > > > > -acbegen
> > > > >  
> > > > > > Thanks
> > > > > > 
> > > > > > Roni Even
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > To: avt@ietf.org
> > > > > > Subject: [AVT] New draft of "Rapid Acquisition of Multicast 
> > > > > > RTP Sessions (RAMS)" available
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > There is a new draft of the "Rapid Acquisition of Multicast 
> > > > > > RTP Sessions (RAMS)" draft available at 
> > > > > > 
> http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > ynchronization-for-rtp-03.txt <http://www.ietf.org/internet->
> > > > > 
> drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > We have incorporated the changes from the technical 
> breakout 
> > > > > > session in San Francisco. The major changes in this version 
> > > > > > of the draft include 
> > > > > > 
> > > > > > 1- Changing the document title to avoid confusion 
> with other 
> > > > > > ongoing "synchronization" drafts
> > > > > > 
> > > > > > 2- changing the message names to reflect the title change
> > > > > > 
> > > > > > 3- clarification of the RTCP message semantics, including 
> > > > > > changes to the "Request" and "Inform" messages
> > > > > > 
> > > > > > 4- additional description/motivation for the 
> various message 
> > > > > > flows has been added
> > > > > > 
> > > > > > 5- RTP/RTCP muxing has been added
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > We hope to make this a Working Group item, and will change 
> > > > > > the name of the draft to avoid conflicts with other 
> > > > > > "synchronization" drafts at that time.
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > Bill VerSteeg
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > >  
> > > > > > 
> > > > > > 
> > > > > 
> > > > > 
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Hi Roni,

Thanks for the detailed answer. Admittedly, I don't know much about RTSP =
beyond its basics. However, I doubt how much it applies in our case. =
"Unicast" in RTSP context mostly refers to video-on-demand kind of =
applications where the primary distribution is over unicast. In =
contrast, our primary distribution is over multicast, and we use unicast =
as a repair stream. Thus, I still believe the SDP we have should work =
fine. RS gathers the IP address of the RR when it receives the RAMS =
request. The samething happens when RR sends a NACK for requesting =
retransmissions.

That is not to say that this SDP is the only SDP that will work in this =
system. There may be others. But, I still don't agree with you when you =
say that this SDP won't work. My expectation is that we will include =
other SDP examples as we move forward.

On your last point, NAT is a different issue and there has to be a way =
to deal with it. Muxing sounds reasonable, and we may or may not mandate =
it, I am not sure at this point. An alternative could be that RR opens =
ports on its NAT device and includes them in the request message it =
sends. These could be in RAMS-R TLV extensions, or could be a separate =
RTCP packet solely for this purpose within the same RTCP compound packet =
that carries the RAMS-R. I believe Bill mentioned these briefly in his =
earlier email. We are open to suggestions at this point and the NAT =
section will cover these issues once we have a WG item.

-acbegen

=20

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Sunday, April 26, 2009 1:14 AM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions (RAMS)" available
>=20
> Ali,
> I looked at RTSP
> http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-20#appendix-D=20
>=20
>=20
>=20
> D.1.2.  Media Streams
>=20
>    The "m=3D" field is used to enumerate the streams.  It is=20
> expected that
>    all the specified streams will be rendered with appropriate
>    synchronization.  If the session is over multicast, the port number
>    indicated SHOULD be used for reception.  The client MAY try to
>    override the destination port, through the Transport header.  The
>    servers MAY allow this, the response will indicate if=20
> allowed or not.
>    If the session is unicast, the port numbers are the ones=20
> RECOMMENDED
>    by the server to the client, about which receiver ports to use; the
>    client MUST still include its receiver ports in its SETUP request.
>    The client MAY ignore this recommendation.  If the server has no
>    preference, it SHOULD set the port number value to zero.
>=20
>=20
> Note that the client has the choice on which port to receive=20
> the unicast
> stream, this is why there should be such a parameter in RAMS-R
>=20
>=20
> Also
>=20
> " D.1.8.  Connection Information
>=20
>    In SDP, the "c=3D" field contains the destination address=20
> for the media
>    stream.  For on-demand unicast streams and some multicast streams,
>    the destination address MAY be specified by the client via=20
> the SETUP
>    request, thus overriding any specified address.  To=20
> identify streams
>    without a fixed destination address, where the client is=20
> required to
>    specify a destination address, the "c=3D" field SHOULD be=20
> set to a null
>    value.  For addresses of type "IP4", this value MUST be "0.0.0.0",
>    and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0"=20
> (can also be
>    written as "::"), i.e. the unspecified address according=20
> to RFC 4291
>    [RFC4291]."
>=20
>=20
> As for the c-line in the unicast stream the whole section you quoted
>=20
> " If the session is multicast, the connection address will be an IP
>       multicast group address.  If the session is not multicast, then
>       the connection address contains the unicast IP address of the
>       expected data source or data relay or data sink as determined by
>       additional attribute fields.  It is not expected that unicast
>       addresses will be given in a session description that is
>       communicated by a multicast announcement, though this is not
>       prohibited."
>=20
> Now the expectation here is for the address of the stream=20
> described by the
> port in the m-line, I do not see any other attribute. Also=20
> look at the above
> text from RTSP. The RS does not know the IP address on the RR=20
> to where to
> send the RTP unicast repair stream. If you assume it is to=20
> the IP address
> from where the RTCP RAMS-R came from this is not mandated=20
> that the client
> will have such implementation. Furthermore the address that=20
> the RS will see
> in the received RTCP RAMS-R packet may have been change by a=20
> NAT or SBC.
>=20
>=20
> Roni
>=20
>=20
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Sunday, April 26, 2009 1:23 AM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> Inline.
>=20
> >         m=3Dvideo 41000 RTP/AVPF 98
> >         i=3DPrimary Multicast Stream #2
> >         c=3DIN IP4 224.1.1.2/255
> >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> >=20
> > Source (192.0.2.2) sends the rtp packets to the multicast=20
> > group (224.1.1.2)
> > with a dest port 41000.
> >=20
> >         a=3Drtpmap:98 MP2T/90000
> >         a=3Drtcp:41001 IN IP4 192.0.2.1
> >=20
> > The feedback target (RS) address for this SSM session is=20
> > 192.0.2.1 and its
> > port is 41001. This is the address/port where RR sends the=20
> RAMS-R, or
> > receiver reports for the SSM session.
> >=20
> >         a=3Drtcp-fb:98 nack
> >         a=3Drtcp-fb:98 nack ssli
> >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> >         a=3Dmid:3
> >         m=3Dvideo 41002 RTP/AVPF 99
> >=20
> > The retransmission packets go to port 41002. This is the port=20
> > RRs listen to
> > for retransmission and RAMS.
> >=20
> > Roni E: This implies that the multicast source defines the=20
> > port that all
> > clients MUST use to receive the unicast information from the=20
> > Burst source. =20
>=20
> Yes, that is what it implies (One nit is that I would call it=20
> unicast RTP
> packets rather than information since we may use another port for RTCP
> packets).
> =20
> >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > Acq. Support)
> >         c=3DIN IP4 192.0.2.1
> >=20
> > This is where the retransmission packets come from, same as=20
> > the feedback
> > target (in this example).
> >=20
> > Roni E: If this is the address of the feedback target it=20
> > belongs to the
> > above m-line (m=3Dvideo 41002 RTP/AVPF 99) that according to=20
> > you gives the
> > port on the RR and not on the RS. The problem is that=20
>=20
> This line actually specifies the sender for the unicast retransmission
> session. In our case, it just happens to be the feedback=20
> target for the
> primary SSM session, too. I recall some discussion about using a
> retransmission source which is different than the feedback=20
> target for the
> SSM. In that case, the address would be different, I guess.
>=20
> > according to SDP the
> > combination of the c=3D and port on the m-line provides the=20
> > transport address
> > to where a stream will be sent so according to SDP this=20
>=20
> The port on the m line is the port where the retranmission=20
> packets are sent.
> However, c line denotes the source for these packets. RFC=20
> 4566 page 13 says
> that "... If the session is not multicast, then the connection address
> contains the unicast IP address of the expected data source=20
> or data relay or
> data sink as determined by additional attribute fields..."
>=20
> Am I missing something?
>=20
> > m-line specify an
> > address and port on the Feedback Target and not on the RR and=20
> > only to this
> > address RTP payload type 99 will be sent. I am positive that=20
> > this is not
> > what you want.
>=20
> There can be other payload types we might need to send on this
> retransmission session in the future. In that case, we will add them.
> =20
> >         a=3Drtpmap:99 rtx/90000
> >         a=3Drtcp:41003
> >=20
> > This is where the RTCP packets for the retransmission=20
> session go. For
> > example, RAMS-I goes to this port on RRs. RAMS-T goes to this=20
> > port on RS.
> >=20
> > Roni E: So now you assume that the RR must use the same port=20
> > as the FT for
> > this communication. I could not find any place that one=20
> > m-line gives the
>=20
> Not the m line, but the "a=3Drtcp" line. Is this wrong?
>=20
> > address for both sides. This RTCP port is on 192.0.2.1 so it=20
> > will be used
> > for RAMS-R but not for RAMS-I.
>=20
> RAMS-R should go to the feedback target address and port for=20
> the primary SSM
> session since it is a feedback message for the SSM session.=20
> But, RAMS-T is a
> feedback message for the retransmission session, thus, it=20
> should go to the
> RTCP port of the retransmission session (41003). I hope this=20
> also clarifies
> your question in the next email. If you think I got the ports=20
> wrong, let me
> know.
>=20
> -acbegen
> =20
> >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> >         a=3Dmid:4
> >=20
> >=20
> > -acbegen
> > =20
> >=20
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > Sent: Saturday, April 25, 2009 1:11 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions (RAMS)" available
> > >=20
> > > Ali,
> > > Can you please write three addresses from this strange SDP.
> > >=20
> > > 1. The address and port of multicast
> > >=20
> > > 2. The address and port of the RS where the RTCP FB should go to
> > >=20
> > > 3. The address and port on the RR where the unicast stream=20
> > > should be sent to
> > >=20
> > > Roni
> > >=20
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > Sent: Saturday, April 25, 2009 11:02 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > Multicast RTPSessions
> > > (RAMS)" available
> > >=20
> > > > Ali,
> > > > The example SDP is syntactically correct but it does not=20
> > > say that the
> > > > rtp/rtcp mux will be used and it does not give the=20
> > > > information to where the
> > > > unicast stream will be sent when the RR sends a RAMS-R.=20
> > >=20
> > > To my knowledge, the first line in the following SDP tells=20
> > > the RRs on which
> > > port they will receive the retransmission/burst packets.
> > >=20
> > >         m=3Dvideo 41002 RTP/AVPF 99
> > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > Acq. Support)
> > >         c=3DIN IP4 192.0.2.1
> > >         a=3Drtpmap:99 rtx/90000
> > >         a=3Drtcp:41003
> > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > >         a=3Dmid:4
> > >=20
> > > There is a typo, you are right. That "a=3Drecvonly" line should=20
> > > only exist in
> > > the SDP sent to RRs. In the SDP sent to RS, we should rather have
> > > "a=3Dsendonly". I will make this correction in the next version.
> > >=20
> > > The feedback target for the SSM session and the RTCP port for the
> > > retransmission session are also defined in the SDP.
> > >=20
> > > Hope this clarifies.
> > >=20
> > > BR,
> > > -acbegen
> > > =20
> > > >=20
> > > > I am not sure why the unicast stream m-line has a port=20
> > > number with an
> > > > attribute of recvonly. What is the use case for that. The=20
> > > > only information
> > > > that the RR will need is the RTCP port on the RS to where to=20
> > > > send the RAMS-R
> > > > message.
> > > > Roni
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > Sent: Friday, April 24, 2009 10:37 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > >=20
> > > > This is a part of an example SDP sent to RS and RR in a SAP=20
> > > > announcement,
> > > > for example. So, the SDP describes what both parties should=20
> > > > do (RR cannot
> > > > say that he wants to receive this multicast on its favorite=20
> > > port). The
> > > > individual SDPs sent to RR or RS may include other portions=20
> > > > of descriptions
> > > > that will contain specific information.=20
> > > >=20
> > > > -acbegen
> > > >=20
> > > > > -----Original Message-----
> > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions (RAMS)" available
> > > > >=20
> > > > > Ali,
> > > > > I think you get it wrong, this SDP is from the RS and not the=20
> > > > > RR so the RS
> > > > > cannot specify to which address it will send it can only=20
> > > > > specify to which
> > > > > address it can receive RTP stream. In this SDP the relevant=20
> > > > > information is
> > > > > that the request for retransmission will be sent by the RR to=20
> > > > > port 41003
> > > > > Roni
> > > > >=20
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > >=20
> > > > > Oh, I see. The burst goes to the port that we would=20
> > > > normally send the
> > > > > retransmissions. For example, consider the SDP from the draft:
> > > > >=20
> > > > >         m=3Dvideo 41000 RTP/AVPF 98
> > > > >         i=3DPrimary Multicast Stream #2
> > > > >         c=3DIN IP4 224.1.1.2/255
> > > > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > > >         a=3Drtpmap:98 MP2T/90000
> > > > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > > > >         a=3Drtcp-fb:98 nack
> > > > >         a=3Drtcp-fb:98 nack ssli
> > > > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > > > >         a=3Dmid:3
> > > > >         m=3Dvideo 41002 RTP/AVPF 99
> > > > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > > > Acq. Support)
> > > > >         c=3DIN IP4 192.0.2.1
> > > > >         a=3Drecvonly
> > > > >         a=3Drtpmap:99 rtx/90000
> > > > >         a=3Drtcp:41003
> > > > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > > > >         a=3Dmid:4
> > > > >=20
> > > > > In this case, the burst will go to port 41002 on the RR.=20
> > > The RAMS-I
> > > > > message(s), which is an RTCP feedback message, will go to=20
> > > > port 41003.
> > > > >=20
> > > > > HTH,
> > > > > -acbegen
> > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > >=20
> > > > > > Ali,
> > > > > > I will try to explain in simple way
> > > > > >=20
> > > > > > When the RS receives the RAMS-R it need to start sending an=20
> > > > > > RTP stream to
> > > > > > the RR.=20
> > > > > > In order to send a unicast packet, the RS needs to know a=20
> > > > > > transport address
> > > > > > on the RR to where the RTP stream will be sent. The current=20
> > > > > > draft does not
> > > > > > say how the RS knows this address. There is no SDP=20
> from RR to=20
> > > > > > RS like you
> > > > > > mention in your response.
> > > > > > This is why I say that the current draft does not specify a=20
> > > > > > solution that
> > > > > > can be implemented as a working solution
> > > > > > Roni
> > > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > > To: Roni Even; avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions
> > > > > > (RAMS)" available
> > > > > >=20
> > > > > > Hi Roni,=20
> > > > > >=20
> > > > > > > -----Original Message-----
> > > > > > > From: avt-bounces@ietf.org=20
> [mailto:avt-bounces@ietf.org] On=20
> > > > > > > Behalf Of Roni Even
> > > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > > To: avt@ietf.org
> > > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > >=20
> > > > > > > Hi,
> > > > > > >=20
> > > > > > > I think that the current draft does not give a=20
> > description of=20
> > > > > > > a system that works since there is no text=20
> > explaining how the=20
> > > > > >=20
> > > > > > What do you mean by "a system that works"?=20
> > > > > >=20
> > > > > > > RS knows the unicast transport address on the RR to=20
> > where to=20
> > > > > > > send the stream.=20
> > > > > >=20
> > > > > > Once RS receives the request packet from an RR, RS=20
> knows its=20
> > > > > > address. Ports
> > > > > > are defined in the SDP. If you are asking about=20
> "NAT" issues,=20
> > > > > > we have a
> > > > > > section for it, and we plan to complete it as we move=20
> > > > > > forward. It is not as
> > > > > > critical as the other parts for now.
> > > > > > =20
> > > > > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > > > > otherwise the RAMS-R should have an optional=20
> parameter that=20
> > > > > > > supplies this information and a flag for using=20
> RTP/RTCP mux.
> > > > > >=20
> > > > > > IMO, we cannot mandate muxing as it is not required to make=20
> > > > > > RAMS work. If
> > > > > > muxing is supported, the SDP should reflect it.
> > > > > >=20
> > > > > > BR,
> > > > > > -acbegen
> > > > > > =20
> > > > > > > Thanks
> > > > > > >=20
> > > > > > > Roni Even
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > From: avt-bounces@ietf.org=20
> [mailto:avt-bounces@ietf.org] On=20
> > > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > > To: avt@ietf.org
> > > > > > > Subject: [AVT] New draft of "Rapid Acquisition of=20
> Multicast=20
> > > > > > > RTP Sessions (RAMS)" available
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > There is a new draft of the "Rapid Acquisition of=20
> Multicast=20
> > > > > > > RTP Sessions (RAMS)" draft available at=20
> > > > > > >=20
> > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > > ynchronization-for-rtp-03.txt=20
> <http://www.ietf.org/internet->
> > > > > >=20
> > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > We have incorporated the changes from the technical=20
> > breakout=20
> > > > > > > session in San Francisco. The major changes in=20
> this version=20
> > > > > > > of the draft include=20
> > > > > > >=20
> > > > > > > 1- Changing the document title to avoid confusion=20
> > with other=20
> > > > > > > ongoing "synchronization" drafts
> > > > > > >=20
> > > > > > > 2- changing the message names to reflect the title change
> > > > > > >=20
> > > > > > > 3- clarification of the RTCP message semantics, including=20
> > > > > > > changes to the "Request" and "Inform" messages
> > > > > > >=20
> > > > > > > 4- additional description/motivation for the=20
> > various message=20
> > > > > > > flows has been added
> > > > > > >=20
> > > > > > > 5- RTP/RTCP muxing has been added
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > We hope to make this a Working Group item, and=20
> will change=20
> > > > > > > the name of the draft to avoid conflicts with other=20
> > > > > > > "synchronization" drafts at that time.
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > Bill VerSteeg
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > > =20
> > > > > > >=20
> > > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

From ron.even.tlv@gmail.com  Sun Apr 26 11:48:38 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <avt@ietf.org>
Date: Sun, 26 Apr 2009 21:48:12 +0300
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Subject: [AVT] SDP example in draft-versteeg-avt-rapid-synchronization-for-rtp-03
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Hi,

 

I am looking for more feedback on the following topic (not by Roni or Ali).

The example in the draft is coming from a distribution source using for
example SAP (multicast distribution).

There are three entities in the communication. The multicast distribution
source, the unicast retransmission which also serves as a feedback target
and a client that wants to receive the multicast stream and will ask for
unicast  resynchronization or retransmission.

 

This SDP is coming from the distribution source and according to the authors
specify the multicast address, the unicast transport address for the RTCP
receiver reports for the multicast (SSM), the uincast transport address on
the feedback target/retransmission server that will  be used by the clients
to ask for retransmission streams using RTCP-FB and the transport address on
the client that will be used by the feedback target to send the unicast RTP
stream to.

 

Does this SDP look like it supplies all this information enabling
interoperable applications.

