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Date: Tue, 01 Apr 2003 17:54:36 +0200
From: Jose Rey <rey@panasonic.de>
Subject: RE: [AVT] SDP signaling for RTCP XR format
In-reply-to: <3E885013.4090804@era.ericsson.se>
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Hi Magnus, 


please see my asnwers inline, I cut much of the text for readibility.

> Yes, I will try to clarify this. My intention is that the SDP 
> parameters 
> should be used if there is SDP signaling available for this 
> application. 
> The one exception is that application  foo always uses these reports. 
> And since you are running the client for application foo it 
> knows that 
> it shall use the report formats.
> 
> >To solve this, why not declare some kind of attribute in SDP 
> that defines the set of feedback messages that the 
> application needs of?
> >
> 
> You mean in the general case for like SDES information types, etc.?

I think Colin's comment solves this question:


	> RTP allows applications to send new types of RTCP packet, 
	> even if these are
	> not explicitly signalled. RTCP XR shouldn't impose additional 
	> restrictions.
	> 
	> Colin

---
---

As to the behaviour in O/A model, I went over it again and this is what I got:

* answers to offers of cat1-params (sendonly from O->A): if present in response -> SHOULD be sent; if not present -> MAY be sent, whereby
 - this last MAY allows for undeclared use of any extension and, 
 - the SHOULD is based on the supposition that if both the offerer and answerer declare a cat1-param it is because it should be better(or even necessary) for the streaming session and will be used, otherwise the A could omit it. 

* answers to offers of cat2-params (sendonly from O->A): whether present in answer or not-> MAY be sent; 
- this is more flexible since the decision (and probably the benefit) on whether to use cat2-params lies at the receiver. A MUST would avoid sending an undeclared extension and also it cannot be assumed the sender is reporting on all extensions it supports. 

Moreover, it should be stated that answerer must have some mechanism to realize that if feedback of cat2 being sent doesn't not get an answer then it should stop doing so. Is this complete? Comments?



> 
> No, I think you are missing one thing where the DLRR blocks 
> can become 
> enormous. If you are part of an ASM group with 100 
> participants and they 
> all send the Receiver Timestamp block in there RTCP reports. Normally 
> you will send on average at the same pace as all others. 
> Therefore your 
> DLRR response will contain 100 sources. As everybody is doing 
> this the 
> average RTCP size will be large.  Therefore the usage of  
> option 1 below 
> seems to be limited to small groups. When using option 2 only 
> a few out 
> of all sending the Receiver Timestamp block actually answers with the 
> delay information. Therefore only the sender reporting will 
> be effected 
> by this.
> 

Ok, I was missing that.


>  I hesitant to write policy that actually is application or profile 
> dependent. However defining a default behavior I find 
> acceptable with a 
> side note that this can be replaced by application or profile 
> specifications.
> 
Yes , the last sentence reflects my intention.

cheers,


JosÃ©



_______________________________________________
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To: Timur Friedman <timur.friedman@lip6.fr>
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Subject: Re: [AVT] SDP signaling for RTCP XR format
References: <200303311546.h2VFkivm002320@purple.nge.isi.edu> <00cc01c2f7b7$dd92daa0$a648e384@lip6.fr>
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Hi

See below:

Timur Friedman wrote:

>Colin Perkins wrote:
>
>  
>
>>--> Magnus Westerlund writes:
>>...
>>    
>>
>>>>2.- Just curious, why are the report blocks used without signaling? It
>>>>would be helpful to include the reason for this in the draft (if it's
>>>>        
>>>>
>not
>  
>
>>>>there already and I missed it)
>>>>        
>>>>
>>>As I understand it the use case for having report blocks without
>>>signaling is that you know that you are using application foo. Foo has
>>>defined that we will use certain report blocks. Then it is not necessary
>>>to use the signaling, which may in fact not exist. Certain report blocks
>>>may be used to discover if the other end parts understand them. For
>>>example the RTT measurement can be tried a few times to see if any
>>>receiver responds to them.
>>>      
>>>
>>RTP allows applications to send new types of RTCP packet, even if these
>>    
>>
>are
>  
>
>>not explicitly signalled. RTCP XR shouldn't impose additional
>>    
>>
>restrictions.
>
>The draft will be sure to specify that SDP signalling is an optional
>mechanism, to avoid any confusion.
>
>- Timur
>
>  
>
Yes, there is a couple of different approaches here.

I agree that the usage of SDP signaling is optional. And the usage needs 
to be decided on application level. However I think that if your 
application is using SDP signaling, the signaling needs to be followed. 
Otherwise it is impossible to have meaningful signaling. For example 
take the RTT functionality. If you understand it and the arranger of a 
multicast session decides that only senders SHALL or SHOULD respond with 
DLRR packets. Unless the signaling allows us to use normative language 
to that effect the whole signaling is useless. Sending DLRR packets from 
all receiver is problem in multicast groups. Therefore there is need for 
using SHOULD or SHALL in the signaling description.

However there seem to be need to clarify that the usage of the signaling 
is optional and depends on the application.


Any comments on this?

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
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avt@ietf.org
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From mailnull@www1.ietf.org  Wed Apr  2 06:47:22 2003
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Subject: Re: [AVT] SDP signaling for RTCP XR format
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Hi Jose,

Comments below:

Jose Rey wrote:

>Hi Magnus, 
>
>
>please see my asnwers inline, I cut much of the text for readibility.
>
>  
>
>>Yes, I will try to clarify this. My intention is that the SDP 
>>parameters 
>>should be used if there is SDP signaling available for this 
>>application. 
>>The one exception is that application  foo always uses these reports. 
>>And since you are running the client for application foo it 
>>knows that 
>>it shall use the report formats.
>>
>>    
>>
>>>To solve this, why not declare some kind of attribute in SDP 
>>>      
>>>
>>that defines the set of feedback messages that the 
>>application needs of?
>>    
>>
>>You mean in the general case for like SDES information types, etc.?
>>    
>>
>
>I think Colin's comment solves this question:
>
>
>	> RTP allows applications to send new types of RTCP packet, 
>	> even if these are
>	> not explicitly signalled. RTCP XR shouldn't impose additional 
>	> restrictions.
>	> 
>	> Colin
>
>---
>---
>
>As to the behaviour in O/A model, I went over it again and this is what I got:
>
>* answers to offers of cat1-params (sendonly from O->A): if present in response -> SHOULD be sent; if not present -> MAY be sent, whereby
> - this last MAY allows for undeclared use of any extension and, 
> - the SHOULD is based on the supposition that if both the offerer and answerer declare a cat1-param it is because it should be better(or even necessary) for the streaming session and will be used, otherwise the A could omit it. 
>  
>

Yes, I can agree with this. There is no significant harm in sending XR 
blocks that are not desired. However I think one should have a good 
motivation why one would not signal this. I would like to have some 
cautioning wording regarding the MAY. That one should have some reason 
why one sends them after all.

>* answers to offers of cat2-params (sendonly from O->A): whether present in answer or not-> MAY be sent; 
>- this is more flexible since the decision (and probably the benefit) on whether to use cat2-params lies at the receiver. A MUST would avoid sending an undeclared extension and also it cannot be assumed the sender is reporting on all extensions it supports. 
>  
>

I think one should make a firm distinction here. If one uses the SDP 
signaling and declares a cat-2 functionality in a offer, then the 
answerer knows that it supports the functionality. If not included a 
answerer MAY try but the situation is unknown. There is also need for a 
receiver of SDP to follow any declaration if present. To avoid this miss 
use of RTCP bandwidth any receiver of the SDP MUST follow the 
declaration that DLRR blocks SHALL NOT be sent.

>Moreover, it should be stated that answerer must have some mechanism to realize that if feedback of cat2 being sent doesn't not get an answer then it should stop doing so. Is this complete? Comments?
>
>  
>
I agree, however I think this belongs in the section on the XR block 
definition. Stop sending timestamp reports if no one is answering. And 
don't send DLRR blocks unless someone is sending timestamp reports.

>  
>
I think we are converging on intentions and what needs to be clarified. 
I will updated the text a republish it. However I suffer from a cold and 
do not know how much I can work. I will do my best to provide an 
 updated text as soon as possible.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Wed Apr  2 07:40:38 2003
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Date: Wed, 02 Apr 2003 14:40:48 +0200
From: Jose Rey <rey@panasonic.de>
Subject: RE: [AVT] SDP signaling for RTCP XR format
In-reply-to: <3E8ACDDA.4060207@era.ericsson.se>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Hi Magnus,

see inline,

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
> Sent: Wednesday, April 02, 2003 1:48 PM
> To: Jose Rey
> Cc: IETF AVT WG
> Subject: Re: [AVT] SDP signaling for RTCP XR format
> 
> 
> Hi Jose,
> 
> Comments below:
> 
> Jose Rey wrote:
> 
> >Hi Magnus, 
> >
> >
> >please see my asnwers inline, I cut much of the text for readibility.
> >
> >  
> >
> >>Yes, I will try to clarify this. My intention is that the SDP 
> >>parameters 
> >>should be used if there is SDP signaling available for this 
> >>application. 
> >>The one exception is that application  foo always uses 
> these reports. 
> >>And since you are running the client for application foo it 
> >>knows that 
> >>it shall use the report formats.
> >>
> >>    
> >>
> >>>To solve this, why not declare some kind of attribute in SDP 
> >>>      
> >>>
> >>that defines the set of feedback messages that the 
> >>application needs of?
> >>    
> >>
> >>You mean in the general case for like SDES information types, etc.?
> >>    
> >>
> >
> >I think Colin's comment solves this question:
> >
> >
> >	> RTP allows applications to send new types of RTCP packet, 
> >	> even if these are
> >	> not explicitly signalled. RTCP XR shouldn't impose additional 
> >	> restrictions.
> >	> 
> >	> Colin
> >
> >---
> >---
> >
> >As to the behaviour in O/A model, I went over it again and 
> this is what I got:
> >
> >* answers to offers of cat1-params (sendonly from O->A): if 
> present in response -> SHOULD be sent; if not present -> MAY 
> be sent, whereby
> > - this last MAY allows for undeclared use of any extension and, 
> > - the SHOULD is based on the supposition that if both the 
> offerer and answerer declare a cat1-param it is because it 
> should be better(or even necessary) for the streaming session 
> and will be used, otherwise the A could omit it. 
> >  
> >
> 
> Yes, I can agree with this. There is no significant harm in 
> sending XR 
> blocks that are not desired. However I think one should have a good 
> motivation why one would not signal this. I would like to have some 
> cautioning wording regarding the MAY. That one should have 
> some reason 
> why one sends them after all.
> 
> >* answers to offers of cat2-params (sendonly from O->A): 
> whether present in answer or not-> MAY be sent; 
> >- this is more flexible since the decision (and probably the 
> benefit) on whether to use cat2-params lies at the receiver. 
> A MUST would avoid sending an undeclared extension and also 
> it cannot be assumed the sender is reporting on all 
> extensions it supports. 
> >  
> I think one should make a firm distinction here. If one uses the SDP 
> signaling and declares a cat-2 functionality in a offer, then the 
> answerer knows that it supports the functionality. If not included a 
> answerer MAY try but the situation is unknown. There is also 
> need for a 
> receiver of SDP to follow any declaration if present. To 
> avoid this miss 
> use of RTCP bandwidth any receiver of the SDP MUST follow the 
> declaration that DLRR blocks SHALL NOT be sent.
>
I am not sure I understand this paragraph. You want to prohibit it if not present in the offer? For that we should require that O always lists ALL of the extensions it supports, so that it makes no sense for A to request a cat-2 parameter. Makes this requirement sense? Also for other future cat-2 parameters? If 
 