 

         v=0

        o=ali 1122334455 1122334466 IN IP4 rams.example.com

        s=Rapid Acquisition Examples

        t=0 0

        a=group:FID 1 2

        a=group:FID 3 4

        a=rtcp-unicast:rsi

        m=video 40000 RTP/AVPF 96

        i=Primary Multicast Stream #1

        c=IN IP4 224.1.1.1/255

        a=source-filter: incl IN IP4 224.1.1.1 192.0.2.2

        a=recvonly

        a=rtpmap:96 MP2T/90000

        a=rtcp:40001 IN IP4 192.0.2.1

        a=rtcp-fb:96 nack

        a=mid:1

        m=video 40002 RTP/AVPF 97

        i=Unicast Retransmission Stream #1 (Ret. Support Only)

        c=IN IP4 192.0.2.1

        a=recvonly

        a=rtpmap:97 rtx/90000

        a=rtcp:40003

        a=fmtp:97 apt=96

        a=fmtp:97 rtx-time=3000

        a=mid:2

 

 

Thanks

Roni Even


------=_NextPart_000_0182_01C9C6B8.B8AD0A40
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xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" =
xmlns=3D"http://www.w3.org/TR/REC-html40">

<head>
<meta http-equiv=3DContent-Type content=3D"text/html; =
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<meta name=3DGenerator content=3D"Microsoft Word 12 (filtered medium)">
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<div class=3DSection1>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Hi,<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>I am looking for more feedback on the following =
topic (not
by Roni or Ali).<o:p></o:p></p>

<p class=3DMsoNormal>The example in the draft is coming from a =
distribution
source using for example SAP (multicast distribution).<o:p></o:p></p>

<p class=3DMsoNormal>There are three entities in the communication. The =
multicast
distribution source, the unicast retransmission which also serves as a =
feedback
target and a client that wants to receive the multicast stream and will =
ask for
unicast &nbsp;resynchronization or retransmission.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>This SDP is coming from the distribution source and
according to the authors specify the multicast address, the unicast =
transport
address for the RTCP receiver reports for the multicast (SSM), the =
uincast
transport address on the feedback target/retransmission server that will =
&nbsp;be
used by the clients to ask for retransmission streams using RTCP-FB and =
the
transport address on the client that will be used by the feedback target =
to
send the unicast RTP stream to.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Does this SDP look like it supplies all this =
information
enabling interoperable applications.<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
v=3D0<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dali =
1122334455
1122334466 IN IP4 rams.example.com<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
s=3DRapid
Acquisition Examples<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0 =
0<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dgroup:FID 1 2<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dgroup:FID 3 4<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
a=3Drtcp-unicast:rsi<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
m=3Dvideo 40000
RTP/AVPF 96<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
i=3DPrimary
Multicast Stream #1<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN =
IP4
224.1.1.1/255<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dsource-filter:
incl IN IP4 224.1.1.1 192.0.2.2<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drecvonly<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtpmap:96
MP2T/90000<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtcp:40001 IN
IP4 192.0.2.1<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtcp-fb:96 nack<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dmid:1<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
m=3Dvideo 40002
RTP/AVPF 97<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
i=3DUnicast
Retransmission Stream #1 (Ret. Support Only)<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN =
IP4
192.0.2.1<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drecvonly<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtpmap:97
rtx/90000<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtcp:40003<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dfmtp:97 apt=3D96<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dfmtp:97
rtx-time=3D3000<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Dmid:2<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

</div>

</body>

</html>

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From ron.even.tlv@gmail.com  Sun Apr 26 11:48:43 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <avt@ietf.org>
References: <81B9B88E2F9D574AA5328A85547CDE9101071D80@xmb-rtp-21d.amer.cisco.com> <49f1a1d6.190c660a.58fe.ffffa90e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C80DA@xmb-sjc-215.amer.cisco.com> <49f208e8.05a0660a.5bd2.6764@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C81C9@xmb-sjc-215.amer.cisco.com> <49f211fc.07a5660a.3f7f.ffff8f4a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8202@xmb-sjc-215.amer.cisco.com> <49f2efa2.1c215e0a.4454.ffffaf91@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8403@xmb-sjc-215.amer.cisco.com> <49f36ee2.0437560a.11df.1eef@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8405@xmb-sjc-215.amer.cisco.com> <49f37fde.09a1660a.2b46.ffffb694@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C8408@xmb-sjc-215.amer.cisco.com> <49f4182f.07a5660a.3f7f.ffffe36a@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D54092C841F@xmb-sjc-215.amer.cisco.com>
In-Reply-To: <04CAD96D4C5A3D48B1919248A8FE0D54092C841F@xmb-sjc-215.amer.cisco.com>
Date: Sun, 26 Apr 2009 21:48:12 +0300
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Cc: "'Bill Ver Steeg \(versteb\)'" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Ali,
Your SDP example will not help anyone who will try to develop interop
application, the c-line in the unicast stream and the port should refer to
the same device since there is no other attribute in the example that points
to a different semantics to anyone who will look at it. The semantics that
you are using where the c-line is for the RS and the port is for the RR and
the RS assuming that the RS somehow knows the IP address of the RR based on
the RAMS-R request may work in some cases but should not be the one shown as
the primary example and should not be the recommended way, The recommended
way should be to convey the address of the RR in the RAMS-R and this is the
reason I pointed to RTSP. The semantics of the SDP in the example is not
documented in your text and is not obvious to anyone who will just look at
it.
   
I also strongly object to mandating that the client MUST use a port defined
by the server to receive the unicast stream. The RS can recommend a port but
it is the client (RR) decision to decide which port to use.



Roni Even

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
Sent: Sunday, April 26, 2009 8:02 PM
To: Roni Even; avt@ietf.org
Cc: Bill Ver Steeg (versteb)
Subject: RE: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions
(RAMS)" available

Hi Roni,

Thanks for the detailed answer. Admittedly, I don't know much about RTSP
beyond its basics. However, I doubt how much it applies in our case.
"Unicast" in RTSP context mostly refers to video-on-demand kind of
applications where the primary distribution is over unicast. In contrast,
our primary distribution is over multicast, and we use unicast as a repair
stream. Thus, I still believe the SDP we have should work fine. RS gathers
the IP address of the RR when it receives the RAMS request. The samething
happens when RR sends a NACK for requesting retransmissions.

That is not to say that this SDP is the only SDP that will work in this
system. There may be others. But, I still don't agree with you when you say
that this SDP won't work. My expectation is that we will include other SDP
examples as we move forward.

On your last point, NAT is a different issue and there has to be a way to
deal with it. Muxing sounds reasonable, and we may or may not mandate it, I
am not sure at this point. An alternative could be that RR opens ports on
its NAT device and includes them in the request message it sends. These
could be in RAMS-R TLV extensions, or could be a separate RTCP packet solely
for this purpose within the same RTCP compound packet that carries the
RAMS-R. I believe Bill mentioned these briefly in his earlier email. We are
open to suggestions at this point and the NAT section will cover these
issues once we have a WG item.

-acbegen

 

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> Sent: Sunday, April 26, 2009 1:14 AM
> To: Ali C. Begen (abegen); avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions (RAMS)" available
> 
> Ali,
> I looked at RTSP
> http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-20#appendix-D 
> 
> 
> 
> D.1.2.  Media Streams
> 
>    The "m=" field is used to enumerate the streams.  It is 
> expected that
>    all the specified streams will be rendered with appropriate
>    synchronization.  If the session is over multicast, the port number
>    indicated SHOULD be used for reception.  The client MAY try to
>    override the destination port, through the Transport header.  The
>    servers MAY allow this, the response will indicate if 
> allowed or not.
>    If the session is unicast, the port numbers are the ones 
> RECOMMENDED
>    by the server to the client, about which receiver ports to use; the
>    client MUST still include its receiver ports in its SETUP request.
>    The client MAY ignore this recommendation.  If the server has no
>    preference, it SHOULD set the port number value to zero.
> 
> 
> Note that the client has the choice on which port to receive 
> the unicast
> stream, this is why there should be such a parameter in RAMS-R
> 
> 
> Also
> 
> " D.1.8.  Connection Information
> 
>    In SDP, the "c=" field contains the destination address 
> for the media
>    stream.  For on-demand unicast streams and some multicast streams,
>    the destination address MAY be specified by the client via 
> the SETUP
>    request, thus overriding any specified address.  To 
> identify streams
>    without a fixed destination address, where the client is 
> required to
>    specify a destination address, the "c=" field SHOULD be 
> set to a null
>    value.  For addresses of type "IP4", this value MUST be "0.0.0.0",
>    and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0" 
> (can also be
>    written as "::"), i.e. the unspecified address according 
> to RFC 4291
>    [RFC4291]."
> 
> 
> As for the c-line in the unicast stream the whole section you quoted
> 
> " If the session is multicast, the connection address will be an IP
>       multicast group address.  If the session is not multicast, then
>       the connection address contains the unicast IP address of the
>       expected data source or data relay or data sink as determined by
>       additional attribute fields.  It is not expected that unicast
>       addresses will be given in a session description that is
>       communicated by a multicast announcement, though this is not
>       prohibited."
> 
> Now the expectation here is for the address of the stream 
> described by the
> port in the m-line, I do not see any other attribute. Also 
> look at the above
> text from RTSP. The RS does not know the IP address on the RR 
> to where to
> send the RTP unicast repair stream. If you assume it is to 
> the IP address
> from where the RTCP RAMS-R came from this is not mandated 
> that the client
> will have such implementation. Furthermore the address that 
> the RS will see
> in the received RTCP RAMS-R packet may have been change by a 
> NAT or SBC.
> 
> 
> Roni
> 
> 
> 
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> Sent: Sunday, April 26, 2009 1:23 AM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> Multicast RTPSessions
> (RAMS)" available
> 
> Inline.
> 
> >         m=video 41000 RTP/AVPF 98
> >         i=Primary Multicast Stream #2
> >         c=IN IP4 224.1.1.2/255
> >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > 
> > Source (192.0.2.2) sends the rtp packets to the multicast 
> > group (224.1.1.2)
> > with a dest port 41000.
> > 
> >         a=rtpmap:98 MP2T/90000
> >         a=rtcp:41001 IN IP4 192.0.2.1
> > 
> > The feedback target (RS) address for this SSM session is 
> > 192.0.2.1 and its
> > port is 41001. This is the address/port where RR sends the 
> RAMS-R, or
> > receiver reports for the SSM session.
> > 
> >         a=rtcp-fb:98 nack
> >         a=rtcp-fb:98 nack ssli
> >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> >         a=mid:3
> >         m=video 41002 RTP/AVPF 99
> > 
> > The retransmission packets go to port 41002. This is the port 
> > RRs listen to
> > for retransmission and RAMS.
> > 
> > Roni E: This implies that the multicast source defines the 
> > port that all
> > clients MUST use to receive the unicast information from the 
> > Burst source.  
> 
> Yes, that is what it implies (One nit is that I would call it 
> unicast RTP
> packets rather than information since we may use another port for RTCP
> packets).
>  
> >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > Acq. Support)
> >         c=IN IP4 192.0.2.1
> > 
> > This is where the retransmission packets come from, same as 
> > the feedback
> > target (in this example).
> > 
> > Roni E: If this is the address of the feedback target it 
> > belongs to the
> > above m-line (m=video 41002 RTP/AVPF 99) that according to 
> > you gives the
> > port on the RR and not on the RS. The problem is that 
> 
> This line actually specifies the sender for the unicast retransmission
> session. In our case, it just happens to be the feedback 
> target for the
> primary SSM session, too. I recall some discussion about using a
> retransmission source which is different than the feedback 
> target for the
> SSM. In that case, the address would be different, I guess.
> 
> > according to SDP the
> > combination of the c= and port on the m-line provides the 
> > transport address
> > to where a stream will be sent so according to SDP this 
> 
> The port on the m line is the port where the retranmission 
> packets are sent.
> However, c line denotes the source for these packets. RFC 
> 4566 page 13 says
> that "... If the session is not multicast, then the connection address
> contains the unicast IP address of the expected data source 
> or data relay or
> data sink as determined by additional attribute fields..."
> 
> Am I missing something?
> 
> > m-line specify an
> > address and port on the Feedback Target and not on the RR and 
> > only to this
> > address RTP payload type 99 will be sent. I am positive that 
> > this is not
> > what you want.
> 
> There can be other payload types we might need to send on this
> retransmission session in the future. In that case, we will add them.
>  
> >         a=rtpmap:99 rtx/90000
> >         a=rtcp:41003
> > 
> > This is where the RTCP packets for the retransmission 
> session go. For
> > example, RAMS-I goes to this port on RRs. RAMS-T goes to this 
> > port on RS.
> > 
> > Roni E: So now you assume that the RR must use the same port 
> > as the FT for
> > this communication. I could not find any place that one 
> > m-line gives the
> 
> Not the m line, but the "a=rtcp" line. Is this wrong?
> 
> > address for both sides. This RTCP port is on 192.0.2.1 so it 
> > will be used
> > for RAMS-R but not for RAMS-I.
> 
> RAMS-R should go to the feedback target address and port for 
> the primary SSM
> session since it is a feedback message for the SSM session. 
> But, RAMS-T is a
> feedback message for the retransmission session, thus, it 
> should go to the
> RTCP port of the retransmission session (41003). I hope this 
> also clarifies
> your question in the next email. If you think I got the ports 
> wrong, let me
> know.
> 
> -acbegen
>  
> >         a=fmtp:99 apt=98; rtx-time=5000
> >         a=mid:4
> > 
> > 
> > -acbegen
> >  
> > 
> > > -----Original Message-----
> > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > Sent: Saturday, April 25, 2009 1:11 PM
> > > To: Ali C. Begen (abegen); avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions (RAMS)" available
> > > 
> > > Ali,
> > > Can you please write three addresses from this strange SDP.
> > > 
> > > 1. The address and port of multicast
> > > 
> > > 2. The address and port of the RS where the RTCP FB should go to
> > > 
> > > 3. The address and port on the RR where the unicast stream 
> > > should be sent to
> > > 
> > > Roni
> > > 
> > > -----Original Message-----
> > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > Sent: Saturday, April 25, 2009 11:02 PM
> > > To: Roni Even; avt@ietf.org
> > > Cc: Bill Ver Steeg (versteb)
> > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > Multicast RTPSessions
> > > (RAMS)" available
> > > 
> > > > Ali,
> > > > The example SDP is syntactically correct but it does not 
> > > say that the
> > > > rtp/rtcp mux will be used and it does not give the 
> > > > information to where the
> > > > unicast stream will be sent when the RR sends a RAMS-R. 
> > > 
> > > To my knowledge, the first line in the following SDP tells 
> > > the RRs on which
> > > port they will receive the retransmission/burst packets.
> > > 
> > >         m=video 41002 RTP/AVPF 99
> > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > Acq. Support)
> > >         c=IN IP4 192.0.2.1
> > >         a=rtpmap:99 rtx/90000
> > >         a=rtcp:41003
> > >         a=fmtp:99 apt=98; rtx-time=5000
> > >         a=mid:4
> > > 
> > > There is a typo, you are right. That "a=recvonly" line should 
> > > only exist in
> > > the SDP sent to RRs. In the SDP sent to RS, we should rather have
> > > "a=sendonly". I will make this correction in the next version.
> > > 
> > > The feedback target for the SSM session and the RTCP port for the
> > > retransmission session are also defined in the SDP.
> > > 
> > > Hope this clarifies.
> > > 
> > > BR,
> > > -acbegen
> > >  
> > > > 
> > > > I am not sure why the unicast stream m-line has a port 
> > > number with an
> > > > attribute of recvonly. What is the use case for that. The 
> > > > only information
> > > > that the RR will need is the RTCP port on the RS to where to 
> > > > send the RAMS-R
> > > > message.
> > > > Roni
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > Sent: Friday, April 24, 2009 10:37 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > > 
> > > > This is a part of an example SDP sent to RS and RR in a SAP 
> > > > announcement,
> > > > for example. So, the SDP describes what both parties should 
> > > > do (RR cannot
> > > > say that he wants to receive this multicast on its favorite 
> > > port). The
> > > > individual SDPs sent to RR or RS may include other portions 
> > > > of descriptions
> > > > that will contain specific information. 
> > > > 
> > > > -acbegen
> > > > 
> > > > > -----Original Message-----
> > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions (RAMS)" available
> > > > > 
> > > > > Ali,
> > > > > I think you get it wrong, this SDP is from the RS and not the 
> > > > > RR so the RS
> > > > > cannot specify to which address it will send it can only 
> > > > > specify to which
> > > > > address it can receive RTP stream. In this SDP the relevant 
> > > > > information is
> > > > > that the request for retransmission will be sent by the RR to 
> > > > > port 41003
> > > > > Roni
> > > > > 
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > > 
> > > > > Oh, I see. The burst goes to the port that we would 
> > > > normally send the
> > > > > retransmissions. For example, consider the SDP from the draft:
> > > > > 
> > > > >         m=video 41000 RTP/AVPF 98
> > > > >         i=Primary Multicast Stream #2
> > > > >         c=IN IP4 224.1.1.2/255
> > > > >         a=source-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > > >         a=rtpmap:98 MP2T/90000
> > > > >         a=rtcp:41001 IN IP4 192.0.2.1
> > > > >         a=rtcp-fb:98 nack
> > > > >         a=rtcp-fb:98 nack ssli
> > > > >         a=ssrc:123321 cname:iptv-ch32@rams.example.com
> > > > >         a=mid:3
> > > > >         m=video 41002 RTP/AVPF 99
> > > > >         i=Unicast Retransmission Stream #2 (Ret. and Rapid 
> > > > > Acq. Support)
> > > > >         c=IN IP4 192.0.2.1
> > > > >         a=recvonly
> > > > >         a=rtpmap:99 rtx/90000
> > > > >         a=rtcp:41003
> > > > >         a=fmtp:99 apt=98; rtx-time=5000
> > > > >         a=mid:4
> > > > > 
> > > > > In this case, the burst will go to port 41002 on the RR. 
> > > The RAMS-I
> > > > > message(s), which is an RTCP feedback message, will go to 
> > > > port 41003.
> > > > > 
> > > > > HTH,
> > > > > -acbegen
> > > > > 
> > > > > > -----Original Message-----
> > > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com] 
> > > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > 
> > > > > > Ali,
> > > > > > I will try to explain in simple way
> > > > > > 
> > > > > > When the RS receives the RAMS-R it need to start sending an 
> > > > > > RTP stream to
> > > > > > the RR. 
> > > > > > In order to send a unicast packet, the RS needs to know a 
> > > > > > transport address
> > > > > > on the RR to where the RTP stream will be sent. The current 
> > > > > > draft does not
> > > > > > say how the RS knows this address. There is no SDP 
> from RR to 
> > > > > > RS like you
> > > > > > mention in your response.
> > > > > > This is why I say that the current draft does not specify a 
> > > > > > solution that
> > > > > > can be implemented as a working solution
> > > > > > Roni
> > > > > > 
> > > > > > -----Original Message-----
> > > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com] 
> > > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > > To: Roni Even; avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of 
> > > > > > Multicast RTPSessions
> > > > > > (RAMS)" available
> > > > > > 
> > > > > > Hi Roni, 
> > > > > > 
> > > > > > > -----Original Message-----
> > > > > > > From: avt-bounces@ietf.org 
> [mailto:avt-bounces@ietf.org] On 
> > > > > > > Behalf Of Roni Even
> > > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > > To: avt@ietf.org
> > > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of 
> > > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > > 
> > > > > > > Hi,
> > > > > > > 
> > > > > > > I think that the current draft does not give a 
> > description of 
> > > > > > > a system that works since there is no text 
> > explaining how the 
> > > > > > 
> > > > > > What do you mean by "a system that works"? 
> > > > > > 
> > > > > > > RS knows the unicast transport address on the RR to 
> > where to 
> > > > > > > send the stream. 
> > > > > > 
> > > > > > Once RS receives the request packet from an RR, RS 
> knows its 
> > > > > > address. Ports
> > > > > > are defined in the SDP. If you are asking about 
> "NAT" issues, 
> > > > > > we have a
> > > > > > section for it, and we plan to complete it as we move 
> > > > > > forward. It is not as
> > > > > > critical as the other parts for now.
> > > > > >  
> > > > > > > If you mandate the use of RTP/RTCP mux it should say so 
> > > > > > > otherwise the RAMS-R should have an optional 
> parameter that 
> > > > > > > supplies this information and a flag for using 
> RTP/RTCP mux.
> > > > > > 
> > > > > > IMO, we cannot mandate muxing as it is not required to make 
> > > > > > RAMS work. If
> > > > > > muxing is supported, the SDP should reflect it.
> > > > > > 
> > > > > > BR,
> > > > > > -acbegen
> > > > > >  
> > > > > > > Thanks
> > > > > > > 
> > > > > > > Roni Even
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > From: avt-bounces@ietf.org 
> [mailto:avt-bounces@ietf.org] On 
> > > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > > To: avt@ietf.org
> > > > > > > Subject: [AVT] New draft of "Rapid Acquisition of 
> Multicast 
> > > > > > > RTP Sessions (RAMS)" available
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > There is a new draft of the "Rapid Acquisition of 
> Multicast 
> > > > > > > RTP Sessions (RAMS)" draft available at 
> > > > > > > 
> > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > > ynchronization-for-rtp-03.txt 
> <http://www.ietf.org/internet->
> > > > > > 
> > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt> 
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > We have incorporated the changes from the technical 
> > breakout 
> > > > > > > session in San Francisco. The major changes in 
> this version 
> > > > > > > of the draft include 
> > > > > > > 
> > > > > > > 1- Changing the document title to avoid confusion 
> > with other 
> > > > > > > ongoing "synchronization" drafts
> > > > > > > 
> > > > > > > 2- changing the message names to reflect the title change
> > > > > > > 
> > > > > > > 3- clarification of the RTCP message semantics, including 
> > > > > > > changes to the "Request" and "Inform" messages
> > > > > > > 
> > > > > > > 4- additional description/motivation for the 
> > various message 
> > > > > > > flows has been added
> > > > > > > 
> > > > > > > 5- RTP/RTCP muxing has been added
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > We hope to make this a Working Group item, and 
> will change 
> > > > > > > the name of the draft to avoid conflicts with other 
> > > > > > > "synchronization" drafts at that time.
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > Bill VerSteeg
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > >  
> > > > > > > 
> > > > > > > 
> > > > > > 
> > > > > > 
> > > > > 
> > > > > 
> > > > 
> > > > 
> > > 
> > > 
> > 
> > 
> 
> 


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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <avt@ietf.org>
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Cc: "Bill Ver Steeg \(versteb\)" <versteb@cisco.com>
Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTPSessions	(RAMS)" available
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Roni,

> Ali,
> Your SDP example will not help anyone who will try to develop interop
> application, the c-line in the unicast stream and the port=20

I believe this is rather a strong claim, considering that I already =
mentioned that we would add more SDP examples as we moved forward.

> should refer to
> the same device since there is no other attribute in the=20
> example that points
> to a different semantics to anyone who will look at it. The=20
> semantics that
> you are using where the c-line is for the RS and the port is=20
> for the RR and
> the RS assuming that the RS somehow knows the IP address of=20
> the RR based on
> the RAMS-R request may work in some cases but should not be=20
> the one shown as
> the primary example and should not be the recommended way,=20

Again, you are missing the point that we will add more SDP examples as =
well as discussion for NAT issues. The current SDP is just one example =
that we came up with, and which, I believe, still works and has been =
there since version -00. It may be modified, removed or replaced.

> The recommended
> way should be to convey the address of the RR in the RAMS-R=20
> and this is the
> reason I pointed to RTSP. The semantics of the SDP in the=20

This is reasonable, rtp/rtcp muxing is also reseanoable and it is indeed =
very neat, too.=20

> example is not
> documented in your text and is not obvious to anyone who will=20
> just look at
> it.

We can certainly add more detailed explanation to the SDP section. But, =
let's focus more on the draft rather than a single SDP example that it =
currently has.
   =20
> I also strongly object to mandating that the client MUST use=20
> a port defined
> by the server to receive the unicast stream. The RS can=20
> recommend a port but
> it is the client (RR) decision to decide which port to use.

We are not mandating anything yet as I mentioned several times before. I =
suggest we drop this discussion and move forward with the draft. Based =
on the comments/suggestions that may be sent to your email about the SDP =
example, we will be more than happy to incorporate what AVT decides to =
do.

-acbegen
=20
>=20
>=20
> Roni Even
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Sunday, April 26, 2009 8:02 PM
> To: Roni Even; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTPSessions
> (RAMS)" available
>=20
> Hi Roni,
>=20
> Thanks for the detailed answer. Admittedly, I don't know much=20
> about RTSP
> beyond its basics. However, I doubt how much it applies in our case.
> "Unicast" in RTSP context mostly refers to video-on-demand kind of
> applications where the primary distribution is over unicast.=20
> In contrast,
> our primary distribution is over multicast, and we use=20
> unicast as a repair
> stream. Thus, I still believe the SDP we have should work=20
> fine. RS gathers
> the IP address of the RR when it receives the RAMS request.=20
> The samething
> happens when RR sends a NACK for requesting retransmissions.
>=20
> That is not to say that this SDP is the only SDP that will=20
> work in this
> system. There may be others. But, I still don't agree with=20
> you when you say
> that this SDP won't work. My expectation is that we will=20
> include other SDP
> examples as we move forward.
>=20
> On your last point, NAT is a different issue and there has to=20
> be a way to
> deal with it. Muxing sounds reasonable, and we may or may not=20
> mandate it, I
> am not sure at this point. An alternative could be that RR=20
> opens ports on
> its NAT device and includes them in the request message it=20
> sends. These
> could be in RAMS-R TLV extensions, or could be a separate=20
> RTCP packet solely
> for this purpose within the same RTCP compound packet that carries the
> RAMS-R. I believe Bill mentioned these briefly in his earlier=20
> email. We are
> open to suggestions at this point and the NAT section will cover these
> issues once we have a WG item.
>=20
> -acbegen
>=20
> =20
>=20
> > -----Original Message-----
> > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > Sent: Sunday, April 26, 2009 1:14 AM
> > To: Ali C. Begen (abegen); avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions (RAMS)" available
> >=20
> > Ali,
> > I looked at RTSP
> >=20
> http://tools.ietf.org/html/draft-ietf-mmusic-rfc2326bis-20#appendix-D=20
> >=20
> >=20
> >=20
> > D.1.2.  Media Streams
> >=20
> >    The "m=3D" field is used to enumerate the streams.  It is=20
> > expected that
> >    all the specified streams will be rendered with appropriate
> >    synchronization.  If the session is over multicast, the=20
> port number
> >    indicated SHOULD be used for reception.  The client MAY try to
> >    override the destination port, through the Transport header.  The
> >    servers MAY allow this, the response will indicate if=20
> > allowed or not.
> >    If the session is unicast, the port numbers are the ones=20
> > RECOMMENDED
> >    by the server to the client, about which receiver ports=20
> to use; the
> >    client MUST still include its receiver ports in its=20
> SETUP request.
> >    The client MAY ignore this recommendation.  If the server has no
> >    preference, it SHOULD set the port number value to zero.
> >=20
> >=20
> > Note that the client has the choice on which port to receive=20
> > the unicast
> > stream, this is why there should be such a parameter in RAMS-R
> >=20
> >=20
> > Also
> >=20
> > " D.1.8.  Connection Information
> >=20
> >    In SDP, the "c=3D" field contains the destination address=20
> > for the media
> >    stream.  For on-demand unicast streams and some=20
> multicast streams,
> >    the destination address MAY be specified by the client via=20
> > the SETUP
> >    request, thus overriding any specified address.  To=20
> > identify streams
> >    without a fixed destination address, where the client is=20
> > required to
> >    specify a destination address, the "c=3D" field SHOULD be=20
> > set to a null
> >    value.  For addresses of type "IP4", this value MUST be=20
> "0.0.0.0",
> >    and for type "IP6", this value MUST be "0:0:0:0:0:0:0:0"=20
> > (can also be
> >    written as "::"), i.e. the unspecified address according=20
> > to RFC 4291
> >    [RFC4291]."
> >=20
> >=20
> > As for the c-line in the unicast stream the whole section you quoted
> >=20
> > " If the session is multicast, the connection address will be an IP
> >       multicast group address.  If the session is not=20
> multicast, then
> >       the connection address contains the unicast IP address of the
> >       expected data source or data relay or data sink as=20
> determined by
> >       additional attribute fields.  It is not expected that unicast
> >       addresses will be given in a session description that is
> >       communicated by a multicast announcement, though this is not
> >       prohibited."
> >=20
> > Now the expectation here is for the address of the stream=20
> > described by the
> > port in the m-line, I do not see any other attribute. Also=20
> > look at the above
> > text from RTSP. The RS does not know the IP address on the RR=20
> > to where to
> > send the RTP unicast repair stream. If you assume it is to=20
> > the IP address
> > from where the RTCP RAMS-R came from this is not mandated=20
> > that the client
> > will have such implementation. Furthermore the address that=20
> > the RS will see
> > in the received RTCP RAMS-R packet may have been change by a=20
> > NAT or SBC.
> >=20
> >=20
> > Roni
> >=20
> >=20
> >=20
> > -----Original Message-----
> > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > Sent: Sunday, April 26, 2009 1:23 AM
> > To: Roni Even; avt@ietf.org
> > Cc: Bill Ver Steeg (versteb)
> > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast RTPSessions
> > (RAMS)" available
> >=20
> > Inline.
> >=20
> > >         m=3Dvideo 41000 RTP/AVPF 98
> > >         i=3DPrimary Multicast Stream #2
> > >         c=3DIN IP4 224.1.1.2/255
> > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > >=20
> > > Source (192.0.2.2) sends the rtp packets to the multicast=20
> > > group (224.1.1.2)
> > > with a dest port 41000.
> > >=20
> > >         a=3Drtpmap:98 MP2T/90000
> > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > >=20
> > > The feedback target (RS) address for this SSM session is=20
> > > 192.0.2.1 and its
> > > port is 41001. This is the address/port where RR sends the=20
> > RAMS-R, or
> > > receiver reports for the SSM session.
> > >=20
> > >         a=3Drtcp-fb:98 nack
> > >         a=3Drtcp-fb:98 nack ssli
> > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > >         a=3Dmid:3
> > >         m=3Dvideo 41002 RTP/AVPF 99
> > >=20
> > > The retransmission packets go to port 41002. This is the port=20
> > > RRs listen to
> > > for retransmission and RAMS.
> > >=20
> > > Roni E: This implies that the multicast source defines the=20
> > > port that all
> > > clients MUST use to receive the unicast information from the=20
> > > Burst source. =20
> >=20
> > Yes, that is what it implies (One nit is that I would call it=20
> > unicast RTP
> > packets rather than information since we may use another=20
> port for RTCP
> > packets).
> > =20
> > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > Acq. Support)
> > >         c=3DIN IP4 192.0.2.1
> > >=20
> > > This is where the retransmission packets come from, same as=20
> > > the feedback
> > > target (in this example).
> > >=20
> > > Roni E: If this is the address of the feedback target it=20
> > > belongs to the
> > > above m-line (m=3Dvideo 41002 RTP/AVPF 99) that according to=20
> > > you gives the
> > > port on the RR and not on the RS. The problem is that=20
> >=20
> > This line actually specifies the sender for the unicast=20
> retransmission
> > session. In our case, it just happens to be the feedback=20
> > target for the
> > primary SSM session, too. I recall some discussion about using a
> > retransmission source which is different than the feedback=20
> > target for the
> > SSM. In that case, the address would be different, I guess.
> >=20
> > > according to SDP the
> > > combination of the c=3D and port on the m-line provides the=20
> > > transport address
> > > to where a stream will be sent so according to SDP this=20
> >=20
> > The port on the m line is the port where the retranmission=20
> > packets are sent.
> > However, c line denotes the source for these packets. RFC=20
> > 4566 page 13 says
> > that "... If the session is not multicast, then the=20
> connection address
> > contains the unicast IP address of the expected data source=20
> > or data relay or
> > data sink as determined by additional attribute fields..."
> >=20
> > Am I missing something?
> >=20
> > > m-line specify an
> > > address and port on the Feedback Target and not on the RR and=20
> > > only to this
> > > address RTP payload type 99 will be sent. I am positive that=20
> > > this is not
> > > what you want.
> >=20
> > There can be other payload types we might need to send on this
> > retransmission session in the future. In that case, we will=20
> add them.
> > =20
> > >         a=3Drtpmap:99 rtx/90000
> > >         a=3Drtcp:41003
> > >=20
> > > This is where the RTCP packets for the retransmission=20
> > session go. For
> > > example, RAMS-I goes to this port on RRs. RAMS-T goes to this=20
> > > port on RS.
> > >=20
> > > Roni E: So now you assume that the RR must use the same port=20
> > > as the FT for
> > > this communication. I could not find any place that one=20
> > > m-line gives the
> >=20
> > Not the m line, but the "a=3Drtcp" line. Is this wrong?
> >=20
> > > address for both sides. This RTCP port is on 192.0.2.1 so it=20
> > > will be used
> > > for RAMS-R but not for RAMS-I.
> >=20
> > RAMS-R should go to the feedback target address and port for=20
> > the primary SSM
> > session since it is a feedback message for the SSM session.=20
> > But, RAMS-T is a
> > feedback message for the retransmission session, thus, it=20
> > should go to the
> > RTCP port of the retransmission session (41003). I hope this=20
> > also clarifies
> > your question in the next email. If you think I got the ports=20
> > wrong, let me
> > know.
> >=20
> > -acbegen
> > =20
> > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > >         a=3Dmid:4
> > >=20
> > >=20
> > > -acbegen
> > > =20
> > >=20
> > > > -----Original Message-----
> > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > Sent: Saturday, April 25, 2009 1:11 PM
> > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions (RAMS)" available
> > > >=20
> > > > Ali,
> > > > Can you please write three addresses from this strange SDP.
> > > >=20
> > > > 1. The address and port of multicast
> > > >=20
> > > > 2. The address and port of the RS where the RTCP FB should go to
> > > >=20
> > > > 3. The address and port on the RR where the unicast stream=20
> > > > should be sent to
> > > >=20
> > > > Roni
> > > >=20
> > > > -----Original Message-----
> > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > Sent: Saturday, April 25, 2009 11:02 PM
> > > > To: Roni Even; avt@ietf.org
> > > > Cc: Bill Ver Steeg (versteb)
> > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > Multicast RTPSessions
> > > > (RAMS)" available
> > > >=20
> > > > > Ali,
> > > > > The example SDP is syntactically correct but it does not=20
> > > > say that the
> > > > > rtp/rtcp mux will be used and it does not give the=20
> > > > > information to where the
> > > > > unicast stream will be sent when the RR sends a RAMS-R.=20
> > > >=20
> > > > To my knowledge, the first line in the following SDP tells=20
> > > > the RRs on which
> > > > port they will receive the retransmission/burst packets.
> > > >=20
> > > >         m=3Dvideo 41002 RTP/AVPF 99
> > > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid=20
> > > > Acq. Support)
> > > >         c=3DIN IP4 192.0.2.1
> > > >         a=3Drtpmap:99 rtx/90000
> > > >         a=3Drtcp:41003
> > > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > > >         a=3Dmid:4
> > > >=20
> > > > There is a typo, you are right. That "a=3Drecvonly" line should=20
> > > > only exist in
> > > > the SDP sent to RRs. In the SDP sent to RS, we should=20
> rather have
> > > > "a=3Dsendonly". I will make this correction in the next version.
> > > >=20
> > > > The feedback target for the SSM session and the RTCP=20
> port for the
> > > > retransmission session are also defined in the SDP.
> > > >=20
> > > > Hope this clarifies.
> > > >=20
> > > > BR,
> > > > -acbegen
> > > > =20
> > > > >=20
> > > > > I am not sure why the unicast stream m-line has a port=20
> > > > number with an
> > > > > attribute of recvonly. What is the use case for that. The=20
> > > > > only information
> > > > > that the RR will need is the RTCP port on the RS to where to=20
> > > > > send the RAMS-R
> > > > > message.
> > > > > Roni
> > > > > -----Original Message-----
> > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > Sent: Friday, April 24, 2009 10:37 PM
> > > > > To: Roni Even; avt@ietf.org
> > > > > Cc: Bill Ver Steeg (versteb)
> > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > Multicast RTPSessions
> > > > > (RAMS)" available
> > > > >=20
> > > > > This is a part of an example SDP sent to RS and RR in a SAP=20
> > > > > announcement,
> > > > > for example. So, the SDP describes what both parties should=20
> > > > > do (RR cannot
> > > > > say that he wants to receive this multicast on its favorite=20
> > > > port). The
> > > > > individual SDPs sent to RR or RS may include other portions=20
> > > > > of descriptions
> > > > > that will contain specific information.=20
> > > > >=20
> > > > > -acbegen
> > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > > Sent: Friday, April 24, 2009 12:23 PM
> > > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions (RAMS)" available
> > > > > >=20
> > > > > > Ali,
> > > > > > I think you get it wrong, this SDP is from the RS=20
> and not the=20
> > > > > > RR so the RS
> > > > > > cannot specify to which address it will send it can only=20
> > > > > > specify to which
> > > > > > address it can receive RTP stream. In this SDP the relevant=20
> > > > > > information is
> > > > > > that the request for retransmission will be sent by=20
> the RR to=20
> > > > > > port 41003
> > > > > > Roni
> > > > > >=20
> > > > > > -----Original Message-----
> > > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > > Sent: Friday, April 24, 2009 10:01 PM
> > > > > > To: Roni Even; avt@ietf.org
> > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > Multicast RTPSessions
> > > > > > (RAMS)" available
> > > > > >=20
> > > > > > Oh, I see. The burst goes to the port that we would=20
> > > > > normally send the
> > > > > > retransmissions. For example, consider the SDP from=20
> the draft:
> > > > > >=20
> > > > > >         m=3Dvideo 41000 RTP/AVPF 98
> > > > > >         i=3DPrimary Multicast Stream #2
> > > > > >         c=3DIN IP4 224.1.1.2/255
> > > > > >         a=3Dsource-filter: incl IN IP4 224.1.1.2 192.0.2.2
> > > > > >         a=3Drtpmap:98 MP2T/90000
> > > > > >         a=3Drtcp:41001 IN IP4 192.0.2.1
> > > > > >         a=3Drtcp-fb:98 nack
> > > > > >         a=3Drtcp-fb:98 nack ssli
> > > > > >         a=3Dssrc:123321 cname:iptv-ch32@rams.example.com
> > > > > >         a=3Dmid:3
> > > > > >         m=3Dvideo 41002 RTP/AVPF 99
> > > > > >         i=3DUnicast Retransmission Stream #2 (Ret. and Rapid =