> >Moreover, it should be stated that answerer must have some 
> mechanism to realize that if feedback of cat2 being sent 
> doesn't not get an answer then it should stop doing so. Is 
> this complete? Comments?
> >
> >  
> >
> I agree, however I think this belongs in the section on the XR block 
> definition. Stop sending timestamp reports if no one is 
> answering. And 
> don't send DLRR blocks unless someone is sending timestamp reports.
> 

Yes.

> >  
> >
> I think we are converging on intentions and what needs to be 
> clarified. 
> I will updated the text a republish it. However I suffer from 
> a cold and 
> do not know how much I can work. I will do my best to provide an 
>  updated text as soon as possible.

Don't worry, just get well first :)

cheers,

JosÃ©

> 
> Best Regards
> 
> Magnus Westerlund 
> 
> Multimedia Technologies, Ericsson Research ERA/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
> 
> 
> 

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Timur Friedman" <timur.friedman@lip6.fr>
To: "Magnus Westerlund" <magnus.westerlund@era.ericsson.se>
Cc: "IETF AVT WG" <avt@ietf.org>
References: <200303311546.h2VFkivm002320@purple.nge.isi.edu> <00cc01c2f7b7$dd92daa0$a648e384@lip6.fr> <3E8ACA89.90000@era.ericsson.se>
Subject: Re: [AVT] SDP signaling for RTCP XR format
Date: Wed, 2 Apr 2003 21:24:25 +0200
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Magnus Westerlund wrote, quoting Timur Friedman :

> >The draft will be sure to specify that SDP signalling is an optional
> >mechanism, to avoid any confusion.
> >
> >- Timur
> >
> >
> >
> Yes, there is a couple of different approaches here.
>
> I agree that the usage of SDP signaling is optional. And the usage needs
> to be decided on application level. However I think that if your
> application is using SDP signaling, the signaling needs to be followed.

Agreed that if the application is using SDP signalling and XR packets then
it must adhere to the SDP signalling mechanism defined for XR packets.
We'll be sure to state this.

- Timur Friedman


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Subject: Re: [AVT] SDP signaling for RTCP XR format
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Jose Rey wrote, quoting Magnus Westerlund:

> > I agree, however I think this belongs in the section on the XR block
> > definition. Stop sending timestamp reports if no one is
> > answering. And
> > don't send DLRR blocks unless someone is sending timestamp reports.
> >
>
> Yes.

How would we define the time after which an application, having received no
DLRRs, should stop sending Timestamp Report blocks?  The time period seems
to me highly application dependent, and so best left out of the document.

Agreed that a DLRR block makes no sense in the absence of a Timestamp Report
block.

- Timur Friedman

_______________________________________________
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Subject: Re: [AVT] SDP signaling for RTCP XR format
Date: Wed, 2 Apr 2003 21:34:24 +0200
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Magnus Westerlund wrote:

> Here one issue actually surface. Is it the right thing to restrict a
> receiver to not send DLRR report block unless the usage has been
> explicitly signaled in multicast? The reason why I see a need to
> restrict it is the potential to create enormous report blocks in large
> multicast groups. And the only result is that the different receivers
> will determine there RTT to other receivers which seems to be rather
> pointless in most application. It seems that you would in fact like to
> control this operation to a few different operation modes in multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
>
> Should we define such a distinction in the signaling?

I agree that the signalling should allow each of these modes.  But also the
mode in which certain receivers calculate RTTs amongst themselves.  While
many applications might not use this mode, we do not want to unnecessarily
rule it out.

Here are two examples of when the mode could be used.  First, receivers
might measure RTTs amongst themselves as a prelude to some of them becoming
senders (multicast "repair servers", retansmitting lost packets, for
instance). Second, a network monitoring tool might wish to measure its RTT
to certain receivers.

- Timur Friedman

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Thu Apr  3 13:30:36 2003
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Subject: [AVT] MWPP and the marker bit ...
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Hi everyone,

	Looking for input on the following MWPP issue ...

From the draft-ietf-avt-mpeg4-simple-07.txt:

   Marker (M) bit: The M bit is set to 1 to indicate that the RTP 
   packet payload contains either the final fragment of a fragmented 
   Access Unit or one or more complete Access Units.

MWPP needs 1-packet-per-access-unit semantics to work properly, and so
MWPP defines the M bit to always 1 for mpeg4-generic use.

Up until now, I've also defined the M bit to always be 1 for "native
MWPP streams" (mwpp MIME type), on the theory the confusion that would
come from having two different M bit uses for the different MIME types
outweighs the advantage of using the M bit for something useful.

However, the recent discussion on the list over voice-codec M bit use
has made me wonder if I should instead define the M bit for the mwpp
MIME type to have a meaning akin to the voice-spurt M bit meaning.
For mwpp, this semantic would be "any packet that codes at least one
new MIDI command", and the purpose would be to let receivers quickly
distinguish empty "guard packets" from packets with new data.

Any input one way or the other is appreciated -- I'm undecided at this
point ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------



_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi everyone,

	Looks like my mailer truncated the posting I just sent to the
list, let's try again ...

---

Hi everyone,

	Looking for input on the following MWPP issue ...

	MWPP supports two MIME types: mpeg4-generic for MPEG work
and mwpp for general use. From draft-ietf-avt-mpeg4-simple-07.txt:

  Marker (M) bit: The M bit is set to 1 to indicate that the RTP
  packet payload contains either the final fragment of a fragmented 
  Access Unit or one or more complete Access Units.

	MWPP needs "one access unit per packet" semantics for
the recovery journal to work, as so for mpeg4-generic use M
is defined to 1 for all packets.

	Currently, M is also defined to 1 for all packets for
use with the mwpp MIME type, on the theory that the confusion
that would result from two separate M bit definitions is best
avoided.

	However, reading over the recent list discussion on
voice codec marker bit use, I'm wondering if I should reverse
this decision, and define M bit semantics for the mwpp MIME type
that are in the spirit of voice-codec use (for MWPP, the right
definition is to use the M bit to differentiate packets with
new MIDI data from empty "guard" packets).

	Any input on this decision is appreciated, I'm undecided
at the point ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: [AVT] Protocol Action: Enhanced Compressed RTP (CRTP) for links with
 high delay, packet loss and reordering to Proposed Standard
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The IESG has approved the Internet-Draft 'Enhanced Compressed RTP
(CRTP) for Links with High Delay, Packet Loss and Reordering'
<draft-ietf-avt-crtp-enhance-07.txt> as a Proposed Standard. This
document is the product of the Audio Video Transport Working Group.
The IESG contact persons are Allison Mankin and Jon Peterson.

 Technical Summary

 This document describes a header compression scheme for point to point
 links with packet loss and long delays. It is based on Compressed
 Real-time Transport Protocol (CRTP), the IP/UDP/RTP header compression
 described in RFC 2508. Original CRTP does not perform well on such
 links: packet loss results in context corruption and due to the long
 delay, excessive packets are discarded before the context is
 repaired. To correct the behavior of CRTP over such links, a few
 extensions to the protocol are specified here. The extensions aim to
 reduce context corruption by changing the way the compressor updates
 the context at the decompressor: updates are repeated and include
 updates to full and differential context parameters. With these
 extensions, ECRTP performs well over links with packet loss, packet
 reordering and long delays.

 The distinction is made that ECRTP is robust for links with reordering
 such as tunnels and multihop, and it is also robust for cellular links
 with high loss but that ROHC (RFC 3095) is expected "to be the preferred
 compression mechanism over links where compression efficiency is important"
 and packet misordering is infrequent.

 Working Group Summary

 The working group supported advancement of this document. There were
 no Last Call concerns with the document, but Area Director comments
 resulted in clarifications on IPv6, Security considerations, and applicability.

 Protocol Quality/Review

 This document was reviewed for the IESG by Allison Mankin.
 The Transport Area seeks information about implementations of ECRTP and
 their performance.
_______________________________________________
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Date: Fri, 04 Apr 2003 18:36:33 -0800
To: thh@dolby.com, jasonfl@microsoft.com
From: Ross Finlayson <finlayson@live.com>
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Subject: [AVT] Some comments on <draft-flaks-avt-rtp-ac3-03.txt>
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I have recently been reviewing (& implementing) the proposed RTP payload 
format for AC-3 Audio <draft-flaks-avt-rtp-ac3-03.txt>, and I have a number 
of comments:

1/ This is now an official AVT Working Group item, correct?  If so, the 
file name of future drafts should have the form "draft-avt-rtp-ac3-*.txt".

2/ I don't see any real need for the "NDU" (number of data units) byte at 
the front of each RTP payload.  If an incoming RTP packet contains more 
than one 'data unit' (frame), then the receiver already needs to figure out 
the frame boundaries within the packet (presumably by computing the size of 
each frame from its internal header bytes).  In doing this, he can count 
how many frames are present - but knowing this count in advance provides no 
useful information.