> > > > > > Acq. Support)
> > > > > >         c=3DIN IP4 192.0.2.1
> > > > > >         a=3Drecvonly
> > > > > >         a=3Drtpmap:99 rtx/90000
> > > > > >         a=3Drtcp:41003
> > > > > >         a=3Dfmtp:99 apt=3D98; rtx-time=3D5000
> > > > > >         a=3Dmid:4
> > > > > >=20
> > > > > > In this case, the burst will go to port 41002 on the RR.=20
> > > > The RAMS-I
> > > > > > message(s), which is an RTCP feedback message, will go to=20
> > > > > port 41003.
> > > > > >=20
> > > > > > HTH,
> > > > > > -acbegen
> > > > > >=20
> > > > > > > -----Original Message-----
> > > > > > > From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> > > > > > > Sent: Friday, April 24, 2009 11:44 AM
> > > > > > > To: Ali C. Begen (abegen); avt@ietf.org
> > > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > >=20
> > > > > > > Ali,
> > > > > > > I will try to explain in simple way
> > > > > > >=20
> > > > > > > When the RS receives the RAMS-R it need to start=20
> sending an=20
> > > > > > > RTP stream to
> > > > > > > the RR.=20
> > > > > > > In order to send a unicast packet, the RS needs to know a=20
> > > > > > > transport address
> > > > > > > on the RR to where the RTP stream will be sent.=20
> The current=20
> > > > > > > draft does not
> > > > > > > say how the RS knows this address. There is no SDP=20
> > from RR to=20
> > > > > > > RS like you
> > > > > > > mention in your response.
> > > > > > > This is why I say that the current draft does not=20
> specify a=20
> > > > > > > solution that
> > > > > > > can be implemented as a working solution
> > > > > > > Roni
> > > > > > >=20
> > > > > > > -----Original Message-----
> > > > > > > From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> > > > > > > Sent: Friday, April 24, 2009 7:38 PM
> > > > > > > To: Roni Even; avt@ietf.org
> > > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > > Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > > Multicast RTPSessions
> > > > > > > (RAMS)" available
> > > > > > >=20
> > > > > > > Hi Roni,=20
> > > > > > >=20
> > > > > > > > -----Original Message-----
> > > > > > > > From: avt-bounces@ietf.org=20
> > [mailto:avt-bounces@ietf.org] On=20
> > > > > > > > Behalf Of Roni Even
> > > > > > > > Sent: Friday, April 24, 2009 4:24 AM
> > > > > > > > To: avt@ietf.org
> > > > > > > > Cc: Bill Ver Steeg (versteb)
> > > > > > > > Subject: Re: [AVT] New draft of "Rapid Acquisition of=20
> > > > > > > > Multicast RTPSessions (RAMS)" available
> > > > > > > >=20
> > > > > > > > Hi,
> > > > > > > >=20
> > > > > > > > I think that the current draft does not give a=20
> > > description of=20
> > > > > > > > a system that works since there is no text=20
> > > explaining how the=20
> > > > > > >=20
> > > > > > > What do you mean by "a system that works"?=20
> > > > > > >=20
> > > > > > > > RS knows the unicast transport address on the RR to=20
> > > where to=20
> > > > > > > > send the stream.=20
> > > > > > >=20
> > > > > > > Once RS receives the request packet from an RR, RS=20
> > knows its=20
> > > > > > > address. Ports
> > > > > > > are defined in the SDP. If you are asking about=20
> > "NAT" issues,=20
> > > > > > > we have a
> > > > > > > section for it, and we plan to complete it as we move=20
> > > > > > > forward. It is not as
> > > > > > > critical as the other parts for now.
> > > > > > > =20
> > > > > > > > If you mandate the use of RTP/RTCP mux it should say so=20
> > > > > > > > otherwise the RAMS-R should have an optional=20
> > parameter that=20
> > > > > > > > supplies this information and a flag for using=20
> > RTP/RTCP mux.
> > > > > > >=20
> > > > > > > IMO, we cannot mandate muxing as it is not=20
> required to make=20
> > > > > > > RAMS work. If
> > > > > > > muxing is supported, the SDP should reflect it.
> > > > > > >=20
> > > > > > > BR,
> > > > > > > -acbegen
> > > > > > > =20
> > > > > > > > Thanks
> > > > > > > >=20
> > > > > > > > Roni Even
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > From: avt-bounces@ietf.org=20
> > [mailto:avt-bounces@ietf.org] On=20
> > > > > > > > Behalf Of Bill Ver Steeg (versteb)
> > > > > > > > Sent: Thursday, April 16, 2009 7:39 PM
> > > > > > > > To: avt@ietf.org
> > > > > > > > Subject: [AVT] New draft of "Rapid Acquisition of=20
> > Multicast=20
> > > > > > > > RTP Sessions (RAMS)" available
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > There is a new draft of the "Rapid Acquisition of=20
> > Multicast=20
> > > > > > > > RTP Sessions (RAMS)" draft available at=20
> > > > > > > >=20
> > > http://www.ietf.org/internet-drafts/draft-versteeg-avt-rapid-s
> > > > > > > ynchronization-for-rtp-03.txt=20
> > <http://www.ietf.org/internet->
> > > > > > >=20
> > > drafts/draft-versteeg-avt-rapid-synchronization-for-rtp-03.txt>=20
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > We have incorporated the changes from the technical=20
> > > breakout=20
> > > > > > > > session in San Francisco. The major changes in=20
> > this version=20
> > > > > > > > of the draft include=20
> > > > > > > >=20
> > > > > > > > 1- Changing the document title to avoid confusion=20
> > > with other=20
> > > > > > > > ongoing "synchronization" drafts
> > > > > > > >=20
> > > > > > > > 2- changing the message names to reflect the=20
> title change
> > > > > > > >=20
> > > > > > > > 3- clarification of the RTCP message semantics,=20
> including=20
> > > > > > > > changes to the "Request" and "Inform" messages
> > > > > > > >=20
> > > > > > > > 4- additional description/motivation for the=20
> > > various message=20
> > > > > > > > flows has been added
> > > > > > > >=20
> > > > > > > > 5- RTP/RTCP muxing has been added
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > We hope to make this a Working Group item, and=20
> > will change=20
> > > > > > > > the name of the draft to avoid conflicts with other=20
> > > > > > > > "synchronization" drafts at that time.
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > Bill VerSteeg
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > > =20
> > > > > > > >=20
> > > > > > > >=20
> > > > > > >=20
> > > > > > >=20
> > > > > >=20
> > > > > >=20
> > > > >=20
> > > > >=20
> > > >=20
> > > >=20
> > >=20
> > >=20
> >=20
> >=20
>=20
>=20

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <avt@ietf.org>
Date: Mon, 27 Apr 2009 21:13:27 +0300
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Subject: [AVT] Comments on draft-ietf-avt-rtp-ipmr-03
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Hi, 

I reviewed the draft and have the following comments most are editorial

 

1.       The draft should have an author name at the top, not just SPIRIT
SDP, there is an author name at the end Sergey Ikonin.

2.       You describe the payload but it is not clear to me how do you know
the size of each speech frame. It may be good to explain it if you can.

3.       In section 3 RTP payload format you should start with a section on
RTP header usage. See for example RFC 5404.

4.       In the security consideration you can use the first two paragraph
from RFC 5404 security section.

5.       You can add a section on congestion control, this codec can handle
congestion so it will be good to mention it.

6.       You can add to section 5.2 a subsection on offer answer
considerstions

 

Some nits

 

1.       In section 2  "This making the bit stream scalable and allows
reduce ."

2.       In section 2 " But because of the scalable nature of IP-MR codec
there is no need to duplicate the whole previous frame - only the core layer
may be retransmitted."

3.       In section 3.2 "BR (3 bits): base rate for core layer of frame(s)
in this packet using the table for CR. "

 

 

Thanks

Roni Even

 

 

 

 

 

 


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<p class=3DMsoNormal>Hi, <o:p></o:p></p>

<p class=3DMsoNormal>I reviewed the draft and have the following =
comments most
are editorial<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>The draft should have an =
author
name at the top, not just SPIRIT SDP, there is an author name at the end =
Sergey
Ikonin.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>You describe the payload =
but it is
not clear to me how do you know the size of each speech frame. It may be =
good
to explain it if you can.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 3 RTP payload =
format
you should start with a section on RTP header usage. See for example RFC =
5404.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>4.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In the security =
consideration you
can use the first two paragraph from RFC 5404 security =
section.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>5.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>You can add a section on
congestion control, this codec can handle congestion so it will be good =
to
mention it.<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l1 =
level1 lfo1'><![if !supportLists]><span
style=3D'mso-list:Ignore'>6.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>You can add to section =
5.2 a
subsection on offer answer considerstions<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Some nits<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>1.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 2 =
&nbsp;&quot;This
making <u>the</u> bit stream scalable and allows reduce =
&#8230;&quot;<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>2.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 2 &quot; But =
because of
<u>the</u> scalable nature of IP-MR codec there is no need to duplicate =
<u>the</u>
whole previous frame - only <u>the</u> core layer may be =
retransmitted.&quot;<o:p></o:p></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo2'><![if !supportLists]><span
style=3D'mso-list:Ignore'>3.<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]><span dir=3DLTR></span>In section 3.2 &quot;BR =
(3 bits):
base rate for core layer of frame(s) in this packet <u>using the table =
for CR</u>.
&quot;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal>Thanks<o:p></o:p></p>

<p class=3DMsoNormal>Roni Even<o:p></o:p></p>

<p class=3DMsoNormal>&nbsp;<o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

</div>

</body>

</html>

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From wwwrun@core3.amsl.com  Mon Apr 27 11:42:15 2009
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From: IETF Secretariat <ietf-ipr@ietf.org>
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Date: Mon, 27 Apr 2009 11:42:15 -0700 (PDT)
Cc: fluffy@cisco.com, tom.taylor@rogers.com, avt@ietf.org, ipr-announce@ietf.org
Subject: [AVT] Posting of IPR Disclosure related to ipr-fraunhofer-gesellschaft-draft-ietf-avt-rapid-rtp-sync-00, draft-schierl-avt-rtp-ntp-for-layered-codecs-00 and draft-perkins-avt-rapid-rtp-sync-02
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Dear Colin Perkins, Thomas Schierl:

An IPR disclosure that pertains to your Internet-Draft entitled "Rapid
Synchronisation of RTP Flows" (draft-ietf-avt-rapid-rtp-sync) was submitted to
the IETF Secretariat on 2009-04-22 and has been posted on the "IETF Page of
Intellectual Property Rights Disclosures"
(https://datatracker.ietf.org/ipr/1140/). The title of the IPR disclosure is
"ipr-fraunhofer-gesellschaft-draft-ietf-avt-rapid-rtp-sync-00,
draft-schierl-avt-rtp-ntp-for-layered-codecs-00 and
draft-perkins-avt-rapid-rtp-sync-02."

The IETF Secretariat



From ron.even.tlv@gmail.com  Mon Apr 27 14:02:33 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'VAN CAENEGEM Tom'" <Tom.Van_Caenegem@alcatel-lucent.be>, <avt@ietf.org>
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Cc: mmusic@ietf.org
Subject: Re: [AVT] Multiple Feedback Targets in "	http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18"
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Tom,

I think this question should go to mmusic. What you are trying to do here is
to define semantics for the content of a RTCP flow pointing to different
devices. The m-line or the RTCP line in SDP does not address what is the
content of the stream that will go in the RTP session.  RFC 4796 addresses
the media content issue but you are looking for something similar to the
RTCP attribute. This may require a new parameter to the rtcp attribute. 

You will also need to look at backward interoperability where the receiver
will not understand this semantics and may change the RTCP message to the
wrong address. 

 

The example for the content attribute was to specify to video m-lines one
with screen capture as a video stream and the other from a camera. SDP does
not define the behavior of the receiver if it receives two m-lines of the
same media type and can only support one. It is not clear which one he will
chose (first or second). The work on capability negotiation tries to address
this issue.

 

Roni Even

 

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of VAN
CAENEGEM Tom
Sent: Friday, April 24, 2009 1:36 PM
To: avt@ietf.org
Subject: [AVT] Multiple Feedback Targets in "
http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18"

 

 

Hi, 

I have two questions/comments mostly regarding the RTCP-SSM draft (
<http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18>
http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18). The background of my
questions is the use case of "broadcast channels" in an IPTV context, where
the channels are distributed over IP multicast (SSM). In this context, the
concept of feedback target(s) is obviously very useful, and actually
necessary given today's restrictions with multicast in upstream in the
"WAN". 

For the IPTV use case, it also makes sense to have multiple feedback
targets, each responsible for a subset of RTP receivers, because of the
large number of RTP receivers involved (100.000s is possible). The way to
signal this by means of SDP is that each local subset of RTP receivers sees
a different IP address in the "a=rtcp: "portnumber" IN IP4 "IP address"
line.  So, multiple "localised" FT  is covered by the draft IMO.

However, in that case, the maximum bandwidth that can be used for RTCP
reporting by RTP receivers only needs to be determined by the size of the
local group of RTP receivers (sending RTCP to the same localised Feedback
Target), especially in the summary FB model. But in my interpretation of the
architecture, the actual RTCP bandwith limit information can only be
retrieved by the RTP receivers from the information sent by the distribution
source in the ssm session. Either by refected RTCP reports from RTP
receivers (not a feasible solution for the given use case) or from a RSI
message -in the summary FB model- indicating the session size or explicitly
the RTCP reporting bandwidth limit. The fact is that this information is
sent to ALL RTP receivers, which may be reporting to different Feedback
targets where each Feedback target is collecting RTCP messages from a
different number of RTP receivers, and hence may have different
"requirements" for total RTCP reporting bandwidth limit.  My question is
whether local Feedback targets have been considered with respect to this
issue?

My second question again has to do with multiple feedback targets for the
same use case. 

In the IPTV use case for rtcpssm, Feedback Targets can also be used as
retransmission sources (RFC 4588) and in
<http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp
-03>
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp-
03, the feedback target architecture is used to allow fast acquisition of
SSM RTP sessions. The draft proposes that  a "burst source" is co-located
with the FT which transmits unicast data (starting with a random access
point of the video channel) from a cache that holds the most recent data
distributed over the RTP SSM. It does transmit the RTP packets at an
accelerated speed, allowing the RTP receiver to join the SSM after some
time.

The idea I would like to bring forward is to allow separate Feedback Targets
for the same RTP receiver (population): one Feedback target acts as RTP
retransmission server (see
<http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp
-03>
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp-
03) and collects RTCP FB messages, while the other Feedback Target is solely
"responsible" for the non RTCP-FB messages, such as RTCP XR messages. I
realise this idea is not necessarily linked with  the rtcp-ssm draft, but
IMO only for a massive scale application that makes use of RTP and leverages
rtcp ssm such a separation is justified. The possibility for separation of
the monitoring service -associated with a server function basically
responsible for collecting RTCP receiver reports for the purpose of
monitoring- and the retransmission service requiring a server function that
is responsible for processing and responding to RTCP FB messages, is driven
from the point of view of server function optimisation (different
requirements on the server platform). 

In terms of signaling two Feedback Targets, we could make use of the
following lines: 

a=rtcp:40001 IN IP4 192.0.2.1 (This is for retransmission) 
a=rtcp-fb:96 nack 
a=rtcp:40001 IN IP4 192.0.2.2 (This is for XR collection) 

 

However, that is probably quite ambiguous..? 
Furthermore I also realise such a solution introduces other questions than
just signaling syntax including the fact that we may need guiding rules on
how RTCP reports are issued by the RTP receiver with two RTCP targets in
place, as in the example above. The new RFC 5506 is certainly helpful here,
but this is probably not sufficient. 

Comments appreciated on this idea. 

Kind regards, 
Tom Van Caenegem 

 

  


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Tom,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I think this question should go to mmusic. What you are =
trying
to do here is to define semantics for the content of a RTCP flow =
pointing to
different devices. The m-line or the RTCP line in SDP does not address =
what is
the content of the stream that will go in the RTP session. &nbsp;RFC =
4796
addresses the media content issue but you are looking for something =
similar to
the RTCP attribute. This may require a new parameter to the rtcp =
attribute. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>You will also need to look at backward interoperability =
where
the receiver will not understand this semantics and may change the RTCP =
message
to the wrong address. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>The example for the content attribute was to specify to =
video
m-lines one with screen capture as a video stream and the other from a =
camera.
SDP does not define the behavior of the receiver if it receives two =
m-lines of
the same media type and can only support one. It is not clear which one =
he will
chose (first or second). The work on capability negotiation tries to =
address
this issue.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of =
</b>VAN
CAENEGEM Tom<br>
<b>Sent:</b> Friday, April 24, 2009 1:36 PM<br>
<b>To:</b> avt@ietf.org<br>
<b>Subject:</b> [AVT] Multiple Feedback Targets in &quot;
http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18&quot;<o:p></o:p></sp=
an></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Hi,</span> =
<o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I =
have two
questions/comments mostly regarding the RTCP-SSM draft (</span><a
href=3D"http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18"><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>http://tools.=
ietf.org/html/draft-ietf-avt-rtcpssm-18</span></a><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>). The =
background of
my questions is the use case of &quot;broadcast channels&quot; in an =
IPTV
context, where the channels are distributed over IP multicast (SSM). In =
this
context, the concept of feedback target(s) is obviously very useful, and
actually necessary given today's restrictions with multicast in upstream =
in the
&quot;WAN&quot;. </span><o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>For =
the IPTV
use case, it also makes sense to have multiple feedback targets, each
responsible for a subset of RTP receivers, because of the large number =
of RTP
receivers involved (100.000s is possible). The way to signal this by =
means of
SDP is that each local subset of RTP receivers sees a different IP =
address in
the &quot;</span><span lang=3DNL-BE =
style=3D'font-size:10.0pt;font-family:"Courier New"'>a=3Drtcp:
&quot;portnumber&quot; IN IP4 &quot;IP address&quot;&nbsp;</span><span
lang=3DNL-BE> </span><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>line.&nbsp;
So, multiple &quot;localised&quot; FT&nbsp; is covered by the draft =
IMO.</span><o:p></o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>However, in
that case, the maximum bandwidth that can be used for RTCP reporting by =
RTP receivers
only needs to be determined by the size of the local group of RTP =
receivers
(sending RTCP to the same localised Feedback Target), especially in the =
summary
FB model. But in my interpretation of the architecture, the actual RTCP
bandwith limit information can only be retrieved by the RTP receivers =
from the
information sent by the distribution source in the ssm session. Either =
by
refected RTCP reports from RTP receivers (not a feasible solution for =
the given
use case) or from a RSI message -in the summary FB model- indicating the
session size or explicitly the RTCP reporting bandwidth limit. The fact =
is that
this information is sent to ALL RTP receivers, which may be reporting to
different Feedback targets where each Feedback target is collecting RTCP =
messages
from a different number of RTP receivers, and hence may have different
&quot;requirements&quot; for total RTCP reporting bandwidth limit.&nbsp; =
My
question is whether local Feedback targets have been considered with =
respect to
this issue?</span><o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>My =
second
question again has to do with multiple feedback targets for the same use =
case. </span><o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In =
the IPTV
use case for rtcpssm, Feedback Targets can also be used as =
retransmission
sources (RFC 4588) and in </span><a
href=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizati=
on-for-rtp-03"><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>http://tools.=
ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp-03</span><=
/a><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>, the =
feedback target
architecture is used to allow fast acquisition of SSM RTP sessions. The =
draft
proposes that&nbsp; a &quot;burst source&quot; is co-located with the FT =
which
transmits unicast data (starting with a random access point of the video
channel) from a cache that holds the most recent data distributed over =
the RTP
SSM. It does transmit the RTP packets at an accelerated speed, allowing =
the RTP
receiver to join the SSM after some time.</span><o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>The =
idea I
would like to bring forward is to allow separate Feedback Targets&nbsp; =
for the
same RTP receiver (population): one Feedback target acts as RTP =
retransmission
server (see </span><a
href=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizati=
on-for-rtp-03"><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>http://tools.=
ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp-03</span><=
/a><span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>) and =
collects RTCP
FB messages, while the other Feedback Target is solely =
&quot;responsible&quot;
for the non RTCP-FB messages, such as RTCP XR messages. I realise this =
idea is
not necessarily linked with&nbsp; the rtcp-ssm draft, but IMO only for a
massive scale application that makes use of RTP and leverages rtcp ssm =
such a
separation is justified. The possibility for separation of the =
monitoring service</span>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&#8211;associ=
ated
with a server function basically responsible for collecting RTCP =
receiver
reports for the purpose of monitoring- and the retransmission service =
requiring
a server function that is responsible for processing and responding to =
RTCP FB
messages, is driven from the point of view of server function =
optimisation
(different requirements on the server platform). </span><o:p></o:p></p>

<p><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>In =
terms of
signaling two Feedback Targets, we could make use of the following =
lines:</span>
<o:p></o:p></p>

<p><span lang=3DNL-BE style=3D'font-size:10.0pt;font-family:"Courier =
New"'>a=3Drtcp:40001
IN IP4 192.0.2.1 (This is for retransmission)</span> <br>
<span lang=3DNL-BE style=3D'font-size:10.0pt;font-family:"Courier =
New"'>a=3Drtcp-fb:96
nack</span> <br>
<span lang=3DNL-BE style=3D'font-size:10.0pt;font-family:"Courier =
New"'>a=3Drtcp:40001
IN IP4 192.0.2.2 (This is for XR collection)</span><span lang=3DNL-BE> =
</span><o:p></o:p></p>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>However,
that is probably quite ambiguous..?</span> <br>
<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Furthermore =
I
also realise such a solution introduces other questions than just =
signaling
syntax including the fact that we may need guiding rules on how RTCP =
reports
are issued by the RTP receiver with two RTCP targets in place, as in the
example above. The new RFC 5506 is certainly helpful here, but this is =
probably
not sufficient. </span><o:p></o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Comments
appreciated on this idea.</span> <o:p></o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Kind
regards,</span> <br>
<span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Tom =
Van
Caenegem</span> <o:p></o:p></p>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p>

<p><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span>=

<o:p></o:p></p>

</div>

</body>

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On 25 Feb 2009, at 19:45, Internet-Drafts@ietf.org wrote:
> 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           : RTCP XR Report Block for Measurement Identity
> 	Author(s)       : G. Hunt, A. Clark
> 	Filename        : draft-ietf-avt-rtcp-xr-meas-identity-01.txt
> 	Pages           : 13
> 	Date            : 2009-02-25
>
> This document defines an RTCP XR Report Block carrying parameters
> which identify a measurement, to which one or more other RTCP XR
> Report Blocks may refer.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-xr-meas-identity-01.txt

The changes look good - thanks!

-- 
Colin Perkins
http://csperkins.org/




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On 25 Feb 2009, at 19:45, Internet-Drafts@ietf.org wrote:
> 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           : RTCP XR Report Block for Packet Delay Variation  
> Metric Reporting
> 	Author(s)       : G. Hunt, A. Clark
> 	Filename        : draft-ietf-avt-rtcp-xr-pdv-01.txt
> 	Pages           : 15
> 	Date            : 2009-02-25
>
> This document defines an RTCP XR Report Block that allows the
> reporting of Packet Delay Variation metrics for a range of RTP
> applications.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-xr-pdv-01.txt


Changes here look good too, thanks.

-- 
Colin Perkins
http://csperkins.org/




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Subject: Re: [AVT] SDP attribute in rtcpssm-18
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Marc Petit-Huguenin wrote:
> Section 10.1 of draft-ietf-avt-rtcpssm-18 states that '[a] new
> session level attribute MUST be used [...]: "rtcp-unicast"'  but the
>  IANA considerations section registers a media level only att-field
> named "unicast-rtcp".  So which one should be used, and is it a
> media or a session level attribute?

Has anybody implemented this draft?

-- 
Marc Petit-Huguenin
Home: marc@petit-huguenin.org
Work: petithug@acm.org

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This revised draft addresses Al Morton's comment at IETF 74, concerning
aligning the descriptions of packet delay variation metrics with the
terminology used in RFC 5481. Review comments on this and any others of
the Clark/Hunt -rtcp-xr-* drafts will be very welcome!

Thanks

Geoff

Geoff Hunt=20
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-----Original Message-----
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Sent: 28 April 2009 17:37
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Subject: New Version Notification for draft-ietf-avt-rtcp-xr-pdv-02=20


A new version of I-D, draft-ietf-avt-rtcp-xr-pdv-02.txt has been
successfuly submitted by Geoff Hunt and posted to the IETF repository.

Filename:	 draft-ietf-avt-rtcp-xr-pdv
Revision:	 02
Title:		 RTCP XR Report Block for Packet Delay Variation Metric
Reporting
Creation_date:	 2009-04-28
WG ID:		 avt
Number_of_pages: 15

Abstract:
This document defines an RTCP XR Report Block that allows the reporting
of Packet Delay Variation metrics for a range of RTP applications.
=20



The IETF Secretariat.



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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           : RTCP XR Report Block for Packet Delay Variation Metric Reporting
	Author(s)       : G. Hunt, A. Clark
	Filename        : draft-ietf-avt-rtcp-xr-pdv-02.txt
	Pages           : 15
	Date            : 2009-04-28

This document defines an RTCP XR Report Block that allows the
reporting of Packet Delay Variation metrics for a range of RTP
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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 format for G.718 speech/audio
	Author(s)       : A. Lakaniemi, Y. Wang
	Filename        : draft-ietf-avt-rtp-g718-01.txt
	Pages           : 31
	Date            : 2009-04-28

This document specifies the Real-Time Transport Protocol (RTP) 
payload format for the Embedded Variable Bit-Rate (EV-VBR) 
speech/audio codec, specified in ITU-T G.718. A media type 
registration for this RTP payload format is also included. 

Conventions used in this document 

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From yekuiwang@huawei.com  Tue Apr 28 10:10:01 2009
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Date: Tue, 28 Apr 2009 13:11:11 -0400
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This is just a keep-alive submission with only updates to my
affiliation/address and the boiler template.  

BR, YK

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On 
> Behalf Of Internet-Drafts@ietf.org
> Sent: Tuesday, April 28, 2009 12:45 PM
> To: i-d-announce@ietf.org
> Cc: avt@ietf.org
> Subject: [AVT] I-D Action:draft-ietf-avt-rtp-g718-01.txt
> 
> 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 format for G.718 speech/audio
> 	Author(s)       : A. Lakaniemi, Y. Wang
> 	Filename        : draft-ietf-avt-rtp-g718-01.txt
> 	Pages           : 31
> 	Date            : 2009-04-28
> 
> This document specifies the Real-Time Transport Protocol 
> (RTP) payload format for the Embedded Variable Bit-Rate 
> (EV-VBR) speech/audio codec, specified in ITU-T G.718. A 
> media type registration for this RTP payload format is also included. 
> 
> Conventions used in this document 
> 
> The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL 
> NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and 
> "OPTIONAL" in this document are to be interpreted as 
> described in RFC 2119 [RFC2119].
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-g718-01.txt
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> Below is the data which will enable a MIME compliant mail 
> reader implementation to automatically retrieve the ASCII 
> version of the Internet-Draft.
> 



From timur.friedman@gmail.com  Tue Apr 28 12:15:32 2009
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From: Timur Friedman <timur.friedman@upmc.fr>
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Subject: Re: [AVT] misleading RFC 3611 statement on RTCP packet stacking
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To follow up on our proposal to file an RFC Errata report, here, for
more direct comparison, is the RFC 3611 text immediately followed by
our suggestion for how it should be read:

> "XR packets supplement the existing RTCP packets, and may be stacked with
> other RTCP packets to form compound RTCP packets [9, Section 6]."

> =93XR packets supplement the existing RTCP packets, and stack with other =
RTCP
> packets to form compound RTCP packets, as described in Sec. 6 of RFC 3550
> [9].=94

Timur


On Wed, Apr 22, 2009 at 8:31 PM, Timur Friedman <timur.friedman@upmc.fr> wr=
ote:
> Some people are reading RFC 3611 as overriding RFC 3550 on RTCP packet
> stacking.
>
> This is what Sec. 1.1 of RFC 3611 says:
>
> "XR packets supplement the existing RTCP packets, and may be stacked with
> other RTCP packets to form compound RTCP packets [9, Section 6]."
>
> Reference 9 is RFC 3550, which says:
>
> "all RTCP packets MUST be sent in a compound packet of at least two
> individual packets"
>
> That is, RFC 3550 says that RTCP packets MUST be stacked into a compound
> packet, and some people are reading RFC 3611 as weakening that mandate to
> say only that RTCP packets MAY be stacked into a compound packet (but als=
o
> MAY NOT, at least in the case of the XR packet).
>
> As one of the authors of RFC 3611, I can say that it was not my intention=
 to
> override RFC 3550. Nor was it the intention of the two of my eight
> co-authors who I have discussed this with. Rather, the purpose of this
> sentence was to remind the reader that XR packets, like any RTCP packets,
> are designed to be stacked, and to helpfully point the reader to the RFC
> 3550 section that deals with RTCP packet stacking for further details.
>
> To the best of my recollection, no discussion in AVT leading up to approv=
al
> of RFC 3611 indicates that any group participants at the time believed th=
at
> RFC 3611 would override RFC 3550 on RTCP packet stacking.
>
> Still, I understand how people are reading RFC 3611 in this way. It does =
say
> "may be stacked", and though "may" is not capitalized, it is nonetheless =
an
> RFC 2119 keyword, meaning that the item is truly optional.
>
> Co-authors Nick Duffield, Ramon Caceres, and myself would like to open
> discussion on how to address this situation. We start by suggesting that =
we
> file an RFC Errata report, indicating that the sentence should read inste=
ad:
>
> =93XR packets supplement the existing RTCP packets, and stack with other =
RTCP
> packets to form compound RTCP packets, as described in Sec. 6 of RFC 3550
> [9].=94
>
> By dropping the "may", we drop any hint that this RFC overrides the RFC 3=
550
> stacking mandate. And by adding "as described in Sec. 6 of RFC 3550" we m=
ake
> clear the informational nature of this sentence.
>
> Timur Friedman
>
>

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Subject: [AVT] Fwd: Last Call: draft-levine-rfb (The Remote Framebuffer Protocol) to Informational RFC
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Just as FYI, this draft is in IETF LC. It specifies the protocol used  
by VNC to share for desktop sharing.

Begin forwarded message:

> From: The IESG <iesg-secretary@ietf.org>
> Date: April 28, 2009 8:05:24 AM MDT (CA)
> To: IETF-Announce <ietf-announce@ietf.org>
> Subject: Last Call: draft-levine-rfb (The Remote Framebuffer  
> Protocol)  to Informational RFC
> Reply-To: ietf@ietf.org
>
> The IESG has received a request from an individual submitter to  
> consider
> the following document:
>
> - 'The Remote Framebuffer Protocol '
>   <draft-levine-rfb-01.txt> as an Informational RFC
>
> The IESG plans to make a decision in the next few weeks, and solicits
> final comments on this action.  Please send substantive comments to  
> the
> ietf@ietf.org mailing lists by 2009-05-26. Exceptionally,
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> The file can be obtained via
> http://www.ietf.org/internet-drafts/draft-levine-rfb-01.txt
>
>
> IESG discussion can be tracked via
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>
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From csp@csperkins.org  Wed Apr 29 10:41:11 2009
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Subject: Re: [AVT] SDP example in draft-versteeg-avt-rapid-synchronization-for-rtp-03
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Roni,

It's been a while since I read SDP and its various extensions in great  
detail, but as far as I can tell, this SDP does provide sufficient  
information to join the session, on the assumption that symmetric RTP/ 
RTCP ports can be used.

Colin



On 26 Apr 2009, at 19:48, Roni Even wrote:
> I am looking for more feedback on the following topic (not by Roni  
> or Ali).
> The example in the draft is coming from a distribution source using  
> for example SAP (multicast distribution).
> There are three entities in the communication. The multicast  
> distribution source, the unicast retransmission which also serves as  
> a feedback target and a client that wants to receive the multicast  
> stream and will ask for unicast  resynchronization or retransmission.
>
> This SDP is coming from the distribution source and according to the  
> authors specify the multicast address, the unicast transport address  
> for the RTCP receiver reports for the multicast (SSM), the uincast  
> transport address on the feedback target/retransmission server that  
> will  be used by the clients to ask for retransmission streams using  
> RTCP-FB and the transport address on the client that will be used by  
> the feedback target to send the unicast RTP stream to.
>
> Does this SDP look like it supplies all this information enabling  
> interoperable applications.
>
>          v=0
>         o=ali 1122334455 1122334466 IN IP4 rams.example.com
>         s=Rapid Acquisition Examples
>         t=0 0
>         a=group:FID 1 2
>         a=group:FID 3 4
>         a=rtcp-unicast:rsi
>         m=video 40000 RTP/AVPF 96
>         i=Primary Multicast Stream #1
>         c=IN IP4 224.1.1.1/255
>         a=source-filter: incl IN IP4 224.1.1.1 192.0.2.2
>         a=recvonly
>         a=rtpmap:96 MP2T/90000
>         a=rtcp:40001 IN IP4 192.0.2.1
>         a=rtcp-fb:96 nack
>         a=mid:1
>         m=video 40002 RTP/AVPF 97
>         i=Unicast Retransmission Stream #1 (Ret. Support Only)
>         c=IN IP4 192.0.2.1
>         a=recvonly
>         a=rtpmap:97 rtx/90000
>         a=rtcp:40003
>         a=fmtp:97 apt=96
>         a=fmtp:97 rtx-time=3000
>         a=mid:2
>
>
> Thanks
> Roni Even


-- 
Colin Perkins
http://csperkins.org/




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Subject: Re: [AVT] Multiple Feedback Targets in "	http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18"
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Roni,

The signalling aspects could certainly go to MMUSIC, and are probably =20=

not too difficult to address. The proposal also involves some =20
potentially significant changes to the RTCP architecture, which should =20=

be considered in AVT (although hopefully as additional documents =20
rather than being folded into the - already long overdue - rtcpssm =20
draft).

Cheers,
Colin



On 27 Apr 2009, at 22:01, Roni Even wrote:
> Tom,
> I think this question should go to mmusic. What you are trying to do =20=

> here is to define semantics for the content of a RTCP flow pointing =20=

> to different devices. The m-line or the RTCP line in SDP does not =20
> address what is the content of the stream that will go in the RTP =20
> session.  RFC 4796 addresses the media content issue but you are =20
> looking for something similar to the RTCP attribute. This may =20
> require a new parameter to the rtcp attribute.
> You will also need to look at backward interoperability where the =20
> receiver will not understand this semantics and may change the RTCP =20=

> message to the wrong address.
>
> The example for the content attribute was to specify to video m-=20
> lines one with screen capture as a video stream and the other from a =20=

> camera. SDP does not define the behavior of the receiver if it =20
> receives two m-lines of the same media type and can only support =20
> one. It is not clear which one he will chose (first or second). The =20=

> work on capability negotiation tries to address this issue.
>
> Roni Even
>
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf =20
> Of VAN CAENEGEM Tom
> Sent: Friday, April 24, 2009 1:36 PM
> To: avt@ietf.org
> Subject: [AVT] Multiple Feedback Targets in " =
http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18=20
> "
>
>
> Hi,
>
> I have two questions/comments mostly regarding the RTCP-SSM draft =
(http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18=20
> ). The background of my questions is the use case of "broadcast =20
> channels" in an IPTV context, where the channels are distributed =20
> over IP multicast (SSM). In this context, the concept of feedback =20
> target(s) is obviously very useful, and actually necessary given =20
> today's restrictions with multicast in upstream in the "WAN".
>
> For the IPTV use case, it also makes sense to have multiple feedback =20=

> targets, each responsible for a subset of RTP receivers, because of =20=

> the large number of RTP receivers involved (100.000s is possible). =20
> The way to signal this by means of SDP is that each local subset of =20=

> RTP receivers sees a different IP address in the "a=3Drtcp: =20
> "portnumber" IN IP4 "IP address"  line.  So, multiple "localised" =20
> FT  is covered by the draft IMO.
>
> However, in that case, the maximum bandwidth that can be used for =20
> RTCP reporting by RTP receivers only needs to be determined by the =20
> size of the local group of RTP receivers (sending RTCP to the same =20
> localised Feedback Target), especially in the summary FB model. But =20=

> in my interpretation of the architecture, the actual RTCP bandwith =20
> limit information can only be retrieved by the RTP receivers from =20
> the information sent by the distribution source in the ssm session. =20=

> Either by refected RTCP reports from RTP receivers (not a feasible =20
> solution for the given use case) or from a RSI message -in the =20
> summary FB model- indicating the session size or explicitly the RTCP =20=

> reporting bandwidth limit. The fact is that this information is sent =20=

> to ALL RTP receivers, which may be reporting to different Feedback =20
> targets where each Feedback target is collecting RTCP messages from =20=

> a different number of RTP receivers, and hence may have different =20
> "requirements" for total RTCP reporting bandwidth limit.  My =20
> question is whether local Feedback targets have been considered with =20=

> respect to this issue?
>
> My second question again has to do with multiple feedback targets =20
> for the same use case.
>
> In the IPTV use case for rtcpssm, Feedback Targets can also be used =20=

> as retransmission sources (RFC 4588) and in =
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rt=
p-03=20
> , the feedback target architecture is used to allow fast acquisition =20=

> of SSM RTP sessions. The draft proposes that  a "burst source" is co-=20=

> located with the FT which transmits unicast data (starting with a =20
> random access point of the video channel) from a cache that holds =20
> the most recent data distributed over the RTP SSM. It does transmit =20=

> the RTP packets at an accelerated speed, allowing the RTP receiver =20
> to join the SSM after some time.
>
> The idea I would like to bring forward is to allow separate Feedback =20=

> Targets  for the same RTP receiver (population): one Feedback target =20=

> acts as RTP retransmission server =
(seehttp://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-fo=
r-rtp-03)=20
>  and collects RTCP FB messages, while the other Feedback Target is =20
> solely "responsible" for the non RTCP-FB messages, such as RTCP XR =20
> messages. I realise this idea is not necessarily linked with  the =20
> rtcp-ssm draft, but IMO only for a massive scale application that =20
> makes use of RTP and leverages rtcp ssm such a separation is =20
> justified. The possibility for separation of the monitoring service =96=20=

> associated with a server function basically responsible for =20
> collecting RTCP receiver reports for the purpose of monitoring- and =20=

> the retransmission service requiring a server function that is =20
> responsible for processing and responding to RTCP FB messages, is =20
> driven from the point of view of server function optimisation =20
> (different requirements on the server platform).
>
> In terms of signaling two Feedback Targets, we could make use of the =20=

> following lines:
>
> a=3Drtcp:40001 IN IP4 192.0.2.1 (This is for retransmission)
> a=3Drtcp-fb:96 nack
> a=3Drtcp:40001 IN IP4 192.0.2.2 (This is for XR collection)
>
>
> However, that is probably quite ambiguous..?
> Furthermore I also realise such a solution introduces other =20
> questions than just signaling syntax including the fact that we may =20=

> need guiding rules on how RTCP reports are issued by the RTP =20
> receiver with two RTCP targets in place, as in the example above. =20
> The new RFC 5506 is certainly helpful here, but this is probably not =20=

> sufficient.
>
> Comments appreciated on this idea.
>
> Kind regards,
> Tom Van Caenegem
>
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