3/ The rules for how fragmented frames are packed into RTP packets are not 
made clear.  In particular, can a RTP packet legally contain the following:
         - n (>=1) complete frames, followed by an initial fragment of a
                 following frame?
or
         - the trailing fragment of a frame, followed by one or more complete
                 frames (or the initial fragment of a following frame)?
In most existing RTP payload formats, to simplify the packing rules, 
neither of these cases is allowed.  I.e., most RTP payload formats (that 
support multiple frames per packet, with fragmentation) specify that a RTP 
packet must contain either (i) one or more complete frames, or (ii) exactly 
one partial fragment of a frame (and nothing more).  The document should 
explain exactly what the packing rules are for this payload format.

4/ What value should the "RDT" field (in the Data Unit Header) take if the 
data unit does not contain any redundant data (i.e., if the "TYP" field is 
00)?  Should "RDT" be 000 in this case?  (It probably doesn't matter, but 
it should be specified.)

In general, it's not clear which of the 4x8=32 possible combinations of the 
(2-bit) "TYP" field and the (3-bit) "RDT" field are actually legal.

5/ What value should the "B" bit have if the 'data unit' is a complete 
frame (i.e., not fragmented)?  Presumably the "B" bit should be 1 in this 
case (because it will, of course, contain the 5/8 point).  But the 
description of "B" is a little ambiguous, because it says "set to 1 if the 
packet contains an AC-3
*fragment* consisting of..."

6/ The "F" bit, as currently specified, seems mostly useless.  Consider, 
for example, the case where a single frame is fragmented over three RTP 
packets.  The first packet has
         F==1, M==0
The second packet has
         F==1, M==0
The third packet has
         F==1, M==1
(where M is the RTP header's "M" bit.)  If the first of these packets gets 
lost, then the receiver will not be able to tell whether the second packet 
is the start of a new frame, or (as is actually the case) the middle of a 
frame.  (I suppose it could figure this out by checking for the AC3 
'syncword' at the start, but then you wouldn't really need the "F" bit at all.)

I suggest that you replace the "F" bit with a "S" (start of frame) 
bit.  This bit would be set to 1 iff the 'data unit' is the start of a 
frame (or a complete frame).  I.e., using the example above, the first 
packet would have
         S==1, M==0
The second packet would have
         S==0, M==0
The third packet would have
         S==0, M==1
(The fourth possible case
         S==1, M==1
would indicate that the packet contains (one or more) *complete* frames.)

7/ The "T" (time code present) bit doesn't seem particularly useful.  It 
requires that a sending implementation have detailed knowledge of the 
internals of AC3 frames, which seems unnecessary.  Instead, I suggest 
omitting this bit, and leaving it up to the receiver to figure out - from 
the internals of each frame - whether it contains a time code (because this 
is something that it presumably needs to do anyway).

8/ As noted in 2/ above, each receiver needs to figure out where the 
boundaries between 'data units' are in each incoming RTP packet.  Can the 
receiver easily do this even when the data unit(s) contain 
redundant/interleaved data - i.e., if "TYP" is 01 or 10?  This wasn't clear 
to me, and I think you should provide a more detailed explanation (and 
perhaps even some 'pseudo-code') that describes how a receiver should parse 
incoming packets.

9/ I found the definition of the '5/8 point' a little confusing:
         "...the 5/8ths point is defined as:
                 5/8-framesize = truncate(framesize/2) + truncate(framesize/8)"
When I first saw this, I thought that "5/8-framesize" was part of the 
formula - i.e., with a division and a subtraction operation!  Instead, I 
suggest changing the definition to:
         "...the 5/8ths point, P, is defined as:
                 P = truncate(framesize/2) + truncate(framesize/8)"

10/ Re. Section 4.2: "SDP usage": Having the number of channels be 
specified in a separate "a=fmtp:" line is unorthodox.  It's far more common 
(and actually suggested in the SDP specification) for the number of 
channels to be specified at the end of the "a=rtpmap:" line - e.g.:
         m=audio 49000 RTP/AVP 100
         a=rtpmap:100 ac3/48000/6

I hope this helps...

         Ross.



_______________________________________________
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Subject: [AVT] I-D ACTION:draft-ietf-avt-rtp-retransmission-07.txt
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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 Retransmission Payload Format
	Author(s)	: J. Rey et al.
	Filename	: draft-ietf-avt-rtp-retransmission-07.txt
	Pages		: 27
	Date		: 2003-4-4
	
RTP retransmission is an effective packet loss recovery technique 
for real-time applications with relaxed delay bounds.  This 
document describes an RTP payload format for performing 
retransmissions.  Retransmitted RTP packets are sent in a separate 
stream from the original RTP stream.  It is assumed that feedback 
from receivers to senders is available.  In particular, it is 
assumed that RTCP feedback as defined in the extended RTP profile 
for RTCP-based feedback (denoted RTP/AVPF), is available in this 
memo.

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Subject: [AVT] I-D ACTION:draft-ietf-avt-rtp-clearmode-00.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		: RTP payload format for a 64 kbit/s transparent call
	Author(s)	: R. Kreuter
	Filename	: draft-ietf-avt-rtp-clearmode-00.txt
	Pages		: 7
	Date		: 2003-4-7
	
This document describes how to carry 64 kbit/s data streams trans-
parently in RTP packets, using a pseudo-codec called 'Clearmode'. 
It also serves as registration for a related MIME type called 
'audio/clearmode'. 
'Clearmode' is a basic feature of VoIP media gateways.

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Hi

There are some applications which define the CLEARMODE name as CCD. Are there also others which use CLEARMODE? If no, can the name be changed to CCD?

Nurit

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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 a 64 kbit/s transparent call
	Author(s)	: R. Kreuter
	Filename	: draft-ietf-avt-rtp-clearmode-00.txt
	Pages		: 7
	Date		: 2003-4-7
	
This document describes how to carry 64 kbit/s data streams trans-
parently in RTP packets, using a pseudo-codec called 'Clearmode'. 
It also serves as registration for a related MIME type called 
'audio/clearmode'. 
'Clearmode' is a basic feature of VoIP media gateways.

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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
To: "'Nurit Shenhar'" <NuritS@audiocodes.com>,
        Kreuter Ruediger
	 <ruediger.kreuter@siemens.com>
Cc: avt@ietf.org
Subject: AW: [AVT] I-D ACTION:draft-ietf-avt-rtp-clearmode-00.txt
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Hi,

what are these applications ?

Ruediger

> -----Ursprüngliche Nachricht-----
> Von: Nurit Shenhar [mailto:NuritS@audiocodes.com]
> Gesendet: Dienstag, 8. April 2003 18:09
> An: ruediger.kreuter@siemens.com
> Cc: avt@ietf.org
> Betreff: RE: [AVT] I-D ACTION:draft-ietf-avt-rtp-clearmode-00.txt
> 
> 
> Hi
> 
> There are some applications which define the CLEARMODE name 
> as CCD. Are there also others which use CLEARMODE? If no, can 
> the name be changed to CCD?
> 
> Nurit
> 
> -----Original Message-----
> From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: Tuesday, April 08, 2003 4:04 PM
> Cc: avt@ietf.org
> Subject: [AVT] I-D ACTION:draft-ietf-avt-rtp-clearmode-00.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 a 64 kbit/s 
> transparent call
> 	Author(s)	: R. Kreuter
> 	Filename	: draft-ietf-avt-rtp-clearmode-00.txt
> 	Pages		: 7
> 	Date		: 2003-4-7
> 	
> This document describes how to carry 64 kbit/s data streams trans-
> parently in RTP packets, using a pseudo-codec called 'Clearmode'. 
> It also serves as registration for a related MIME type called 
> 'audio/clearmode'. 
> 'Clearmode' is a basic feature of VoIP media gateways.
> 
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-clearmo
de-00.txt

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From mailnull@www1.ietf.org  Fri Apr 11 17:14:50 2003
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To: "'Timur Friedman'" <timur.friedman@lip6.fr>,
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Subject: RE: [AVT] SDP signaling for RTCP XR format
Date: Fri, 11 Apr 2003 17:04:14 -0400
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Timur, Colin, Steve

At the recent IETF meeting we agreed that the existing draft was getting fairly stable - there were some good detail points brought up during the AVT discussion however these were easily accomodated.  

We also agreed at the AVT meeting to include some simple SDP signaling at Magnus' request however felt at the time that this would be fairly straightforward and would not take much time.  The discussion on SDP appears to be getting much more involved than anticipated and there is no apparent end in sight.  

I would like to propose that we move to a final draft without SDP in order that we can reach closure.  In San Francisco we discussed the idea of creating a separate document to describe the use of SDP for negotiating reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was suggested precisely because of some concern that the negotiation mechanism could get complex and that there were a wider variety of scenarios than anticipated .. which appears to be the case.

Regards

Alan Clark


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
Friedman
Sent: Wednesday, April 02, 2003 2:34 PM
To: Magnus Westerlund; Jose Rey
Cc: IETF AVT WG
Subject: Re: [AVT] SDP signaling for RTCP XR format


Magnus Westerlund wrote:

> Here one issue actually surface. Is it the right thing to restrict a
> receiver to not send DLRR report block unless the usage has been
> explicitly signaled in multicast? The reason why I see a need to
> restrict it is the potential to create enormous report blocks in large
> multicast groups. And the only result is that the different receivers
> will determine there RTT to other receivers which seems to be rather
> pointless in most application. It seems that you would in fact like to
> control this operation to a few different operation modes in multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
>
> Should we define such a distinction in the signaling?

I agree that the signalling should allow each of these modes.  But also the
mode in which certain receivers calculate RTTs amongst themselves.  While
many applications might not use this mode, we do not want to unnecessarily
rule it out.

Here are two examples of when the mode could be used.  First, receivers
might measure RTTs amongst themselves as a prelude to some of them becoming
senders (multicast "repair servers", retansmitting lost packets, for
instance). Second, a network monitoring tool might wish to measure its RTT
to certain receivers.