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<html><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; " class=3D"">Roni,<div class=3D"" =
style=3D""><br class=3D"" style=3D""></div><div class=3D"" style=3D"">The =
signalling aspects could certainly go to MMUSIC, and are probably not =
too difficult to address. The proposal also involves some potentially =
significant changes to the RTCP architecture, which should be considered =
in AVT (although hopefully as additional documents rather than being =
folded into the - already long overdue - rtcpssm draft).&nbsp;</div><div =
class=3D"" style=3D""><br class=3D"" style=3D""></div><div class=3D"" =
style=3D"">Cheers,</div><div class=3D"" style=3D"">Colin</div><div =
class=3D"" style=3D""><br class=3D"" style=3D""></div><div class=3D"" =
style=3D""><br class=3D"" style=3D""></div><div class=3D"" style=3D""><br =
class=3D"" style=3D""></div><div style=3D""><div class=3D"" style=3D"">On =
27 Apr 2009, at 22:01, Roni Even wrote:</div><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: auto; text-indent: 0px; text-transform: none; white-space: =
normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple" class=3D"" style=3D""><div class=3D"Section1" =
style=3D""><div class=3D"" style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: =
'Times New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); =
">Tom,<o:p></o:p></span></div><div class=3D"" style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">I =
think this question should go to mmusic. What you are trying to do here =
is to define semantics for the content of a RTCP flow pointing to =
different devices. The m-line or the RTCP line in SDP does not address =
what is the content of the stream that will go in the RTP session. =
&nbsp;RFC 4796 addresses the media content issue but you are looking for =
something similar to the RTCP attribute. This may require a new =
parameter to the rtcp attribute.<o:p></o:p></span></div><div class=3D"" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">You will also need to look at =
backward interoperability where the receiver will not understand this =
semantics and may change the RTCP message to the wrong =
address.<o:p></o:p></span></div><div class=3D"" style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div class=3D"" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">The example for the content =
attribute was to specify to video m-lines one with screen capture as a =
video stream and the other from a camera. SDP does not define the =
behavior of the receiver if it receives two m-lines of the same media =
type and can only support one. It is not clear which one he will chose =
(first or second). The work on capability negotiation tries to address =
this issue.<o:p></o:p></span></div><div class=3D"" style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div class=3D"" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Roni =
Even<o:p></o:p></span></div><div class=3D"" style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div class=3D"" style=3D""><div class=3D""=
 style=3D"border-right-style: none; border-bottom-style: none; =
border-left-style: none; border-width: initial; border-color: initial; =
border-top-style: solid; border-top-color: rgb(181, 196, 223); =
border-top-width: 1pt; padding-top: 3pt; padding-right: 0in; =
padding-bottom: 0in; padding-left: 0in; "><div class=3D"" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><b><span style=3D"font-size: 10pt; font-family: Tahoma, =
sans-serif; ">From:</span></b><span style=3D"font-size: 10pt; =
font-family: Tahoma, sans-serif; "><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt-bounces@ietf.org">avt-bounces@ietf.org</a> [<a =
href=3D"mailto:avt-bounces@ietf.org" style=3D"color: blue; =
text-decoration: underline; ">mailto:avt-bounces@ietf.org</a>]<span =
class=3D"Apple-converted-space">&nbsp;</span><b>On Behalf Of<span =
class=3D"Apple-converted-space">&nbsp;</span></b>VAN CAENEGEM Tom<br =
class=3D"" style=3D""><b>Sent:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>Friday, April 24, 2009 1:36 =
PM<br class=3D"" style=3D""><b>To:</b><span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"mailto:avt@ietf.org" style=3D"color: blue; text-decoration: =
underline; ">avt@ietf.org</a><br class=3D"" =
style=3D""><b>Subject:</b><span =
class=3D"Apple-converted-space">&nbsp;</span>[AVT] Multiple Feedback =
Targets in "<span class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18" =
style=3D"color: blue; text-decoration: underline; =
">http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18</a>"<o:p></o:p></sp=
an></div></div></div><div class=3D"" style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div><div class=3D"" style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><o:p>&nbsp;</o:p></div><p =
align=3D"" class=3D"" style=3D"margin-right: 0in; margin-left: 0in; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">Hi,</span><o:p></o:p></p><p align=3D"" class=3D"" style=3D"margin-right:=
 0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">I have two questions/comments mostly regarding the RTCP-SSM draft =
(</span><a href=3D"http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18" =
style=3D"color: blue; text-decoration: underline; "><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">http://tools.ietf.org/html/draft-ietf-avt-rtcpssm-18</span></a><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; ">). The =
background of my questions is the use case of "broadcast channels" in an =
IPTV context, where the channels are distributed over IP multicast =
(SSM). In this context, the concept of feedback target(s) is obviously =
very useful, and actually necessary given today's restrictions with =
multicast in upstream in the "WAN".</span><o:p></o:p></p><p align=3D"" =
class=3D"" style=3D"margin-right: 0in; margin-left: 0in; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; ">For the IPTV use case, it also =
makes sense to have multiple feedback targets, each responsible for a =
subset of RTP receivers, because of the large number of RTP receivers =
involved (100.000s is possible). The way to signal this by means of SDP =
is that each local subset of RTP receivers sees a different IP address =
in the "</span><span lang=3D"NL-BE" style=3D"font-size: 10pt; =
font-family: 'Courier New'; ">a=3Drtcp: "portnumber" IN IP4 "IP =
address"&nbsp;</span><span lang=3D"NL-BE"><span =
class=3D"Apple-converted-space">&nbsp;</span></span><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; ">line.&nbsp; =
So, multiple "localised" FT&nbsp; is covered by the draft =
IMO.</span><o:p></o:p></p><p align=3D"" class=3D"" style=3D"margin-right: =
0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">However, in that case, the maximum bandwidth that can be used for RTCP =
reporting by RTP receivers only needs to be determined by the size of =
the local group of RTP receivers (sending RTCP to the same localised =
Feedback Target), especially in the summary FB model. But in my =
interpretation of the architecture, the actual RTCP bandwith limit =
information can only be retrieved by the RTP receivers from the =
information sent by the distribution source in the ssm session. Either =
by refected RTCP reports from RTP receivers (not a feasible solution for =
the given use case) or from a RSI message -in the summary FB model- =
indicating the session size or explicitly the RTCP reporting bandwidth =
limit. The fact is that this information is sent to ALL RTP receivers, =
which may be reporting to different Feedback targets where each Feedback =
target is collecting RTCP messages from a different number of RTP =
receivers, and hence may have different "requirements" for total RTCP =
reporting bandwidth limit.&nbsp; My question is whether local Feedback =
targets have been considered with respect to this =
issue?</span><o:p></o:p></p><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; ">My second question again has to do =
with multiple feedback targets for the same use =
case.</span><o:p></o:p></p><p align=3D"" class=3D"" style=3D"margin-right:=
 0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">In the IPTV use case for rtcpssm, Feedback Targets can also be used as =
retransmission sources (RFC 4588) and in<span =
class=3D"Apple-converted-space">&nbsp;</span></span><a =
href=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizatio=
n-for-rtp-03" style=3D"color: blue; text-decoration: underline; "><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-=
rtp-03</span></a><span style=3D"font-size: 10pt; font-family: Arial, =
sans-serif; ">, the feedback target architecture is used to allow fast =
acquisition of SSM RTP sessions. The draft proposes that&nbsp; a "burst =
source" is co-located with the FT which transmits unicast data (starting =
with a random access point of the video channel) from a cache that holds =
the most recent data distributed over the RTP SSM. It does transmit the =
RTP packets at an accelerated speed, allowing the RTP receiver to join =
the SSM after some time.</span><o:p></o:p></p><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; ">The idea I would like to bring forward =
is to allow separate Feedback Targets&nbsp; for the same RTP receiver =
(population): one Feedback target acts as RTP retransmission server =
(see</span><a =
href=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizatio=
n-for-rtp-03" style=3D"color: blue; text-decoration: underline; "><span =
style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-=
rtp-03</span></a><span style=3D"font-size: 10pt; font-family: Arial, =
sans-serif; ">) and collects RTCP FB messages, while the other Feedback =
Target is solely "responsible" for the non RTCP-FB messages, such as =
RTCP XR messages. I realise this idea is not necessarily linked =
with&nbsp; the rtcp-ssm draft, but IMO only for a massive scale =
application that makes use of RTP and leverages rtcp ssm such a =
separation is justified. The possibility for separation of the =
monitoring service</span><span =
class=3D"Apple-converted-space">&nbsp;</span><span style=3D"font-size: =
10pt; font-family: Arial, sans-serif; ">=96associated with a server =
function basically responsible for collecting RTCP receiver reports for =
the purpose of monitoring- and the retransmission service requiring a =
server function that is responsible for processing and responding to =
RTCP FB messages, is driven from the point of view of server function =
optimisation (different requirements on the server =
platform).</span><o:p></o:p></p><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; ">In terms of signaling two Feedback =
Targets, we could make use of the following =
lines:</span><o:p></o:p></p><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span lang=3D"NL-BE" =
style=3D"font-size: 10pt; font-family: 'Courier New'; ">a=3Drtcp:40001 =
IN IP4 192.0.2.1 (This is for retransmission)</span><span =
class=3D"Apple-converted-space">&nbsp;</span><br class=3D"" =
style=3D""><span lang=3D"NL-BE" style=3D"font-size: 10pt; font-family: =
'Courier New'; ">a=3Drtcp-fb:96 nack</span><span =
class=3D"Apple-converted-space">&nbsp;</span><br class=3D"" =
style=3D""><span lang=3D"NL-BE" style=3D"font-size: 10pt; font-family: =
'Courier New'; ">a=3Drtcp:40001 IN IP4 192.0.2.2 (This is for XR =
collection)</span><span lang=3D"NL-BE"></span><o:p></o:p></p><div =
class=3D"" style=3D"margin-top: 0in; margin-right: 0in; margin-left: =
0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; ">However, that is probably quite =
ambiguous..?</span><span class=3D"Apple-converted-space">&nbsp;</span><br =
class=3D"" style=3D""><span style=3D"font-size: 10pt; font-family: =
Arial, sans-serif; ">Furthermore I also realise such a solution =
introduces other questions than just signaling syntax including the fact =
that we may need guiding rules on how RTCP reports are issued by the RTP =
receiver with two RTCP targets in place, as in the example above. The =
new RFC 5506 is certainly helpful here, but this is probably not =
sufficient.</span><o:p></o:p></p><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; ">Comments appreciated on this =
idea.</span><o:p></o:p></p><p align=3D"" class=3D"" style=3D"margin-right:=
 0in; margin-left: 0in; font-size: 12pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 10pt; font-family: Arial, sans-serif; =
">Kind regards,</span><span =
class=3D"Apple-converted-space">&nbsp;</span><br class=3D"" =
style=3D""><span style=3D"font-size: 10pt; font-family: Arial, =
sans-serif; ">Tom Van Caenegem</span><o:p></o:p></p><div class=3D"" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><o:p>&nbsp;</o:p></div><p align=3D"" class=3D"" =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 12pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: 10pt; =
font-family: Arial, sans-serif; =
">&nbsp;</span><o:p></o:p></p></div>______________________________________=
_________<br class=3D"" style=3D"">Audio/Video Transport Working =
Group<br class=3D"" style=3D""><a href=3D"mailto:avt@ietf.org" =
style=3D"color: blue; text-decoration: underline; ">avt@ietf.org</a><br =
class=3D"" style=3D""><a =
href=3D"https://www.ietf.org/mailman/listinfo/avt" style=3D"color: blue; =
text-decoration: underline; =
">https://www.ietf.org/mailman/listinfo/avt</a><br class=3D"" =
style=3D""></div></span></blockquote></div><br class=3D"" style=3D""><div =
apple-content-edited=3D"true" class=3D"" style=3D""> <span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div class=3D"" style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div class=3D"" =
style=3D""><br class=3D"Apple-interchange-newline" style=3D""><br =
class=3D"khtml-block-placeholder" style=3D""></div><div class=3D"" =
style=3D"">--&nbsp;</div><div class=3D"" style=3D""></div><div class=3D"" =
style=3D"">Colin Perkins</div><div class=3D"" style=3D""><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline" style=3D""></span><br =
class=3D"Apple-interchange-newline" style=3D""></div></span> </div><br =
class=3D"" style=3D""></body></html>=

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Colin Perkins'" <csp@csperkins.org>
References: <49f4acd2.0c11660a.6549.ffffb32b@mx.google.com> <84F83FF0-B9B8-4FA6-A9F1-364782D91E8D@csperkins.org>
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Subject: Re: [AVT] SDP example in draft-versteeg-avt-rapid-synchronization-for-rtp-03
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Colin,
I agree that the assumption is that the client can use this ports but this
is why I suggest that the client should be able to override the offer if it
cannot use these ports. One example is when the client tries to join two
different multicast streams and both offer the same ports.

My other concern is that the client IP address is not known from this SDP,
so there is another assumption that the IP used in the RTCP RAMS=R from the
client is the IP address for the unicast burst.

Roni

-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org] 
Sent: Wednesday, April 29, 2009 7:34 PM
To: Roni Even
Cc: avt@ietf.org
Subject: Re: [AVT] SDP example in
draft-versteeg-avt-rapid-synchronization-for-rtp-03

Roni,

It's been a while since I read SDP and its various extensions in great  
detail, but as far as I can tell, this SDP does provide sufficient  
information to join the session, on the assumption that symmetric RTP/ 
RTCP ports can be used.

Colin



On 26 Apr 2009, at 19:48, Roni Even wrote:
> I am looking for more feedback on the following topic (not by Roni  
> or Ali).
> The example in the draft is coming from a distribution source using  
> for example SAP (multicast distribution).
> There are three entities in the communication. The multicast  
> distribution source, the unicast retransmission which also serves as  
> a feedback target and a client that wants to receive the multicast  
> stream and will ask for unicast  resynchronization or retransmission.
>
> This SDP is coming from the distribution source and according to the  
> authors specify the multicast address, the unicast transport address  
> for the RTCP receiver reports for the multicast (SSM), the uincast  
> transport address on the feedback target/retransmission server that  
> will  be used by the clients to ask for retransmission streams using  
> RTCP-FB and the transport address on the client that will be used by  
> the feedback target to send the unicast RTP stream to.
>
> Does this SDP look like it supplies all this information enabling  
> interoperable applications.
>
>          v=0
>         o=ali 1122334455 1122334466 IN IP4 rams.example.com
>         s=Rapid Acquisition Examples
>         t=0 0
>         a=group:FID 1 2
>         a=group:FID 3 4
>         a=rtcp-unicast:rsi
>         m=video 40000 RTP/AVPF 96
>         i=Primary Multicast Stream #1
>         c=IN IP4 224.1.1.1/255
>         a=source-filter: incl IN IP4 224.1.1.1 192.0.2.2
>         a=recvonly
>         a=rtpmap:96 MP2T/90000
>         a=rtcp:40001 IN IP4 192.0.2.1
>         a=rtcp-fb:96 nack
>         a=mid:1
>         m=video 40002 RTP/AVPF 97
>         i=Unicast Retransmission Stream #1 (Ret. Support Only)
>         c=IN IP4 192.0.2.1
>         a=recvonly
>         a=rtpmap:97 rtx/90000
>         a=rtcp:40003
>         a=fmtp:97 apt=96
>         a=fmtp:97 rtx-time=3000
>         a=mid:2
>
>
> Thanks
> Roni Even


-- 
Colin Perkins
http://csperkins.org/




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From vincent.roca@inrialpes.fr  Thu Apr 30 03:57:15 2009
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Hello Ali,

I'm not in favor of finalizing the WGLC for this document. IMHO we must,
first of all, finalize the framework document itself.

For instance, the terminology used by the interleaved-fec-scheme I-D
(section 3.1.) is not in line with the terminology used by the
fecframe-framework-03 I-D (Section 2):
	"Source Flow"   vs. "Source data flow"
	"Repair Flow"   vs. "Repair data flow"
	"Source Symbol" vs. nothing
	"Repair Symbol" vs. nothing
	"Source Packet" vs. nothing
	"Repair Packet" vs. nothing
And as I said before, IMHO the terminology proposed in the framework
I-D should be discussed too.

So let's do things in the right order, especially as the framework
WGLC finishes soon.

Additionally, I may have some comments for the interleaved-fec-scheme-04
I-D. Since I missed the WGLC deadline, you have the right to blame me ;-)
and I apologize in advance.

Regards,

  Vincent


Ali C. Begen (abegen) wrote:
> This version addresses the comments received from fecframe and avt so far. 
> http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interleaved-fec-scheme-04.txt
> 
> 
> Fecframe chairs,
> 
> Could we proceed and finalize WGLC?
> 
> -acbegen
>  
> 
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org] 
> Sent: Wednesday, April 29, 2009 2:56 PM
> To: Ali C. Begen (abegen)
> Subject: New Version Notification for draft-ietf-fecframe-interleaved-fec-scheme-04 
> 
> 
> A new version of I-D, draft-ietf-fecframe-interleaved-fec-scheme-04.txt has been successfuly submitted by Ali Begen and posted to the IETF repository.
> 
> Filename:	 draft-ietf-fecframe-interleaved-fec-scheme
> Revision:	 04
> Title:		 RTP Payload Format for 1-D Interleaved Parity FEC
> Creation_date:	 2009-04-29
> WG ID:		 fecframe
> Number_of_pages: 31
> 
> Abstract:
> This document defines a new RTP payload format for the Forward Error
> Correction (FEC) that is generated by the 1-D interleaved parity code
> from a source media encapsulated in RTP.  The 1-D interleaved parity
> code is a systematic code, where a number of repair symbols are
> generated from a set of source symbols and sent in a repair flow
> separate from the source flow that carries the source symbols.  The
> 1-D interleaved parity code offers a good protection against bursty
> packet losses at a cost of decent complexity.  The new payload format
> defined in this document is used (with some exceptions) as a part of
> the DVB Application-layer FEC specification.
>                                                                                   
> 
> 
> The IETF Secretariat.
> 
> 
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe


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From: Colin Perkins <csp@csperkins.org>
To: Roni Even <ron.even.tlv@gmail.com>
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Subject: Re: [AVT] SDP example in draft-versteeg-avt-rapid-synchronization-for-rtp-03
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Roni,

On 29 Apr 2009, at 19:58, Roni Even wrote:
> I agree that the assumption is that the client can use this ports  
> but this
> is why I suggest that the client should be able to override the  
> offer if it
> cannot use these ports. One example is when the client tries to join  
> two
> different multicast streams and both offer the same ports.

That's always been an issue with declarative SDP for multicast  
sessions, I'm not sure there's anything special about this SDP.

> My other concern is that the client IP address is not known from  
> this SDP,
> so there is another assumption that the IP used in the RTCP RAMS=R  
> from the
> client is the IP address for the unicast burst.

Sure - this gets into NAT traversal issues, of course.

Colin



> Roni
>
> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Wednesday, April 29, 2009 7:34 PM
> To: Roni Even
> Cc: avt@ietf.org
> Subject: Re: [AVT] SDP example in
> draft-versteeg-avt-rapid-synchronization-for-rtp-03
>
> Roni,
>
> It's been a while since I read SDP and its various extensions in great
> detail, but as far as I can tell, this SDP does provide sufficient
> information to join the session, on the assumption that symmetric RTP/
> RTCP ports can be used.
>
> Colin
>
>
>
> On 26 Apr 2009, at 19:48, Roni Even wrote:
>> I am looking for more feedback on the following topic (not by Roni
>> or Ali).
>> The example in the draft is coming from a distribution source using
>> for example SAP (multicast distribution).
>> There are three entities in the communication. The multicast
>> distribution source, the unicast retransmission which also serves as
>> a feedback target and a client that wants to receive the multicast
>> stream and will ask for unicast  resynchronization or retransmission.
>>
>> This SDP is coming from the distribution source and according to the
>> authors specify the multicast address, the unicast transport address
>> for the RTCP receiver reports for the multicast (SSM), the uincast
>> transport address on the feedback target/retransmission server that
>> will  be used by the clients to ask for retransmission streams using
>> RTCP-FB and the transport address on the client that will be used by
>> the feedback target to send the unicast RTP stream to.
>>
>> Does this SDP look like it supplies all this information enabling
>> interoperable applications.
>>
>>         v=0
>>        o=ali 1122334455 1122334466 IN IP4 rams.example.com
>>        s=Rapid Acquisition Examples
>>        t=0 0
>>        a=group:FID 1 2
>>        a=group:FID 3 4
>>        a=rtcp-unicast:rsi
>>        m=video 40000 RTP/AVPF 96
>>        i=Primary Multicast Stream #1
>>        c=IN IP4 224.1.1.1/255
>>        a=source-filter: incl IN IP4 224.1.1.1 192.0.2.2
>>        a=recvonly
>>        a=rtpmap:96 MP2T/90000
>>        a=rtcp:40001 IN IP4 192.0.2.1
>>        a=rtcp-fb:96 nack
>>        a=mid:1
>>        m=video 40002 RTP/AVPF 97
>>        i=Unicast Retransmission Stream #1 (Ret. Support Only)
>>        c=IN IP4 192.0.2.1
>>        a=recvonly
>>        a=rtpmap:97 rtx/90000
>>        a=rtcp:40003
>>        a=fmtp:97 apt=96
>>        a=fmtp:97 rtx-time=3000
>>        a=mid:2
>>
>>
>> Thanks
>> Roni Even
>
>
> -- 
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



-- 
Colin Perkins
http://csperkins.org/




From ron.even.tlv@gmail.com  Thu Apr 30 05:45:26 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Colin Perkins'" <csp@csperkins.org>
References: <49f4acd2.0c11660a.6549.ffffb32b@mx.google.com> <84F83FF0-B9B8-4FA6-A9F1-364782D91E8D@csperkins.org> <49f8a3c9.25e2660a.5cb1.fffff25e@mx.google.com> <8AEFABDC-708C-42B6-A596-DC3C6F07E238@csperkins.org>
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Subject: Re: [AVT] SDP example in draft-versteeg-avt-rapid-synchronization-for-rtp-03
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Colin,
The special case here is also mentioned in RFC 4566 section 5.7 in the first
bullet in the last sentence.

"It is not expected that unicast addresses will be given in a session
description that is communicated by a multicast announcement, though this is
not prohibited."


Roni

-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org] 
Sent: Thursday, April 30, 2009 2:41 PM
To: Roni Even
Cc: avt@ietf.org
Subject: Re: [AVT] SDP example in
draft-versteeg-avt-rapid-synchronization-for-rtp-03

Roni,

On 29 Apr 2009, at 19:58, Roni Even wrote:
> I agree that the assumption is that the client can use this ports  
> but this
> is why I suggest that the client should be able to override the  
> offer if it
> cannot use these ports. One example is when the client tries to join  
> two
> different multicast streams and both offer the same ports.

That's always been an issue with declarative SDP for multicast  
sessions, I'm not sure there's anything special about this SDP.

> My other concern is that the client IP address is not known from  
> this SDP,
> so there is another assumption that the IP used in the RTCP RAMS=R  
> from the
> client is the IP address for the unicast burst.

Sure - this gets into NAT traversal issues, of course.

Colin



> Roni
>
> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Wednesday, April 29, 2009 7:34 PM
> To: Roni Even
> Cc: avt@ietf.org
> Subject: Re: [AVT] SDP example in
> draft-versteeg-avt-rapid-synchronization-for-rtp-03
>
> Roni,
>
> It's been a while since I read SDP and its various extensions in great
> detail, but as far as I can tell, this SDP does provide sufficient
> information to join the session, on the assumption that symmetric RTP/
> RTCP ports can be used.
>
> Colin
>
>
>
> On 26 Apr 2009, at 19:48, Roni Even wrote:
>> I am looking for more feedback on the following topic (not by Roni
>> or Ali).
>> The example in the draft is coming from a distribution source using
>> for example SAP (multicast distribution).
>> There are three entities in the communication. The multicast
>> distribution source, the unicast retransmission which also serves as
>> a feedback target and a client that wants to receive the multicast
>> stream and will ask for unicast  resynchronization or retransmission.
>>
>> This SDP is coming from the distribution source and according to the
>> authors specify the multicast address, the unicast transport address
>> for the RTCP receiver reports for the multicast (SSM), the uincast
>> transport address on the feedback target/retransmission server that
>> will  be used by the clients to ask for retransmission streams using
>> RTCP-FB and the transport address on the client that will be used by
>> the feedback target to send the unicast RTP stream to.
>>
>> Does this SDP look like it supplies all this information enabling
>> interoperable applications.
>>
>>         v=0
>>        o=ali 1122334455 1122334466 IN IP4 rams.example.com
>>        s=Rapid Acquisition Examples
>>        t=0 0
>>        a=group:FID 1 2
>>        a=group:FID 3 4
>>        a=rtcp-unicast:rsi
>>        m=video 40000 RTP/AVPF 96
>>        i=Primary Multicast Stream #1
>>        c=IN IP4 224.1.1.1/255
>>        a=source-filter: incl IN IP4 224.1.1.1 192.0.2.2
>>        a=recvonly
>>        a=rtpmap:96 MP2T/90000
>>        a=rtcp:40001 IN IP4 192.0.2.1
>>        a=rtcp-fb:96 nack
>>        a=mid:1
>>        m=video 40002 RTP/AVPF 97
>>        i=Unicast Retransmission Stream #1 (Ret. Support Only)
>>        c=IN IP4 192.0.2.1
>>        a=recvonly
>>        a=rtpmap:97 rtx/90000
>>        a=rtcp:40003
>>        a=fmtp:97 apt=96
>>        a=fmtp:97 rtx-time=3000
>>        a=mid:2
>>
>>
>> Thanks
>> Roni Even
>
>
> -- 
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



-- 
Colin Perkins
http://csperkins.org/




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Thread-Topic: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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From: "Ingemar Johansson S" <ingemar.s.johansson@ericsson.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>, <avt@ietf.org>
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Subject: Re: [AVT] New draft of "Rapid Acquisition of Multicast RTP Sessions(RAMS)" available
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Hi

Some more comments below

Regards
/Ingemar=20

> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: den 25 april 2009 22:51
> To: Ingemar Johansson S; avt@ietf.org
> Cc: Bill Ver Steeg (versteb)
> Subject: RE: [AVT] New draft of "Rapid Acquisition of=20
> Multicast RTP Sessions(RAMS)" available
>=20
> > > I would not say it would be "fairly straighforward." To make the=20
> > > multicast bursting effective, many parameters need to be adjusted=20
> > > properly and optimized. Ow, it can easily degrade the performance=20
> > > even at the expense of additional complexity.
> > >=20
> > > A concern I have which is not addressed in your document=20
> is that to=20
> > > make the proposal make sense, Td needs to be small=20
> enough. But, to=20
> > > ensure that all receivers join the burst mcast session before the=20
> > > burst starts, Td cannot be very small and RS should not admit=20
> > > last-minute RAMS requests for that multicast burst (they need to=20
> > > wait for the next batch).
> > > This is because if some receivers miss the first few=20
> packets of the=20
> > > burst, their acquisition will fail. Also, how many=20
> clients can the=20
> > > retransmission server aggregate in a single burst batch in a=20
> > > practical scenario? The eligible requests should be=20
> received within=20
> > > +/- a few hundereds of milliseconds. How practical is this, I=20
> > > wonder.
> > The neat thing is that during low load conditions Td can be=20
> made very=20
> > low such that in practice only one user is served by each request.=20
> > This can be implemented with either unicast or multicast=20
> bursts that=20
> > only serve one STB. As he load increases Td increases and=20
> more users=20
> > that send a requests will be served by one multicast=20
> channel. As Td is=20
> > limited some requests will be too late but they can too be=20
> collected=20
> > and served by additional multicast feeds. Then depending on the=20
> > distribuion the tail may be handled with unicast bursts.
> > This is more or less an optimization issue on the servers.
>=20
> Agree.
>=20
> > The idea is not that the unicast and multicast alternatives should=20
> > compete with one another, rather they should complement each other,=20
> > due to the highly dynamic nature of channel switching behavior in a=20
> > population both alternatives are useful. I do believe that the=20
> > multicast solution can help to reduce deplyment cost.
>=20
> It can. However, remember that in a highly-loaded=20
> environment, if RS cannot provide (unicast) acceleration, it=20
> can deny the request. For the sake of doing multicast=20
> bursting (to save bandwidth), if RRs experience a slower=20
> acquisition (due to a large Td, for example) compared to a=20
> simple join, that is not good.
My feeling here is that a multicast solution for RAMS would help to keep
the load on a lower level and thus will avoid that a RAMS request is
denied.

>=20
> > I have tried to look for statistics on channel switching=20
> behavior and=20
> > of what I have seen sofar the "peak-to-average"
> > ratio of channel swithing density can be more than 5. This=20
> is seen in=20
> > a master thesis with the title "TPTV traffic measuring &=20
> modeling" by=20
> > Geng Yu (have not found it published anywhere yet).
>=20
> I don't think I have that study, either. But, there is a=20
> recent study in IMC 2008. Here is the link.
> http://portal.acm.org/ft_gateway.cfm?id=3D1452529&type=3Dpdf&coll=3D
portal&dl=3DACM
>=20
> The results are in general more or less similar to what we=20
> found in one of our own studies. The figure you will find=20
> interesting is Figure 14.=20
>=20
> For the most popular channel, the entire data set shows that=20
> on average 14 viewers tune into that channel per second=20
> (Figure 14a). Of course, there are occasions where a lot more=20
> viewers may tune at approximately the same time (within the=20
> same second), but this is for the top-ranked channel and the=20
> number 14 includes all viewers in the data set, which is=20
> about 250K (This is what I got from the text). If we consider=20
> only a subset of these viewers will be served by the same RS,=20
> the concurrency we will get in a single server will be far=20
> smaller, impying that multicast burst will not be needed most=20
> of the time.
>=20
> Different studies may suggest different numbers based on the=20
> regional characteristics. So, I don't think we should read=20
> too much into this study. If there are other studies, please=20
> post them here. So, we can make a better judgment.
I don't have any public links to the other reports (sorry) but I would
guess that the results are approximately similar. I see however one risk
in trying to compute median and mean from data that may have a very
large standard deviation like one can suspect in cases like this where
the the behavior is very bursty.=20

>=20
> > > The selection of a common burst bitrate is also critical. If=20
> > > everybody supports a different burst bitrate and there is a large=20
> > > variation, multicast bursting will perform poorly for the=20
> > > well-connected clients. One solution could be to include the=20
> > > receivers with an available bw that is smaller than R1 and larger=20
> > > than R2 in the multicast pool. The very fast and very=20
> slow receivers=20
> > > should be still served via unicast.
> > Sounds reasonable.
> >=20
> > >=20
> > > My personal view is that the multicast bursting proposal=20
> will be a=20
> > > nice solution when the issues above (and possible
> > > others) are addressed, however, I think it should be a separate=20
> > > draft that normatively references the unicast draft.
> > > This way, we can keep the unicast draft more focused.
> > Not sure that I agree here, it all depends on how the signaling is=20
> > standardized. For instance an STB likely needs to declare its=20
> > unicast/multicast preference or capability in the RAMS-R as=20
> the SDP's=20
> > are likely only declarative.
>=20
> There can be a TLV in the request to do this, no?=20
Possible, need to investigate this option closer.=20



>=20
> BR,
> -acbegen
>=20

From tme@multicasttech.com  Thu Apr 30 06:50:40 2009
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From: Marshall Eubanks <tme@multicasttech.com>
To: "Ali C. Begen (abegen)" <abegen@cisco.com>
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On Apr 30, 2009, at 8:53 AM, Ali C. Begen (abegen) wrote:

> Hi Vincent
>
> This draft is not an fec scheme so it is not related to fec  
> framework. It is an rtp payload format draft. The wglc has been  
> running almost a month now. If you have comments in this  
> perspective, pls submit them soon.
>
>

<Chair hat on>

I was about to close this WGLC, but we can wait a short while for new  
comments.

<Chair hat off>

Marshall

>
>
>
> -acbegen
>
> ----- Original Message -----
> From: Vincent Roca <vincent.roca@inrialpes.fr>
> To: Ali C. Begen (abegen)
> Cc: fecframe@ietf.org <fecframe@ietf.org>; avt@ietf.org <avt@ietf.org>
> Sent: Thu Apr 30 03:58:35 2009
> Subject: Re: [Fecframe] FW: New Version Notification for         
> draft-ietf-fecframe-interleaved-fec-scheme-04
>
> Hello Ali,
>
> I'm not in favor of finalizing the WGLC for this document. IMHO we  
> must,
> first of all, finalize the framework document itself.
>
> For instance, the terminology used by the interleaved-fec-scheme I-D
> (section 3.1.) is not in line with the terminology used by the
> fecframe-framework-03 I-D (Section 2):
>        "Source Flow"   vs. "Source data flow"
>        "Repair Flow"   vs. "Repair data flow"
>        "Source Symbol" vs. nothing
>        "Repair Symbol" vs. nothing
>        "Source Packet" vs. nothing
>        "Repair Packet" vs. nothing
> And as I said before, IMHO the terminology proposed in the framework
> I-D should be discussed too.
>
> So let's do things in the right order, especially as the framework
> WGLC finishes soon.
>
> Additionally, I may have some comments for the interleaved-fec- 
> scheme-04
> I-D. Since I missed the WGLC deadline, you have the right to blame  
> me ;-)
> and I apologize in advance.
>
> Regards,
>
>  Vincent
>
>
> Ali C. Begen (abegen) wrote:
> > This version addresses the comments received from fecframe and avt  
> so far.
> > http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interleaved-fec-scheme-04.txt
> >
> >
> > Fecframe chairs,
> >
> > Could we proceed and finalize WGLC?
> >
> > -acbegen
> >
> >
> > -----Original Message-----
> > From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> > Sent: Wednesday, April 29, 2009 2:56 PM
> > To: Ali C. Begen (abegen)
> > Subject: New Version Notification for draft-ietf-fecframe- 
> interleaved-fec-scheme-04
> >
> >
> > A new version of I-D, draft-ietf-fecframe-interleaved-fec- 
> scheme-04.txt has been successfuly submitted by Ali Begen and posted  
> to the IETF repository.
> >
> > Filename:      draft-ietf-fecframe-interleaved-fec-scheme
> > Revision:      04
> > Title:                 RTP Payload Format for 1-D Interleaved  
> Parity FEC
> > Creation_date:         2009-04-29
> > WG ID:                 fecframe
> > Number_of_pages: 31
> >
> > Abstract:
> > This document defines a new RTP payload format for the Forward Error
> > Correction (FEC) that is generated by the 1-D interleaved parity  
> code
> > from a source media encapsulated in RTP.  The 1-D interleaved parity
> > code is a systematic code, where a number of repair symbols are
> > generated from a set of source symbols and sent in a repair flow
> > separate from the source flow that carries the source symbols.  The
> > 1-D interleaved parity code offers a good protection against bursty
> > packet losses at a cost of decent complexity.  The new payload  
> format
> > defined in this document is used (with some exceptions) as a part of
> > the DVB Application-layer FEC specification.
> >
> >
> >
> > The IETF Secretariat.
> >
> >
> > _______________________________________________
> > Fecframe mailing list
> > Fecframe@ietf.org
> > https://www.ietf.org/mailman/listinfo/fecframe
>
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe

Regards
Marshall Eubanks
CEO / AmericaFree.TV


From ron.even.tlv@gmail.com  Thu Apr 30 11:13:08 2009
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <vincent.roca@inrialpes.fr>
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Subject: Re: [AVT] [Fecframe] FW: New Version Notification	for	draft-ietf-fecframe-interleaved-fec-scheme-04
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Ali,

I noticed that the fec framework will allow SSRC multiplexing, are you =
going to specify this in this draft in the SDP offer answer section.=20

Maybe you should also discuss when to use application media type and =
when to use video or audio

=20

Roni

=20

From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of =
Ali C. Begen (abegen)
Sent: Thursday, April 30, 2009 3:54 PM
To: vincent.roca@inrialpes.fr
Cc: avt@ietf.org; fecframe@ietf.org
Subject: Re: [AVT] [Fecframe] FW: New Version Notification for =
draft-ietf-fecframe-interleaved-fec-scheme-04

=20

Hi Vincent

This draft is not an fec scheme so it is not related to fec framework. =
It is an rtp payload format draft. The wglc has been running almost a =
month now. If you have comments in this perspective, pls submit them =
soon.




-acbegen

----- Original Message -----
From: Vincent Roca <vincent.roca@inrialpes.fr>
To: Ali C. Begen (abegen)
Cc: fecframe@ietf.org <fecframe@ietf.org>; avt@ietf.org <avt@ietf.org>
Sent: Thu Apr 30 03:58:35 2009
Subject: Re: [Fecframe] FW: New Version Notification for        =
draft-ietf-fecframe-interleaved-fec-scheme-04

Hello Ali,

I'm not in favor of finalizing the WGLC for this document. IMHO we must,
first of all, finalize the framework document itself.

For instance, the terminology used by the interleaved-fec-scheme I-D
(section 3.1.) is not in line with the terminology used by the
fecframe-framework-03 I-D (Section 2):
        "Source Flow"   vs. "Source data flow"
        "Repair Flow"   vs. "Repair data flow"
        "Source Symbol" vs. nothing
        "Repair Symbol" vs. nothing
        "Source Packet" vs. nothing
        "Repair Packet" vs. nothing
And as I said before, IMHO the terminology proposed in the framework
I-D should be discussed too.

So let's do things in the right order, especially as the framework
WGLC finishes soon.

Additionally, I may have some comments for the interleaved-fec-scheme-04
I-D. Since I missed the WGLC deadline, you have the right to blame me =
;-)
and I apologize in advance.

Regards,

  Vincent


Ali C. Begen (abegen) wrote:
> This version addresses the comments received from fecframe and avt so =
far.
> =
http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interleaved-fec-s=
cheme-04.txt
>
>
> Fecframe chairs,
>
> Could we proceed and finalize WGLC?
>
> -acbegen
>=20
>
> -----Original Message-----
> From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> Sent: Wednesday, April 29, 2009 2:56 PM
> To: Ali C. Begen (abegen)
> Subject: New Version Notification for =
draft-ietf-fecframe-interleaved-fec-scheme-04
>
>
> A new version of I-D, =
draft-ietf-fecframe-interleaved-fec-scheme-04.txt has been successfuly =
submitted by Ali Begen and posted to the IETF repository.
>
> Filename:      draft-ietf-fecframe-interleaved-fec-scheme
> Revision:      04
> Title:                 RTP Payload Format for 1-D Interleaved Parity =
FEC
> Creation_date:         2009-04-29
> WG ID:                 fecframe
> Number_of_pages: 31
>
> Abstract:
> This document defines a new RTP payload format for the Forward Error
> Correction (FEC) that is generated by the 1-D interleaved parity code
> from a source media encapsulated in RTP.  The 1-D interleaved parity
> code is a systematic code, where a number of repair symbols are
> generated from a set of source symbols and sent in a repair flow
> separate from the source flow that carries the source symbols.  The
> 1-D interleaved parity code offers a good protection against bursty
> packet losses at a cost of decent complexity.  The new payload format
> defined in this document is used (with some exceptions) as a part of
> the DVB Application-layer FEC specification.
>                                                                        =
         =20
>
>
> The IETF Secretariat.
>
>
> _______________________________________________
> Fecframe mailing list
> Fecframe@ietf.org
> https://www.ietf.org/mailman/listinfo/fecframe


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<title>Re: [Fecframe] FW: New Version Notification for
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<div class=3DSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Ali,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I noticed that the fec framework will allow SSRC =
multiplexing,
are you going to specify this in this draft in the SDP offer answer =
section. <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Maybe you should also discuss when to use application =
media type
and when to use video or audio<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Roni<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p>&nbsp;</o:p></span></p>

<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
0in 0in 0in'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] <b>On Behalf Of =
</b>Ali C.
Begen (abegen)<br>
<b>Sent:</b> Thursday, April 30, 2009 3:54 PM<br>
<b>To:</b> vincent.roca@inrialpes.fr<br>
<b>Cc:</b> avt@ietf.org; fecframe@ietf.org<br>
<b>Subject:</b> Re: [AVT] [Fecframe] FW: New Version Notification for
draft-ietf-fecframe-interleaved-fec-scheme-04<o:p></o:p></span></p>

</div>

</div>

<p class=3DMsoNormal><o:p>&nbsp;</o:p></p>

<p style=3D'margin-bottom:12.0pt'><span style=3D'font-size:10.0pt'>Hi =
Vincent<br>
<br>
This draft is not an fec scheme so it is not related to fec framework. =
It is an
rtp payload format draft. The wglc has been running almost a month now. =
If you
have comments in this perspective, pls submit them soon.<br>
<br>
<br>
<br>
<br>
-acbegen<br>
<br>
----- Original Message -----<br>
From: Vincent Roca &lt;vincent.roca@inrialpes.fr&gt;<br>
To: Ali C. Begen (abegen)<br>
Cc: fecframe@ietf.org &lt;fecframe@ietf.org&gt;; avt@ietf.org
&lt;avt@ietf.org&gt;<br>
Sent: Thu Apr 30 03:58:35 2009<br>
Subject: Re: [Fecframe] FW: New Version Notification
for&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
draft-ietf-fecframe-interleaved-fec-scheme-04<br>
<br>
Hello Ali,<br>
<br>
I'm not in favor of finalizing the WGLC for this document. IMHO we =
must,<br>
first of all, finalize the framework document itself.<br>
<br>
For instance, the terminology used by the interleaved-fec-scheme I-D<br>
(section 3.1.) is not in line with the terminology used by the<br>
fecframe-framework-03 I-D (Section 2):<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Source =
Flow&quot;&nbsp;&nbsp;
vs. &quot;Source data flow&quot;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Repair =
Flow&quot;&nbsp;&nbsp;
vs. &quot;Repair data flow&quot;<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Source Symbol&quot; vs.
nothing<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Repair Symbol&quot; vs.
nothing<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Source Packet&quot; vs.
nothing<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &quot;Repair Packet&quot; vs. =
nothing<br>
And as I said before, IMHO the terminology proposed in the framework<br>
I-D should be discussed too.<br>
<br>
So let's do things in the right order, especially as the framework<br>
WGLC finishes soon.<br>
<br>
Additionally, I may have some comments for the =
interleaved-fec-scheme-04<br>
I-D. Since I missed the WGLC deadline, you have the right to blame me =
;-)<br>
and I apologize in advance.<br>
<br>
Regards,<br>
<br>
&nbsp; Vincent<br>
<br>
<br>
Ali C. Begen (abegen) wrote:<br>
&gt; This version addresses the comments received from fecframe and avt =
so far.<br>
&gt; <a
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interleav=
ed-fec-scheme-04.txt">http://www.ietf.org/internet-drafts/draft-ietf-fecf=
rame-interleaved-fec-scheme-04.txt</a><br>
&gt;<br>
&gt;<br>
&gt; Fecframe chairs,<br>
&gt;<br>
&gt; Could we proceed and finalize WGLC?<br>
&gt;<br>
&gt; -acbegen<br>
&gt;&nbsp;<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: IETF I-D Submission Tool [<a =
href=3D"mailto:idsubmission@ietf.org">mailto:idsubmission@ietf.org</a>]<b=
r>
&gt; Sent: Wednesday, April 29, 2009 2:56 PM<br>
&gt; To: Ali C. Begen (abegen)<br>
&gt; Subject: New Version Notification for =
draft-ietf-fecframe-interleaved-fec-scheme-04<br>
&gt;<br>
&gt;<br>
&gt; A new version of I-D, =
draft-ietf-fecframe-interleaved-fec-scheme-04.txt
has been successfuly submitted by Ali Begen and posted to the IETF =
repository.<br>
&gt;<br>
&gt; Filename:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
draft-ietf-fecframe-interleaved-fec-scheme<br>
&gt; Revision:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 04<br>
&gt; Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RTP Payload Format for =
1-D
Interleaved Parity FEC<br>
&gt; Creation_date:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2009-04-29<br>
&gt; WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; fecframe<br>
&gt; Number_of_pages: 31<br>
&gt;<br>
&gt; Abstract:<br>
&gt; This document defines a new RTP payload format for the Forward =
Error<br>
&gt; Correction (FEC) that is generated by the 1-D interleaved parity =
code<br>
&gt; from a source media encapsulated in RTP.&nbsp; The 1-D interleaved =
parity<br>
&gt; code is a systematic code, where a number of repair symbols are<br>
&gt; generated from a set of source symbols and sent in a repair =
flow<br>
&gt; separate from the source flow that carries the source =
symbols.&nbsp; The<br>
&gt; 1-D interleaved parity code offers a good protection against =
bursty<br>
&gt; packet losses at a cost of decent complexity.&nbsp; The new payload =
format<br>
&gt; defined in this document is used (with some exceptions) as a part =
of<br>
&gt; the DVB Application-layer FEC specification.<br>
&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<br>
&gt;<br>
&gt;<br>
&gt; The IETF Secretariat.<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; Fecframe mailing list<br>
&gt; Fecframe@ietf.org<br>
&gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/fecframe">https://www.ietf.=
org/mailman/listinfo/fecframe</a></span><o:p></o:p></p>

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Hi Roni and all other IETF AVT WG participants,

Considering that the "Rapid Acquisition of Multicast RTP Sessions
(RAMS)" draft (revision 3)

-has taken into account many of the remarks that came out of the
break-out session in the San Francisco meeting
-includes the flexible burst rate adaptation capability, the prime
differentiator in the "draft-peilin-avt-rtp-burst" from previous
versions of the RAMS draft, and as such the RAMS draft has now merged
and integrated the solutions proposed in three different originating
drafts.=20
-has included adequate flexibility by allowing TLV fields in the
signaling messages, allowing other concepts to leverage the solution in
the future (including potentially a multicast-based solution as proposed
in draft-johansson-avt-mcast-based-rams-00).