- Timur Friedman

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] RTCP XR 
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Thread-Topic: [AVT] SDP signaling for RTCP XR format
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From: "Gamze Seckin" <gamze@vidiator.com>
To: "IETF AVT WG" <avt@ietf.org>
Cc: <alan@telchemy.com>, "Timur Friedman" <timur.friedman@lip6.fr>,
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hi,

Regarding the draft rtcp xr, has there been an attempt/discussion to include any video/audio streaming quality metrics?

-gamze

-----Original Message-----
From: Alan Clark [mailto:alan@telchemy.com]
Sent: Friday, April 11, 2003 2:04 PM
To: 'Timur Friedman'; 'Colin Perkins'; 'Stephen Casner'; 'Magnus
Westerlund'
Cc: 'IETF AVT WG'
Subject: RE: [AVT] SDP signaling for RTCP XR format


Timur, Colin, Steve

At the recent IETF meeting we agreed that the existing draft was getting fairly stable - there were some good detail points brought up during the AVT discussion however these were easily accomodated.  

We also agreed at the AVT meeting to include some simple SDP signaling at Magnus' request however felt at the time that this would be fairly straightforward and would not take much time.  The discussion on SDP appears to be getting much more involved than anticipated and there is no apparent end in sight.  

I would like to propose that we move to a final draft without SDP in order that we can reach closure.  In San Francisco we discussed the idea of creating a separate document to describe the use of SDP for negotiating reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was suggested precisely because of some concern that the negotiation mechanism could get complex and that there were a wider variety of scenarios than anticipated .. which appears to be the case.

Regards

Alan Clark


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
Friedman
Sent: Wednesday, April 02, 2003 2:34 PM
To: Magnus Westerlund; Jose Rey
Cc: IETF AVT WG
Subject: Re: [AVT] SDP signaling for RTCP XR format


Magnus Westerlund wrote:

> Here one issue actually surface. Is it the right thing to restrict a
> receiver to not send DLRR report block unless the usage has been
> explicitly signaled in multicast? The reason why I see a need to
> restrict it is the potential to create enormous report blocks in large
> multicast groups. And the only result is that the different receivers
> will determine there RTT to other receivers which seems to be rather
> pointless in most application. It seems that you would in fact like to
> control this operation to a few different operation modes in multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
>
> Should we define such a distinction in the signaling?

I agree that the signalling should allow each of these modes.  But also the
mode in which certain receivers calculate RTTs amongst themselves.  While
many applications might not use this mode, we do not want to unnecessarily
rule it out.

Here are two examples of when the mode could be used.  First, receivers
might measure RTTs amongst themselves as a prelude to some of them becoming
senders (multicast "repair servers", retansmitting lost packets, for
instance). Second, a network monitoring tool might wish to measure its RTT
to certain receivers.

- Timur Friedman

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "'Gamze Seckin'" <gamze@vidiator.com>, "'IETF AVT WG'" <avt@ietf.org>
Subject: RE: [AVT] RTCP XR 
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The VoIP metrics could be applied to streaming video and audio.  For
streaming audio you would basically ignore the effects of delay and =
focus on
listening quality metrics. =20

For streaming video the same approach can be used however it may be
advantageous to set the gmin value to say 128, due to the increased
sensitivity of video to packet loss.  There was some suggestion at the =
IETF
meeting to use the jitter buffer levels to provide feedback to the
transmitter and help to manage packet flows.

Regards

Alan Clark
Telchemy


-----Original Message-----
From: Gamze Seckin [mailto:gamze@vidiator.com]
Sent: Friday, April 11, 2003 5:40 PM
To: IETF AVT WG
Cc: alan@telchemy.com; Timur Friedman; Colin Perkins; Stephen Casner;
Magnus Westerlund
Subject: RE: [AVT] RTCP XR=20



hi,

Regarding the draft rtcp xr, has there been an attempt/discussion to =
include
any video/audio streaming quality metrics?

-gamze

-----Original Message-----
From: Alan Clark [mailto:alan@telchemy.com]
Sent: Friday, April 11, 2003 2:04 PM
To: 'Timur Friedman'; 'Colin Perkins'; 'Stephen Casner'; 'Magnus
Westerlund'
Cc: 'IETF AVT WG'
Subject: RE: [AVT] SDP signaling for RTCP XR format


Timur, Colin, Steve

At the recent IETF meeting we agreed that the existing draft was getting
fairly stable - there were some good detail points brought up during the =
AVT
discussion however these were easily accomodated. =20

We also agreed at the AVT meeting to include some simple SDP signaling =
at
Magnus' request however felt at the time that this would be fairly
straightforward and would not take much time.  The discussion on SDP =
appears
to be getting much more involved than anticipated and there is no =
apparent
end in sight. =20

I would like to propose that we move to a final draft without SDP in =
order
that we can reach closure.  In San Francisco we discussed the idea of
creating a separate document to describe the use of SDP for negotiating
reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was
suggested precisely because of some concern that the negotiation =
mechanism
could get complex and that there were a wider variety of scenarios than
anticipated .. which appears to be the case.

Regards

Alan Clark


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
Friedman
Sent: Wednesday, April 02, 2003 2:34 PM
To: Magnus Westerlund; Jose Rey
Cc: IETF AVT WG
Subject: Re: [AVT] SDP signaling for RTCP XR format


Magnus Westerlund wrote:

> Here one issue actually surface. Is it the right thing to restrict a
> receiver to not send DLRR report block unless the usage has been
> explicitly signaled in multicast? The reason why I see a need to
> restrict it is the potential to create enormous report blocks in large
> multicast groups. And the only result is that the different receivers
> will determine there RTT to other receivers which seems to be rather
> pointless in most application. It seems that you would in fact like to
> control this operation to a few different operation modes in =
multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
>
> Should we define such a distinction in the signaling?

I agree that the signalling should allow each of these modes.  But also =
the
mode in which certain receivers calculate RTTs amongst themselves.  =
While
many applications might not use this mode, we do not want to =
unnecessarily
rule it out.

Here are two examples of when the mode could be used.  First, receivers
might measure RTTs amongst themselves as a prelude to some of them =
becoming
senders (multicast "repair servers", retansmitting lost packets, for
instance). Second, a network monitoring tool might wish to measure its =
RTT
to certain receivers.

- Timur Friedman

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Fri Apr 11 17:57:56 2003
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Subject: RE: [AVT] RTCP XR 
Date: Fri, 11 Apr 2003 14:54:54 -0700
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Thread-Topic: [AVT] RTCP XR 
Thread-Index: AcMAct9CKXkxbX8ST5KZuKDDiFLk0AAAUfdQ
From: "Gamze Seckin" <gamze@vidiator.com>
To: <alan@telchemy.com>, "IETF AVT WG" <avt@ietf.org>
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I agree, however there could be more than loss to determine streaming quality, especially if the receiver is a mobile handset for example. You would want (if possible) codec specific information (like freezes, corruptions) or number of buffering etc. It could be also a good idea to separate this information from the VoIP block since the report-trigger requirements could be different, and VoIP block has much more than the streaming block would need. 

-thanks
-g

Gamze Seckin, Ph.D. 
Director of Research 
Vidiator Technology (US) Inc., The multimedia technology company of Hutchison Whampoa Limited 
office:650.390.9989 x27 fax:650.390.9911 
This e-mail is a confidential communication. If you are not the intended recipient of this e-mail, please return the message to the sender and delete the original. Thank you.



-----Original Message-----
From: Alan Clark [mailto:alan@telchemy.com]
Sent: Friday, April 11, 2003 2:40 PM
To: Gamze Seckin; 'IETF AVT WG'
Subject: RE: [AVT] RTCP XR 



The VoIP metrics could be applied to streaming video and audio.  For streaming audio you would basically ignore the effects of delay and focus on listening quality metrics.  

For streaming video the same approach can be used however it may be advantageous to set the gmin value to say 128, due to the increased sensitivity of video to packet loss.  There was some suggestion at the IETF meeting to use the jitter buffer levels to provide feedback to the transmitter and help to manage packet flows.

Regards

Alan Clark
Telchemy


-----Original Message-----
From: Gamze Seckin [mailto:gamze@vidiator.com]
Sent: Friday, April 11, 2003 5:40 PM
To: IETF AVT WG
Cc: alan@telchemy.com; Timur Friedman; Colin Perkins; Stephen Casner;
Magnus Westerlund
Subject: RE: [AVT] RTCP XR 



hi,

Regarding the draft rtcp xr, has there been an attempt/discussion to include any video/audio streaming quality metrics?

-gamze

-----Original Message-----
From: Alan Clark [mailto:alan@telchemy.com]
Sent: Friday, April 11, 2003 2:04 PM
To: 'Timur Friedman'; 'Colin Perkins'; 'Stephen Casner'; 'Magnus
Westerlund'
Cc: 'IETF AVT WG'
Subject: RE: [AVT] SDP signaling for RTCP XR format


Timur, Colin, Steve

At the recent IETF meeting we agreed that the existing draft was getting fairly stable - there were some good detail points brought up during the AVT discussion however these were easily accomodated.  

We also agreed at the AVT meeting to include some simple SDP signaling at Magnus' request however felt at the time that this would be fairly straightforward and would not take much time.  The discussion on SDP appears to be getting much more involved than anticipated and there is no apparent end in sight.  

I would like to propose that we move to a final draft without SDP in order that we can reach closure.  In San Francisco we discussed the idea of creating a separate document to describe the use of SDP for negotiating reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was suggested precisely because of some concern that the negotiation mechanism could get complex and that there were a wider variety of scenarios than anticipated .. which appears to be the case.

Regards

Alan Clark


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
Friedman
Sent: Wednesday, April 02, 2003 2:34 PM
To: Magnus Westerlund; Jose Rey
Cc: IETF AVT WG
Subject: Re: [AVT] SDP signaling for RTCP XR format


Magnus Westerlund wrote:

> Here one issue actually surface. Is it the right thing to restrict a
> receiver to not send DLRR report block unless the usage has been
> explicitly signaled in multicast? The reason why I see a need to
> restrict it is the potential to create enormous report blocks in large
> multicast groups. And the only result is that the different receivers
> will determine there RTT to other receivers which seems to be rather
> pointless in most application. It seems that you would in fact like to
> control this operation to a few different operation modes in multicast.
> 1. fully meshed, both senders and receivers return DLRR blocks.
> 2. Sender only, only senders return DLRR blocks.
> 3. functionality off, no one responds with DLRR blocks.
>
> Should we define such a distinction in the signaling?

I agree that the signalling should allow each of these modes.  But also the
mode in which certain receivers calculate RTTs amongst themselves.  While
many applications might not use this mode, we do not want to unnecessarily
rule it out.

Here are two examples of when the mode could be used.  First, receivers
might measure RTTs amongst themselves as a prelude to some of them becoming
senders (multicast "repair servers", retansmitting lost packets, for
instance). Second, a network monitoring tool might wish to measure its RTT
to certain receivers.

- Timur Friedman

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] RTCP XR 
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Hello;

Can you give references to the additional metric you have in mind ?

                                  Regards
                                  Marshall Eubanks



On Friday, April 11, 2003, at 05:39 PM, Alan Clark wrote:

>
> The VoIP metrics could be applied to streaming video and audio.  For
> streaming audio you would basically ignore the effects of delay and 
> focus on
> listening quality metrics.
>
> For streaming video the same approach can be used however it may be
> advantageous to set the gmin value to say 128, due to the increased
> sensitivity of video to packet loss.  There was some suggestion at the 
> IETF
> meeting to use the jitter buffer levels to provide feedback to the
> transmitter and help to manage packet flows.
>
> Regards
>
> Alan Clark
> Telchemy
>
>
> -----Original Message-----
> From: Gamze Seckin [mailto:gamze@vidiator.com]
> Sent: Friday, April 11, 2003 5:40 PM
> To: IETF AVT WG
> Cc: alan@telchemy.com; Timur Friedman; Colin Perkins; Stephen Casner;
> Magnus Westerlund
> Subject: RE: [AVT] RTCP XR
>
>
>
> hi,
>
> Regarding the draft rtcp xr, has there been an attempt/discussion to 
> include
> any video/audio streaming quality metrics?
>
> -gamze
>
> -----Original Message-----
> From: Alan Clark [mailto:alan@telchemy.com]
> Sent: Friday, April 11, 2003 2:04 PM
> To: 'Timur Friedman'; 'Colin Perkins'; 'Stephen Casner'; 'Magnus
> Westerlund'
> Cc: 'IETF AVT WG'
> Subject: RE: [AVT] SDP signaling for RTCP XR format
>
>
> Timur, Colin, Steve
>
> At the recent IETF meeting we agreed that the existing draft was 
> getting
> fairly stable - there were some good detail points brought up during 
> the AVT
> discussion however these were easily accomodated.
>
> We also agreed at the AVT meeting to include some simple SDP signaling 
> at
> Magnus' request however felt at the time that this would be fairly
> straightforward and would not take much time.  The discussion on SDP 
> appears
> to be getting much more involved than anticipated and there is no 
> apparent
> end in sight.
>
> I would like to propose that we move to a final draft without SDP in 
> order
> that we can reach closure.  In San Francisco we discussed the idea of
> creating a separate document to describe the use of SDP for negotiating
> reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was
> suggested precisely because of some concern that the negotiation 
> mechanism
> could get complex and that there were a wider variety of scenarios than
> anticipated .. which appears to be the case.
>
> Regards
>
> Alan Clark
>
>
> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
> Friedman
> Sent: Wednesday, April 02, 2003 2:34 PM
> To: Magnus Westerlund; Jose Rey
> Cc: IETF AVT WG
> Subject: Re: [AVT] SDP signaling for RTCP XR format
>
>
> Magnus Westerlund wrote:
>
>> Here one issue actually surface. Is it the right thing to restrict a
>> receiver to not send DLRR report block unless the usage has been
>> explicitly signaled in multicast? The reason why I see a need to
>> restrict it is the potential to create enormous report blocks in large
>> multicast groups. And the only result is that the different receivers
>> will determine there RTT to other receivers which seems to be rather
>> pointless in most application. It seems that you would in fact like to
>> control this operation to a few different operation modes in 
>> multicast.
>> 1. fully meshed, both senders and receivers return DLRR blocks.
>> 2. Sender only, only senders return DLRR blocks.
>> 3. functionality off, no one responds with DLRR blocks.
>>
>> Should we define such a distinction in the signaling?
>
> I agree that the signalling should allow each of these modes.  But 
> also the
> mode in which certain receivers calculate RTTs amongst themselves.  
> While
> many applications might not use this mode, we do not want to 
> unnecessarily
> rule it out.
>
> Here are two examples of when the mode could be used.  First, receivers
> might measure RTTs amongst themselves as a prelude to some of them 
> becoming
> senders (multicast "repair servers", retansmitting lost packets, for
> instance). Second, a network monitoring tool might wish to measure its 
> RTT
> to certain receivers.
>
> - Timur Friedman
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> <winmail.dat>

T.M. Eubanks
Multicast Technologies, Inc.
e-mail : tme@multicasttech.com
http://www.multicasttech.com

  Our New Multicast Workshop :
  http://www.multicasttech.com/workshop
Test your network for multicast :
http://www.multicasttech.com/mt/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Gamze Seckin" <gamze@vidiator.com>
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Hi Marshall and all,

There are not many of them, but definitely worth separating them form the VoIP metrics for efficiency purposes.
We presented a set of parameters in 3gpp sa4 (in sa25 and sa25bis, documents 30008 and 30130, accessible via www.3gpp.org) in an collaborated work with 3 UK. Actually a new version is due end-of this month i can forward you the updated document/link soon. We proposed to send the information back over rtsp , but having an rtcp extension is also a valuable approach, we believe.

-g

Gamze Seckin, Ph.D. 
Director of Research 
Vidiator Technology (US) Inc., The multimedia technology company of Hutchison Whampoa Limited 
office:650.390.9989 x27 fax:650.390.9911 
This e-mail is a confidential communication. If you are not the intended recipient of this e-mail, please return the message to the sender and delete the original. Thank you.



-----Original Message-----
From: Marshall Eubanks [mailto:tme@multicasttech.com]
Sent: Friday, April 11, 2003 3:12 PM
To: alan@telchemy.com
Cc: Gamze Seckin; 'IETF AVT WG'
Subject: Re: [AVT] RTCP XR 


Hello;

Can you give references to the additional metric you have in mind ?

                                  Regards
                                  Marshall Eubanks



On Friday, April 11, 2003, at 05:39 PM, Alan Clark wrote:

>
> The VoIP metrics could be applied to streaming video and audio.  For
> streaming audio you would basically ignore the effects of delay and 
> focus on
> listening quality metrics.
>
> For streaming video the same approach can be used however it may be
> advantageous to set the gmin value to say 128, due to the increased
> sensitivity of video to packet loss.  There was some suggestion at the 
> IETF
> meeting to use the jitter buffer levels to provide feedback to the
> transmitter and help to manage packet flows.
>
> Regards
>
> Alan Clark
> Telchemy
>
>
> -----Original Message-----
> From: Gamze Seckin [mailto:gamze@vidiator.com]
> Sent: Friday, April 11, 2003 5:40 PM
> To: IETF AVT WG
> Cc: alan@telchemy.com; Timur Friedman; Colin Perkins; Stephen Casner;
> Magnus Westerlund
> Subject: RE: [AVT] RTCP XR
>
>
>
> hi,
>
> Regarding the draft rtcp xr, has there been an attempt/discussion to 
> include
> any video/audio streaming quality metrics?
>
> -gamze
>
> -----Original Message-----
> From: Alan Clark [mailto:alan@telchemy.com]
> Sent: Friday, April 11, 2003 2:04 PM
> To: 'Timur Friedman'; 'Colin Perkins'; 'Stephen Casner'; 'Magnus
> Westerlund'
> Cc: 'IETF AVT WG'
> Subject: RE: [AVT] SDP signaling for RTCP XR format
>
>
> Timur, Colin, Steve
>
> At the recent IETF meeting we agreed that the existing draft was 
> getting
> fairly stable - there were some good detail points brought up during 
> the AVT
> discussion however these were easily accomodated.
>
> We also agreed at the AVT meeting to include some simple SDP signaling 
> at
> Magnus' request however felt at the time that this would be fairly
> straightforward and would not take much time.  The discussion on SDP 
> appears
> to be getting much more involved than anticipated and there is no 
> apparent
> end in sight.
>
> I would like to propose that we move to a final draft without SDP in 
> order
> that we can reach closure.  In San Francisco we discussed the idea of
> creating a separate document to describe the use of SDP for negotiating
> reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was
> suggested precisely because of some concern that the negotiation 
> mechanism
> could get complex and that there were a wider variety of scenarios than
> anticipated .. which appears to be the case.
>
> Regards
>
> Alan Clark
>
>
> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Timur
> Friedman
> Sent: Wednesday, April 02, 2003 2:34 PM
> To: Magnus Westerlund; Jose Rey
> Cc: IETF AVT WG
> Subject: Re: [AVT] SDP signaling for RTCP XR format
>
>
> Magnus Westerlund wrote:
>
>> Here one issue actually surface. Is it the right thing to restrict a
>> receiver to not send DLRR report block unless the usage has been
>> explicitly signaled in multicast? The reason why I see a need to
>> restrict it is the potential to create enormous report blocks in large
>> multicast groups. And the only result is that the different receivers
>> will determine there RTT to other receivers which seems to be rather
>> pointless in most application. It seems that you would in fact like to
>> control this operation to a few different operation modes in 
>> multicast.
>> 1. fully meshed, both senders and receivers return DLRR blocks.
>> 2. Sender only, only senders return DLRR blocks.
>> 3. functionality off, no one responds with DLRR blocks.
>>
>> Should we define such a distinction in the signaling?
>
> I agree that the signalling should allow each of these modes.  But 
> also the
> mode in which certain receivers calculate RTTs amongst themselves.  
> While
> many applications might not use this mode, we do not want to 
> unnecessarily
> rule it out.
>
> Here are two examples of when the mode could be used.  First, receivers
> might measure RTTs amongst themselves as a prelude to some of them 
> becoming
> senders (multicast "repair servers", retansmitting lost packets, for
> instance). Second, a network monitoring tool might wish to measure its 
> RTT
> to certain receivers.
>
> - Timur Friedman
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
> <winmail.dat>

T.M. Eubanks
Multicast Technologies, Inc.
e-mail : tme@multicasttech.com
http://www.multicasttech.com

  Our New Multicast Workshop :
  http://www.multicasttech.com/workshop
Test your network for multicast :
http://www.multicasttech.com/mt/

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] SDP signaling for RTCP XR format 
In-Reply-To: Your message of "Fri, 11 Apr 2003 17:04:14 EDT."
             <005401c3006d$e8f68090$6801a8c0@telchemy.com> 
Date: Sat, 12 Apr 2003 15:46:13 -0400
From: Colin Perkins <csp@csperkins.org>
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Alan,

--> "Alan Clark" writes:
>At the recent IETF meeting we agreed that the existing draft was getting
>fairly stable - there were some good detail points brought up during the
>AVT discussion however these were easily accomodated.  
>
>We also agreed at the AVT meeting to include some simple SDP signaling at
>Magnus' request however felt at the time that this would be fairly
>straightforward and would not take much time.  The discussion on SDP
>appears to be getting much more involved than anticipated and there is no
>apparent end in sight.  
>
>I would like to propose that we move to a final draft without SDP in order
>that we can reach closure.  In San Francisco we discussed the idea of
>creating a separate document to describe the use of SDP for negotiating
>reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was
>suggested precisely because of some concern that the negotiation mechanism
>could get complex and that there were a wider variety of scenarios than
>anticipated .. which appears to be the case.