, I would like to know whether there is still a blocking issue for
preventing this draft from becoming a working group item.

I followed with great interest in the last weeks the SDP discussions
related to the SDP signaling example in the RAMS v 3 draft (I am
unfortunately not sufficiently experienced in that domain to
contribute), where the latest shared opinions make me think that this is
not really a blocking item.

Considering that the 1st draft was submitted back in august last year,
and the various subsequent revisions and merges since then, can we now
proceed with the draft as a working group subject?

Kind Regards,
Tom Van Caenegem (co-author of the "Rapid Acquisition of Multicast
RTPSessions (RAMS)" draft)

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<BR>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Hi Roni and all =
other IETF AVT WG participants,</FONT></SPAN>
</P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Considering that =
the &quot;Rapid Acquisition of Multicast RTP Sessions (RAMS)&quot; draft =
(revision 3)</FONT></SPAN>
</P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">-has taken into =
account many of the remarks that came out of the break-out session in =
the San Francisco meeting</FONT></SPAN>

<BR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">-includes the =
flexible burst rate adaptation capability, the prime differentiator in =
the &quot;draft-peilin-avt-rtp-burst&quot; from previous versions of the =
RAMS draft, and as such the RAMS draft has now merged and integrated the =
solutions proposed in three different originating =
drafts.</FONT></SPAN><SPAN LANG=3D"nl-be"> </SPAN></P>

<P><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Arial">-has included =
adequate flexibility by allowing TLV fields in the signaling messages, =
allowing other concepts to leverage the solution in the future =
(including potentially a multicast-based solution as proposed =
in</FONT></SPAN><SPAN LANG=3D"en-us"> <FONT SIZE=3D2 FACE=3D"Courier =
New">draft-johansson-avt-mcast-based-rams-0</FONT><FONT SIZE=3D2 =
FACE=3D"Courier New">0).</FONT></SPAN></P>

<P><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Arial">, I would like to =
know whether there is still a blocking issue for preventing this draft =
from becoming a working group item.</FONT></SPAN></P>

<P><SPAN LANG=3D"nl-be"><FONT SIZE=3D2 FACE=3D"Arial">I followed with =
great interest in the last weeks the SDP discussions related to the SDP =
signaling example in the RAMS v 3 draft (I am unfortunately not =
sufficiently experienced in that domain to contribute), where the latest =
shared opinions make me think that this is not really a blocking =
item.</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Considering that =
the 1st draft was submitted back in august last year, and the various =
subsequent revisions and merges since then, can we now proceed with the =
draft as a working group subject?</FONT></SPAN></P>

<P><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Kind =
Regards,</FONT></SPAN>

<BR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">Tom Van Caenegem =
(co-author of the &quot;Rapid Acquisition of Multicast RTPSessions =
(RAMS)&quot; draft)</FONT></SPAN><SPAN LANG=3D"nl-be"></SPAN>
</P>

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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>, <vincent.roca@inrialpes.fr>
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Subject: Re: [AVT] [Fecframe] FW: New Version Notification	for	draft-ietf-fecframe-interleaved-fec-scheme-04
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Ali,
You can find examples in section 8.6 to 8.8 of RFC 4588, notice that =
there is an optional parameter for the rtx stream that defines the =
relation between the original and the rtx stream. Similar solution is =
also specified for redundant encoding red payload type (RFC 2198). You =
are offering both streams with different payload type numbers in the =
same m-line.
In this case the media type must be video or audio same as the media =
type of the protected stream. This is why you needed the registration =
for type audio and video

Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
Sent: Thursday, April 30, 2009 10:01 PM
To: Roni Even; vincent.roca@inrialpes.fr
Cc: avt@ietf.org; fecframe@ietf.org
Subject: RE: [AVT] [Fecframe] FW: New Version Notification for =
draft-ietf-fecframe-interleaved-fec-scheme-04

Hi Roni,=20

> I noticed that the fec framework will allow SSRC=20

Yes, it will not be the recommended way of using FEC, but it will be =
supported. The 4756bis draft also supports it now.

> multiplexing, are you going to specify this in this draft in=20
> the SDP offer answer section.=20

I am not sure what I need to specify additionally in this section =
regarding SSRC multiplexing. Could you let me know if there is such an =
example?
=20
> Maybe you should also discuss when to use application media=20
> type and when to use video or audio

I believe the FEC is always application type, we did registration for =
video, audio, etc. since FEC could relate to sessions carrying such =
types.

BR,
-acbegen
=20
> =20
>=20
> Roni
>=20
> =20
>=20
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> Behalf Of Ali C. Begen (abegen)
> Sent: Thursday, April 30, 2009 3:54 PM
> To: vincent.roca@inrialpes.fr
> Cc: avt@ietf.org; fecframe@ietf.org
> Subject: Re: [AVT] [Fecframe] FW: New Version Notification=20
> for draft-ietf-fecframe-interleaved-fec-scheme-04
>=20
> =20
>=20
> Hi Vincent
>=20
> This draft is not an fec scheme so it is not related to fec=20
> framework. It is an rtp payload format draft. The wglc has=20
> been running almost a month now. If you have comments in this=20
> perspective, pls submit them soon.
>=20
>=20
>=20
>=20
> -acbegen
>=20
> ----- Original Message -----
> From: Vincent Roca <vincent.roca@inrialpes.fr>
> To: Ali C. Begen (abegen)
> Cc: fecframe@ietf.org <fecframe@ietf.org>; avt@ietf.org <avt@ietf.org>
> Sent: Thu Apr 30 03:58:35 2009
> Subject: Re: [Fecframe] FW: New Version Notification for     =20
>   draft-ietf-fecframe-interleaved-fec-scheme-04
>=20
> Hello Ali,
>=20
> I'm not in favor of finalizing the WGLC for this document.=20
> IMHO we must,
> first of all, finalize the framework document itself.
>=20
> For instance, the terminology used by the interleaved-fec-scheme I-D
> (section 3.1.) is not in line with the terminology used by the
> fecframe-framework-03 I-D (Section 2):
>         "Source Flow"   vs. "Source data flow"
>         "Repair Flow"   vs. "Repair data flow"
>         "Source Symbol" vs. nothing
>         "Repair Symbol" vs. nothing
>         "Source Packet" vs. nothing
>         "Repair Packet" vs. nothing
> And as I said before, IMHO the terminology proposed in the framework
> I-D should be discussed too.
>=20
> So let's do things in the right order, especially as the framework
> WGLC finishes soon.
>=20
> Additionally, I may have some comments for the=20
> interleaved-fec-scheme-04
> I-D. Since I missed the WGLC deadline, you have the right to=20
> blame me ;-)
> and I apologize in advance.
>=20
> Regards,
>=20
>   Vincent
>=20
>=20
> Ali C. Begen (abegen) wrote:
> > This version addresses the comments received from fecframe=20
> and avt so far.
> >=20
> http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interl
eaved-fec-scheme-04.txt
> >
> >
> > Fecframe chairs,
> >
> > Could we proceed and finalize WGLC?
> >
> > -acbegen
> >=20
> >
> > -----Original Message-----
> > From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> > Sent: Wednesday, April 29, 2009 2:56 PM
> > To: Ali C. Begen (abegen)
> > Subject: New Version Notification for=20
> draft-ietf-fecframe-interleaved-fec-scheme-04
> >
> >
> > A new version of I-D,=20
> draft-ietf-fecframe-interleaved-fec-scheme-04.txt has been=20
> successfuly submitted by Ali Begen and posted to the IETF repository.
> >
> > Filename:      draft-ietf-fecframe-interleaved-fec-scheme
> > Revision:      04
> > Title:                 RTP Payload Format for 1-D=20
> Interleaved Parity FEC
> > Creation_date:         2009-04-29
> > WG ID:                 fecframe
> > Number_of_pages: 31
> >
> > Abstract:
> > This document defines a new RTP payload format for the Forward Error
> > Correction (FEC) that is generated by the 1-D interleaved=20
> parity code
> > from a source media encapsulated in RTP.  The 1-D interleaved parity
> > code is a systematic code, where a number of repair symbols are
> > generated from a set of source symbols and sent in a repair flow
> > separate from the source flow that carries the source symbols.  The
> > 1-D interleaved parity code offers a good protection against bursty
> > packet losses at a cost of decent complexity.  The new=20
> payload format
> > defined in this document is used (with some exceptions) as a part of
> > the DVB Application-layer FEC specification.
> >                                                            =20
>                     =20
> >
> >
> > The IETF Secretariat.
> >
> >
> > _______________________________________________
> > Fecframe mailing list
> > Fecframe@ietf.org
> > https://www.ietf.org/mailman/listinfo/fecframe
>=20
>=20


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: <avt@ietf.org>
Date: Thu, 30 Apr 2009 23:12:34 +0300
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Subject: [AVT] Adopting Rapid Acquisition of Multicast RTP Sessions (RAMS) as a working group item
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Hi,

I would like to ask if there is interest to adopt Rapid Acquisition of
Multicast RTP Sessions (RAMS) as a working group item. The suggestion is to
start the work using
http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronization-for-rtp-
03 as the initial text for the work.

 

I would also like to suggest that Bill Versteeg will be the editor of the WG
document if adopted.

 

Please send any feedback to the mailing list by Friday May 8th. If there
will be a consensus we will accept this topic and have the above draft as
the initial document.

 

Thanks

Roni Even

AVT co-chair


------=_NextPart_000_00C2_01C9C9E9.302C1C90
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<div class=3DSection1>

<p class=3DMsoNormal>Hi,<o:p></o:p></p>

<p class=3DMsoNormal>I would like to ask if there is interest to adopt =
<span
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Rapid =
Acquisition of
Multicast RTP Sessions (RAMS) as a working group item. The suggestion is =
to
start the work using <a
href=3D"http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchronizati=
on-for-rtp-03">http://tools.ietf.org/html/draft-versteeg-avt-rapid-synchr=
onization-for-rtp-03</a>
as the initial text for the work.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>I
would also like to suggest that Bill Versteeg will be the editor of the =
WG
document if adopted.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Please
send any feedback to the mailing list by Friday May 8<sup>th</sup>. =
</span>If
there will be a consensus we will accept this topic and have the above =
draft as
the initial document.<span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p></o:p></=
span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;</=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks<o:p></=
o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Roni
Even<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>AVT
co-chair<o:p></o:p></span></p>

</div>

</body>

</html>

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: "Roni Even" <ron.even.tlv@gmail.com>, <vincent.roca@inrialpes.fr>
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ali C. Begen \(abegen\)'" <abegen@cisco.com>
References: <04CAD96D4C5A3D48B1919248A8FE0D54066184FC@xmb-sjc-215.amer.cisco.com> <49f9ea83.0637560a.4488.62d2@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D540934913F@xmb-sjc-215.amer.cisco.com> <49fa0498.0405560a.615f.677e@mx.google.com> <04CAD96D4C5A3D48B1919248A8FE0D540934937B@xmb-sjc-215.amer.cisco.com>
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Cc: avt@ietf.org, fecframe@ietf.org
Subject: Re: [AVT] [Fecframe] FW: New Version Notification	for	draft-ietf-fecframe-interleaved-fec-scheme-04
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Hi Ali,
This is up to the WG to decide if they want to have support for SSRC =
multiplexing and the implications of this support. I am not sure what =
you mean by "we cannot have a parameter", I see only your view on the =
list.

If only session multiplexing is allowed then the draft should clearly =
state it and in this case I am not sure why you are registering the =
video, audio and text media subtypes.

Roni

-----Original Message-----
From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
Sent: Friday, May 01, 2009 2:45 AM
To: Roni Even; vincent.roca@inrialpes.fr
Cc: avt@ietf.org; fecframe@ietf.org
Subject: RE: [AVT] [Fecframe] FW: New Version Notification for =
draft-ietf-fecframe-interleaved-fec-scheme-04

Hi Roni,

Thanks for the references. However, we cannot have a parameter that will =
relate the FEC stream to the source as RFC 4588 did with "apt" since a =
lot of possibilities (multiple sources and/or multiple repair flows) are =
possible with FEC. Instead, we use grouping and this works just fine in =
session multiplexing.

Regarding SSRC multiplexing of source and FEC streams, I don't think =
giving an explicit example is needed or it is a good idea for this draft =
since SSRC-level grouping is just introduced in 4756bis and it will =
include such examples anyway (actually the version I submitted today =
includes an example). We should better only provide session multiplexing =
examples in this draft. This also emphasizes the recommended way of =
doing FEC.

Reg the registration of the media types and SDP mappings, I believe the =
draft has the necessary details.

BR,
-acbegen

> -----Original Message-----
> From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
> Sent: Thursday, April 30, 2009 1:03 PM
> To: Ali C. Begen (abegen); vincent.roca@inrialpes.fr
> Cc: avt@ietf.org; fecframe@ietf.org
> Subject: RE: [AVT] [Fecframe] FW: New Version Notification=20
> for draft-ietf-fecframe-interleaved-fec-scheme-04
>=20
> Ali,
> You can find examples in section 8.6 to 8.8 of RFC 4588,=20
> notice that there is an optional parameter for the rtx stream=20
> that defines the relation between the original and the rtx=20
> stream. Similar solution is also specified for redundant=20
> encoding red payload type (RFC 2198). You are offering both=20
> streams with different payload type numbers in the same m-line.
> In this case the media type must be video or audio same as=20
> the media type of the protected stream. This is why you=20
> needed the registration for type audio and video
>=20
> Roni
>=20
> -----Original Message-----
> From: Ali C. Begen (abegen) [mailto:abegen@cisco.com]=20
> Sent: Thursday, April 30, 2009 10:01 PM
> To: Roni Even; vincent.roca@inrialpes.fr
> Cc: avt@ietf.org; fecframe@ietf.org
> Subject: RE: [AVT] [Fecframe] FW: New Version Notification=20
> for draft-ietf-fecframe-interleaved-fec-scheme-04
>=20
> Hi Roni,=20
>=20
> > I noticed that the fec framework will allow SSRC=20
>=20
> Yes, it will not be the recommended way of using FEC, but it=20
> will be supported. The 4756bis draft also supports it now.
>=20
> > multiplexing, are you going to specify this in this draft in=20
> > the SDP offer answer section.=20
>=20
> I am not sure what I need to specify additionally in this=20
> section regarding SSRC multiplexing. Could you let me know if=20
> there is such an example?
> =20
> > Maybe you should also discuss when to use application media=20
> > type and when to use video or audio
>=20
> I believe the FEC is always application type, we did=20
> registration for video, audio, etc. since FEC could relate to=20
> sessions carrying such types.
>=20
> BR,
> -acbegen
> =20
> > =20
> >=20
> > Roni
> >=20
> > =20
> >=20
> > From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
> > Behalf Of Ali C. Begen (abegen)
> > Sent: Thursday, April 30, 2009 3:54 PM
> > To: vincent.roca@inrialpes.fr
> > Cc: avt@ietf.org; fecframe@ietf.org
> > Subject: Re: [AVT] [Fecframe] FW: New Version Notification=20
> > for draft-ietf-fecframe-interleaved-fec-scheme-04
> >=20
> > =20
> >=20
> > Hi Vincent
> >=20
> > This draft is not an fec scheme so it is not related to fec=20
> > framework. It is an rtp payload format draft. The wglc has=20
> > been running almost a month now. If you have comments in this=20
> > perspective, pls submit them soon.
> >=20
> >=20
> >=20
> >=20
> > -acbegen
> >=20
> > ----- Original Message -----
> > From: Vincent Roca <vincent.roca@inrialpes.fr>
> > To: Ali C. Begen (abegen)
> > Cc: fecframe@ietf.org <fecframe@ietf.org>; avt@ietf.org=20
> <avt@ietf.org>
> > Sent: Thu Apr 30 03:58:35 2009
> > Subject: Re: [Fecframe] FW: New Version Notification for     =20
> >   draft-ietf-fecframe-interleaved-fec-scheme-04
> >=20
> > Hello Ali,
> >=20
> > I'm not in favor of finalizing the WGLC for this document.=20
> > IMHO we must,
> > first of all, finalize the framework document itself.
> >=20
> > For instance, the terminology used by the interleaved-fec-scheme I-D
> > (section 3.1.) is not in line with the terminology used by the
> > fecframe-framework-03 I-D (Section 2):
> >         "Source Flow"   vs. "Source data flow"
> >         "Repair Flow"   vs. "Repair data flow"
> >         "Source Symbol" vs. nothing
> >         "Repair Symbol" vs. nothing
> >         "Source Packet" vs. nothing
> >         "Repair Packet" vs. nothing
> > And as I said before, IMHO the terminology proposed in the framework
> > I-D should be discussed too.
> >=20
> > So let's do things in the right order, especially as the framework
> > WGLC finishes soon.
> >=20
> > Additionally, I may have some comments for the=20
> > interleaved-fec-scheme-04
> > I-D. Since I missed the WGLC deadline, you have the right to=20
> > blame me ;-)
> > and I apologize in advance.
> >=20
> > Regards,
> >=20
> >   Vincent
> >=20
> >=20
> > Ali C. Begen (abegen) wrote:
> > > This version addresses the comments received from fecframe=20
> > and avt so far.
> > >=20
> > http://www.ietf.org/internet-drafts/draft-ietf-fecframe-interl
> eaved-fec-scheme-04.txt
> > >
> > >
> > > Fecframe chairs,
> > >
> > > Could we proceed and finalize WGLC?
> > >
> > > -acbegen
> > >=20
> > >
> > > -----Original Message-----
> > > From: IETF I-D Submission Tool [mailto:idsubmission@ietf.org]
> > > Sent: Wednesday, April 29, 2009 2:56 PM
> > > To: Ali C. Begen (abegen)
> > > Subject: New Version Notification for=20
> > draft-ietf-fecframe-interleaved-fec-scheme-04
> > >
> > >
> > > A new version of I-D,=20
> > draft-ietf-fecframe-interleaved-fec-scheme-04.txt has been=20
> > successfuly submitted by Ali Begen and posted to the IETF=20
> repository.
> > >
> > > Filename:      draft-ietf-fecframe-interleaved-fec-scheme
> > > Revision:      04
> > > Title:                 RTP Payload Format for 1-D=20
> > Interleaved Parity FEC
> > > Creation_date:         2009-04-29
> > > WG ID:                 fecframe
> > > Number_of_pages: 31
> > >
> > > Abstract:
> > > This document defines a new RTP payload format for the=20
> Forward Error
> > > Correction (FEC) that is generated by the 1-D interleaved=20
> > parity code
> > > from a source media encapsulated in RTP.  The 1-D=20
> interleaved parity
> > > code is a systematic code, where a number of repair symbols are
> > > generated from a set of source symbols and sent in a repair flow
> > > separate from the source flow that carries the source=20
> symbols.  The
> > > 1-D interleaved parity code offers a good protection=20
> against bursty
> > > packet losses at a cost of decent complexity.  The new=20
> > payload format
> > > defined in this document is used (with some exceptions)=20
> as a part of
> > > the DVB Application-layer FEC specification.
> > >                                                            =20
> >                     =20
> > >
> > >
> > > The IETF Secretariat.
> > >
> > >
> > > _______________________________________________
> > > Fecframe mailing list
> > > Fecframe@ietf.org
> > > https://www.ietf.org/mailman/listinfo/fecframe
> >=20
> >=20
>=20
>=20