We should certainly consider this, as one way of meeting the ITU deadline.
Before we go down this path, though, I'd like to see some discussion of the
use cases for RTCP XR packets: in which scenarios is the signalling needed?
If typical scenarios require signalling, then we have no choice but to move
forward with the SDP extensions and their integration with SIP/RTSP. On the
other hand, if signalling is only required in unusual cases, it is much
easier to separate into a new draft.

Regards,
Colin
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Clark" <alan@telchemy.com>
To: "Colin Perkins" <csp@csperkins.org>
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        "'Stephen Casner'" <casner@acm.org>,
        "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        "'IETF AVT WG'" <avt@ietf.org>
Subject: RE: [AVT] SDP signaling for RTCP XR format 
Date: Sat, 12 Apr 2003 20:26:15 -0400
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Colin,

Excellent advice - I'll put some thoughts together this weekend.

Regards

Alan


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Saturday, April 12, 2003 3:46 PM
To: alan@telchemy.com
Cc: 'Timur Friedman'; 'Stephen Casner'; 'Magnus Westerlund'; 'IETF AVT
WG'
Subject: Re: [AVT] SDP signaling for RTCP XR format 


Alan,

--> "Alan Clark" writes:
>At the recent IETF meeting we agreed that the existing draft was getting
>fairly stable - there were some good detail points brought up during the
>AVT discussion however these were easily accomodated.  
>
>We also agreed at the AVT meeting to include some simple SDP signaling at
>Magnus' request however felt at the time that this would be fairly
>straightforward and would not take much time.  The discussion on SDP
>appears to be getting much more involved than anticipated and there is no
>apparent end in sight.  
>
>I would like to propose that we move to a final draft without SDP in order
>that we can reach closure.  In San Francisco we discussed the idea of
>creating a separate document to describe the use of SDP for negotiating
>reporting/ feedback for RTP XR, RTCP and other mechanisms.  This was
>suggested precisely because of some concern that the negotiation mechanism
>could get complex and that there were a wider variety of scenarios than
>anticipated .. which appears to be the case.

We should certainly consider this, as one way of meeting the ITU deadline.
Before we go down this path, though, I'd like to see some discussion of the
use cases for RTCP XR packets: in which scenarios is the signalling needed?
If typical scenarios require signalling, then we have no choice but to move
forward with the SDP extensions and their integration with SIP/RTSP. On the
other hand, if signalling is only required in unusual cases, it is much
easier to separate into a new draft.

Regards,
Colin

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Attached are the minutes for the first session (Wednesday afternoon)
of the AVT meeting at IETF 56 in San Francisco.  Colin is separately
writing the minutes for the second session.

These minutes need to be turned in by the end of the day tomorrow
(Monday), so any comments you can provide quickly would be
appreciated.

                                                        -- Steve

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--0-738478263-1050266884=:26225--
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Sun, 13 Apr 2003 15:13:51 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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To: avt@ietf.org
Subject: [AVT] Re: Draft minutes for first IETF56 AVT session
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> Stephen Casner <casner@acm.org> writes:
>
> Audio/Video Transport Working Group Minutes [...]
>
> RTP Payload Format for uncompressed video [...]
>
> The last question was whether there should be a means 
> to independently represent SMPTE timecodes [...]
>
> Stephan advised we need input on real usage cases in
> order to produce anything useful. 

I'd recommend this paper -- it was very useful in helping me
think through how MTC (MIDI Time Code, i.e. SMPTE 12M over
MIDI 1.0 DIN cables) sent over MWPP could end up being used
in multi-stream RTP sessions:

Terry K, Todd C. An open model for audio elementary stream encoding
in a DTV distribution chain. [Journal Paper] SMPTE Journal-Society of
Motion Picture & Television Engineers, vol.107, no.11, Nov. 1998,
pp.950-60. Publisher: Soc. Motion Picture & Telev. Eng, USA.

Abstract

This paper addresses issues of audio distribution for modern digital
television (DTV) systems focusing on digital broadcast systems. In
particular, a model is described for handling audio and video
elementary streams in existing distribution networks. The model
supports new system architectures that allow audio and video coding to
take place at many points in the distribution chain. It covers both
physical and logical interface formats, synchronization with
associated video signals, and carriage of metadata associated with the
audio. The model also demonstrates that many audio features of new
digital delivery formats can be exploited by existing broadcast
systems with minimal additional equipment investment. It makes use of
existing AES and SMPTE standards; however, new standards and
recommended practices to support the open model may be desired by the
industry. Specific recommendations are discussed. In addition,
examples of systems applying the model to DTV broadcast facilities are
described including examples of products that implement the audio
coding function using the model.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: Re: [AVT] Draft minutes for first IETF56 AVT session 
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--> Stephen Casner writes:
>
>Attached are the minutes for the first session (Wednesday afternoon)
>of the AVT meeting at IETF 56 in San Francisco.  Colin is separately
>writing the minutes for the second session.
>
>These minutes need to be turned in by the end of the day tomorrow
>(Monday), so any comments you can provide quickly would be
>appreciated.

Here is a revision, including the second session. 
Colin


--==_Exmh_4543271920
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Content-Description: minutes.txt
Content-Disposition: attachment; filename="minutes.txt"


Audio/Video Transport Working Group Minutes

Reported by Stephen Casner and Colin Perkins

   The AVT working group met in two sessions at the 56th IETF meeting in 
   San Francisco. In the first session, the group discussed the status of
   several documents in progress including the profile for Secure RTP, plus
   RTP framing over TCP, multiple RTCP extensions, and two video payload
   formats.  In the second session, the discussion covered seven payload
   formats related to audio (including MIDI and distributed speech
   recognition).

Introduction, Document Status, and Open Issues

   This meeting began with an update by Steve Casner on document publication
   status.  Two momentous steps were achieved in the days preceding the
   meeting:  The revised RTP specification and A/V profile (revisions of RFC
   1889 and 1890) were approved by the IESG for publication as Draft
   Standards, and the payload format for MPEG-4 was submitted by the working
   group to the IESG with a request for publication as a Proposed Standard.
   The RTP spec and A/V profile had been tentatively approved before the
   previous meeting, but there were several "RFC Editor note" clarifications
   requested by the IESG.  In addition, a number of questions regarding the
   text have been posted to the WG mailing list during the IESG review.
   Consequently, Steve Casner wrote clarifications to address these
   questions and WG participants discussed them on the mailing list after
   the previous meeting.  Ultimately, there were enough changes that the
   IESG agreed that the drafts could be revised before submission to the RFC
   editor: draft-ietf-avt-rtp-new-12 and draft-ietf-avt-profile-new-13.  The
   IESG gave final approval of these revised drafts and sent them to the RFC
   Editor.  Steve briefly reviewed the list of changes at this meeting and
   asked those who made comments in the mailing list discussion to verify
   that the results are satisfactory.

   We continued the pattern of publishing at least one RFC since the last
   meeting (RFC 3497 on the payload format for SMPTE-292M video), and four
   other drafts are in the RFC editor queue (the MIME registration for the
   payload formats in the RTP profile and the SDP bandwidth modifiers for
   RTCP bandwidth, both waiting for the RTP specification, and the payload
   formats for EVRC/SMV and ETSI ES 201 108 DSR).  Four drafts are with the
   IESG (ECRTP which is in Last Call, TCRTP, SRTP, and MPEG-4).

   Several drafts are either in WG last call or may be ready.  The draft on
   RTCP Feedback profile AVPF (draft-ietf-avt-rtcp-feedback-05) has been in
   last call for some time and is awaiting a final revision.  The companion
   draft which reports on simulations to validate the protocol design has
   been revised (draft-burmeister-avt-rtcp-feedback-sim-01) and will be
   published as Informational at the same time.  The RTP Retransmission
   draft, which uses the AVPF, has been revised based on WG comments and is
   ready for WG last call when the AVPF draft is updated.  Two drafts on
   Uneven Level Protection and Unequal Erasure Protection are being
   considered together as alternative methods; a new approach is proposed
   for ULP to revise RFC 2733 to fix its non-standard use of RTP header
   bits, but this was not completed before the meeting.  Lastly,
   draft-ietf-avt-rtcp-report-extns-03 on RTCP Extended Reports (XR) was
   discussed at this meeting to consider whether it is ready for last call.


Secure Real-time Transport Protocol

   Mark Baugher reported that SRTP (draft-ietf-avt-srtp-05) was submitted to
   the IESG in mid-2002 and IETF Last Call was issued, but the Security and
   Transport Area Directors expressed concerns regarding the interaction
   between the encryption ciphers and authentication.  Those concerns were
   discussed at the previous IETF AVT meeting and subsequently with the ADs.
   Mark summarized the agreed modifications: authentication using an 80-bit
   MAC is mandated for SRTCP and the default for SRTP, but user may choose
   null authentication for SRTP.  Precise language will be crafted for the
   draft about when this is acceptable and to make clear the risks.
   Discussion of error tolerance will be removed from the draft, even though
   it was one of the original design requirements, because it has become a
   point of controversy due to its dependence on changes at several other
   layers of the protocol stack.  Other considerations are sufficient to
   justify the design decisions.  Automatic key management is mandated to
   ensure that the key streams are not repeated in order to avoid a serious
   failure case with the default counter mode cipher.  The authors also
   requested permission to make some small changes/clarifications reflecting
   recent feedback from implementers; these will be sent in a separate note
   to the list.  Mark stated the authors' intention to produce a new draft
   in a couple of weeks.  Steve Casner reported that our AD Allison Mankin
   had mentioned in a side conversation at this meeting that she wants this
   draft to go ahead quickly.

    
Framing RTP over Connection-Oriented Transport

   John Lazzaro presented draft-lazzaro-avt-rtp-framing-contrans-00, a new
   draft that restates the description of RTP framing over TCP that was
   removed when the RTP/AVP Profile (RFC 1890) was revised since no report
   of interoperability of the feature was obtained.  It is now claimed that
   there are implementations, so it would be appropriate to have it
   documented.  As this draft was prepared, two related problems became
   apparent:  1) the MMUSIC comedia draft specifies SDP session descriptions
   using TCP transport, but does not specify a format parameter to indicate
   RTP/AVP inside the TCP; and 2) RTSP (RFC 2326) specifies a single TCP
   connection carrying interleaved control and data, but not separate data
   streams carried in TCP.  This draft includes sections to address these
   last two topics, but it is an open issue for the WG whether these should
   be kept or abandoned.

   As an alternative to this "classical" framing of RTP in TCP using a
   16-bit frame length field between frames, some people have suggested on
   the mailing list that the multiplexed framing specified in RTSP be used
   instead.  John asked for input from the group whether one or the other
   (or both or neither) approach should be followed.  Steve Casner noted
   that the TCP framing was removed in the profile revision due to lack of
   interop, and asked whether there really is sufficient interest now to
   warrant publishing a spec.  John replied that this is needed in MIDI over
   RTP (see MWPP, later) because much of the user community is uncomfortable
   with UDP.  Ross Finlayson spoke in favor of using the RTSP framing which
   is implemented in several RTSP servers including Apple's and his own
   (Live.com).  Henning Schulzrinne countered that it is only really
   applicable if there's a control protocol in the same stream, and
   suggested that we just specify the "classical" framing now and leave for
   the future a specification of how RTSP framing can be used separately
   from RTSP (requiring a different format parameter in SDP to indicate that
   framing).  One reason is that RTSP is undergoing update now.  Ross
   suggested that specifying a default "channel ID" would suffice for using
   RTSP framing without a control protocol, but others were concerned that
   there would be a desire to use multiplexing with other control protocols
   where the syntax of the RTSP framing would conflict, so it is not
   general.  Colin Perkins summarized the discussion by saying that this
   draft should restrict itself to the classical framing and leave the RTSP
   framing to be specified somewhere else if there is interest or dropped.
   The SDP format parameter "TCP RTP/AVP" would indicate this simple case;
   other parameters could be defined for multiplexed framing as needed.

   Regarding the other problem of specifying how data can be carried in
   separate TCP streams under RTSP, John concluded and Magnus Westerlund
   agreed that this should be left to the revised RTP specification.

RTCP Extended Reports (XR)

   Timur Friedman reviewed the changes to the RTCP reporting extensions in
   draft-ietf-avt-rtcp-report-extns-02 with minor additional updates in -03.
   The RTCP packet type number is changed to 207 to avoid the conflict with
   the RTCP Feedback draft, and the numbering and structure of a few of the
   report blocks have been simplified.  There are several open issues
   regarding the -03 revision, the first of which is that the title was
   changed to avoid the need to spell out the RTCP acronym.  Steve Casner
   feels that the title should remain RTCP because that is the accurate
   topic.  The title would be "RTP Control Protocol (RTCP) Extended
   Reports", which is not too long.  Section 4.3 specifies how packet
   arrival timestamps in RTP timestamp format may be reported back; Magnus
   Westerlund has suggested that these timestamps should be converted to
   some fixed units to avoid problems if the RTP timestamp clock rate
   changes.  Steve Casner said that is not usually done because it causes a
   number of other complications, and that you should be able to use the
   sequence numbers that are returned with the timestamps to select the
   correct rate.  However, Steve asked for clarification in the text that
   these are arrival timestamps, not the sender's timestamps.

   In Section 4.4, the definition of standard deviation and TTL will be
   clarified.  The variable geometry of this packet type, indicated by a bit
   field, was simplified to a single format.  Some people would prefer the
   space savings of the variable format, while others prefer the simplicity.
   Steve suggested that if particular subsets are most useful, then those
   subsets should be defined as distinct block types.

   Alan Clark discussed the changes to the VoIP metrics in Section 4.7.  In
   -02, echo level measurements were added since a number of implementations
   produce that data.  In -03, to support for sample-based codecs as well as
   frame-based codecs, the jitter buffer metric was changed to units of time
   (5ms) rather than frames.  A couple of people raised the question whether
   the 5ms unit should be 1ms instead.  Alan responded that 5ms seemed a
   better tradeoff between resolution and range for the performance effects
   to be measured.  It could be done, but would require increasing three
   fields from 8 to 16 bits.  Anwar ??? asked about the requirement that all
   fields of the VoIP report block be supported.  Alan responded that
   although the block structure is fixed, some of the fields allow an
   "undefined" value for use when the metric does not apply.  Steve Casner
   asked for the use of the undefined value to be clarified.  Anwar also
   asked if there would be a XR block type for one-way delay.  Timur
   responded that this could be added in the future if someone develops a
   method that proves useful.  Philippe Gentric wants to get reports of
   instantaneous fill level rather than average; Alan said the collection of
   measures should allow this to be determined.

   Steve Casner raised a few issues regarding the completeness of the draft
   and the merging of the VoIP metrics with the other sections.  The VoIP
   metrics include a measure of round trip time, but as specified, this is
   only possible if data is being transmitted in both directions.  The draft
   needs to explain that, or to explain what to do if that's not the case.
   The VoIP section could refer to the the non-sender RTT measurement method
   in Section 4.5, but it doesn't.  The draft needs an applicability
   statement at the top to explain which of the tools in this toolbox should
   be used in which situations, and how they would be used.  The draft
   should be careful to avoid unstated assumptions about the use cases.
   Another problem is that Section 4.7 gives rules for how often the VoIP
   metric blocks must be sent.  This draft is not allowed to say that.  The
   RTP spec defines the basic RTCP packet timing, and the RTCP Feedback
   profile defines alternate timing under a more restrictive use case.  This
   draft could refer to use under the RTCP Feedback profile, but it can't
   specify timing of its own.  Alan responded that this would be done, and
   commented that in some cases the more useful information in the VoIP
   report block would allow less frequent reporting, saving bandwidth.
   Steve agreed that it would be good to add a sentence pointing this out
   because the increased in RTCP packet size caused by the inclusion of the
   VoIP metrics will mean that the interval between RTCP packets is
   increased.

   A more complicated issue raised by Magnus Westerlund is that the draft
   should specify SDP signaling for its use.  This is a valid request, but
   might introduce too much delay in completing the draft.  Colin Perkins
   pointed out the need to update the document quickly in order to meet a
   May deadline for consideration by ITU H.460.9.  The specification of SDP
   signaling could be a separate draft, but it appears feasible to include a
   specification of SDP for point-to-point applications and leave more
   complicated scenarios for a separate draft.


RTCP Extensions for SSM with unicast feedback

   Julian Chesterfield discussed updates in draft-ietf-avt-rtcpssm-03 which
   introduces some additional summary statistics, some clarifications and
   examples of security requirements and how the security methods should be
   used for RTSP and SIP sessions, and updates to the IANA considerations.
   The new summary sub-blocks are IPv6 feedback address, BYE list, and RTCP
   receiver bandwidth.  This revision is now aligned with the RTCP XR format
   that Timur just presented, as was suggested in a previous AVT meeting.
   Steve Casner clarified that the reason for the suggestion was not to save
   an RTCP packet type code point, rather to ask if the same reporting
   mechanisms could be shared for both purposes rather than defining
   different ones.  If there is no feasible merging of functionality, then
   there is not a requirement to fit the SSM reports into the XR block
   structure with extra overhead.  The authors will reconsider this issue.

   Some open issues remain for the draft: clarification that the sender must
   forward a group size report whenever the size changes to allow correct
   operation of RTP timer reconsideration rules; reporting in the summary
   what the data corresponds to in terms of sample group size and receiver
   report age; and the alignment with XR.  The authors would like to know
   whether there are any other implementations of this draft, and whether it
   will be considered ready for WG last call when these open issues have
   been addressed.

   Timur Friedman suggests using XR block type 8 rather than 10.  John
   Lazzaro asked about summarization of the "extended highest sequence
   number" field from RTCP reports.  MWPP uses this value; he would want the
   summary to be the minimum value reported by receivers.  He will think
   about the details and make a suggestion.  Steve Casner asked how the
   summarization methods in this draft compare with the summarization
   methods in the XR draft.  Eve Schooler responded that the nature of the
   summary methods in the two drafts is different.  The XR draft reports
   sampling of data, while SSM reports mathematical distributions.  The
   authors of both drafts agreed to meet after the session to discuss this
   in more detail.  Steve commented that the reason for raising the point
   was that if there is no common applicability of the two methods, then
   binding them together does not make sense.

    
RTP Payload Format for JVT Video

   Stephan Wenger discussed draft-ietf-avt-rtp-h264-01.  The spec for the
   JVT codec itself was finalized in a meeting the previous week with
   acceptance for ISO FDIS status and with "Consent" in the ITU-T due on
   March 28, so it is now time to finalize the packetization here and
   Stephan hopes to accelerate the process.  A -02 revision is to be
   produced shortly, and perhaps one more revision if there are comments.
   The hope is to go to WG last call before the next meeting.  There were
   many editorial changes from -00, plus a few technical additions.
   Fragmentation was added, but needs more description and an example.  A
   "decoding order number" was added to facilitate some special handling of
   the packets without needing to decode the bitstream, but the description
   in this revision is inadequate.  A section was added on MIME registration
   and SDP usage, but this may also need refinement.

   The main issue for this payload format is interoperability with the
   MPEG-4 Simple payload format.  A new informational draft will be written
   to describe how this interoperation may be done.  Colin Perkins asked
   whether interoperation with RFC 3016 is possible and there is any
   pressure for this.  Stephan replied that he thought not, but will
   investigate for the interop draft.  As the h264-01 draft now stands, it
   is possible to transmit a subset of JVT in MPEG-4 Simple access unit
   fragments, although the utility of this is not clear.  It was also agreed
   at the JVT meeting to add a new form of STAP to allow more useful
   interoperation.  But it has also been suggested that this payload format
   should re-model the interleaving of MPEG-4 Simple and that the STAP/MTAP
   syntax should be aligned with MPEG-4 Simple.  This would achieve
   syntactic alignment allowing some re-use of code, but there would be a
   huge semantic difference.  Dave Singer expressed the position that this
   draft should be made the best it can be for H.264 first, and then
   consider interop.  The only solid requirement for interop is that it must
   be possible for a video stream using this payload format to be part of an
   MPEG-4 presentation.  That is already possible.  It was agreed not to
   introduce mythical similarities.

   At the previous AVT meeting, Stephan was resisting the addition of
   media-unaware fragmentation, but has added it in this revision because it
   may be necessary to transport a NALU of size greater than 64KB which is
   the most that IPv4 fragmentation can handle.  In addition, in there may
   be content pre-recorded with a NALU size that does not fit the MTU size
   of a delivery network.  Doing the fragmentation at the application layer
   allows application of tools such as RFC 2733 FEC for better protection
   efficiency.

   The security section of the document needs to be expanded beyond the
   minimal text in -01.  The main issue is the vulnerability of Parameter
   Sets when transmitted in-band.  This is addressed in the codec spec, and
   Colin Perkins said a normative reference could be made to that.  Steve
   Casner asked why the in-band transmission of Parameter Sets should be
   allowed at all, given that the draft says it is a bad idea.  Stephan
   replied that it is necessary in some scenarios involving gateways.
   Philippe Gentric expressed serious concern that if this practice is
   allowed then people will use it when it shouldn't be used, and it will
   cause endless headaches similar to those already seen with RFC 3016.
   Stephan said the draft will say SHOULD NOT to discourage its use.  The
   codec spec also includes the means to transmit data of unknown type that
   may or may not be active information, for example a set-top box software
   update that could contain a virus.  Stephan asks for help on security
   language to address this problem.

   Another open issue is video conferencing support.  Stephan would like to
   specify MIME codepoints for levels of operation beyond those included in
   the codec spec, e.g. 704x576 images at 7.5 fps.  Colin asked whether
   these modes will be added in future revisions of the codec spec, which
   would result in two ways to specify the same mode.  Stephan said no.
   There is also a question whether this payload format is the right place
   to specify whether the response to a request for a full intraframe is
   mandatory.  It was agreed that this payload format cannot specify that
   because it is subject to congestion control.  Colin said this topic will
   be discussed separately in come combination of AVT and MMUSIC.

   The last open issue is regarding timestamps for Field mode, where there
   may be separate sampling instants for the two fields of a picture.
   Discussion requires detailed knowledge of H.264, so was deferred.

    
RTP Payload Format for uncompressed video

   Ladan Gharai reviewed the modifications, additions, and remaining open
   issues for draft-ietf-avt-uncomp-video-02.  The payload header now
   extends the sequence number in the RTP header to 32 bits to accommodate
   high data rates.  Additional color codings for 4:1:1 and 4:2:0 interlaced
   and progressive were added, and separate timestamps are specified on
   interlaced fields to accommodate reversing 3:2 pulldown (an issue that
   was raised at the previous meeting).  The draft now specifies required
   and optional SDP parameters (rate, pgroup, color-mode, etc.) and covers
   congestion control in Security Considerations (a serious issue for very
   high data rates).

   There are some open issues.  At the previous IETF it was suggested that a
   planar video mode be added to the draft, i.e., sending Y, Cb and Cr
   planes separately.  This requires a 2-bit field, which could be stolen
   from the length field, to indicate which plane.  The main concern is that
   a substantial portion of the draft related to pgroups and color
   subsampling would be irrelevant to this mode.  Does it make sense to
   combine these disparate modes in one draft?  Stephan Wenger recommended
   keeping them separate.  Two bits may not be enough since there are some
   color models with more than 4 planes, and the plane numbering would need
   to be specified for all these models.  He also felt this would not be
   used because display systems are built to receive data in that form.
   Philippe Gentric countered that systems doing processing, rather than
   display, do operate on the planes separately, and since they operate at
   gigabit rates, saving memory shuffling is important.  However, after some
   study he concluded that the implementations differ so much that a single
   format won't help.  It was agreed to drop planar mode.

   On the other hand, the group felt it was worthwhile to keep the detailed
   listing of pgroup values rather than just the text explaining the rules
   for determining the values.  Ladan asked if anyone has additional color
   subsampling or colorimetry values that should be included in the draft.
   Stephan Wenger said he had several that he would discuss after the
   meeting.  Steve Casner said the draft needs to specify the means for
   defining and registering new values after the draft is published.

   The last question was whether there should be a means to independently
   represent SMPTE timecodes, or are the RTP and RTCP SP timestamps
   sufficient.  Dave Singer said the SMPTE timecodes are required for some
   video applications because they can have gaps in edited programs, but it
   is orthogonal to the data format of the video.  Colin Perkins asked
   whether the NPT mapping in RTSP solves this problem or whether some means
   is needed in the payload format.  Philippe Gentric said this is really
   needed for studio work, and recommended that there be a separate payload
   format defined for carrying just the SMPTE timecodes that could then be
   synchronized with RTP mechanisms to any payload format.  Colin recalled a
   proposal for that a couple of years ago, and suggested that we resurrect
   it.  Stephan advised we need input on real usage cases in order to
   produce anything useful.  John Lazzaro said that MIDI timecodes are SMPTE
   timecodes encoded in MIDI, and MWPP provides a means to transport these
   with resiliency.  This would be a solution, perhaps a gross one.


RTP Payload Format for a 64 kbit/s transparent call

   The second session started with a discussion of the RTP Payload format
   for 64 kbps transparent calls (draft-kreuter-avt-rtp-clearmode-02.txt)
   by Ruediger Kreuter. This draft describes a single 64kbps clearmode
   connection, and MIME type registration. It is distinct from Nx64kbps
   transport (draft-vainshtein-cesopsn-XX.txt) work which is done in the
   PWE3 working group. 

   The MIME type "audio/clearmode" is registered. Ruediger asked if this is
   appropriate? Flemming Andreason said yes, using "audio" as the MIME type
   makes things easier in SDP since it can be included in the same media
   stream as other codecs. Ruediger also noted that the ITU use "audio" for
   this, as do existing products, so that helps compatibility. Ruediger
   noted that Magnus Westerlund sent a number of comments on the draft. Due
   to these he will include the "maxptime" parameter, and an explanation of
   the mapping from MIME parameters to SDP parameters. 

   The chairs asked if there were objections to making this a work item?
   None were raised, so the next version will be submitted as an AVT work
   item, subject to area director approval.  Colin Perkins noted that it 
   is necessary to avoid overlap with work in PWE3 (i.e. limit this to a
   single 64kbps channel) if we are to accept it as a work item.


The MIDI Wire Protocol Packetization (MWPP)

   John Lazzaro discussed MWPP (draft-ietf-avt-mwpp-midi-rtp-06.txt) and 
   coding guidelines (draft-lazzaro-avt-mwpp-coding-guidelines-02.txt). 
   He noted that he has been in discussion with the MMA and AMEI (the MIDI
   manufacturers associations for the US and Japan), and is incorporating
   their feedback as appropriate. He also noted that IEEE P1639 has related
   work, solving the problem at layer 2, and outlined differences between 
   the two approaches. 

   John noted that equipment today uses MIDI cables, asynchronous serial
   cables, which do not use timestamps. Instead, the timing of the signal
   on the wire denoted the media timing. People want pseudo-wire emulation
   in RTP, so they can use the precise packet times when operating on a LAN
   or other low-jitter environment.  Wants a parameter to indicate that the
   sender is "precision", so the last command in the packet indicates the
   sending time. 

   The other issue is synchronization between direct digital audio output
   and synth MIDI output. Options are to use an external clock sync, and
   slave everything to it, or to specify packet buffer latency and use
   manual calibration.

   There was much discussion around this subject between John, Magnus
   Westerlund, Steve Casner and Colin Perkins. This leant towards the first
   option: somehow convey the mapping between RTP timestamp and an external
   sync clock, and use the external clock to determine when to playout the
   media. Steve Casner gave a reference to the BBN synchronization protocol 
   (J. Escobar, C. Partridge and D. Deutsch, Flow Synchronization Protocol,
   IEEE/ACM Transactions on Networking, Volume 2, Number 2, April 1994),
   which is an example of this approach. 
   
   Colin Perkins noted that RTSP has a similar signalling mechanism, to
   convey a mapping between RTP timestamps and a SMPTE timecode. John noted
   that MWPP may also be used with SIP, so a more general parameter may be
   useful, but is nervous about splitting this out into a separate document
   due to feature creep. It may be appropriate for the MWPP document to
   outline the form of the solution, but to leave the actual solution to a
   different document (since the solution to synchronizing playout from
   multiple sources is more general than MWPP).


RTP Payload Format for ATRAC-X

   Matthew Romaine discussed draft-hatanaka-avt-rtp-atracx-01.txt, the RTP
   payload format for ATRAC-X. A number of questions were raised at the last 
   meeting: ambiguity in the timestamp definition, concern that the reasoning
   behind the multiplexing was not convincing, the reason why RFC 2198 was
   not used for redundancy, and concern about possible decoding ambiguity
   when fragmentation is used. Since then, the timestamp and sample rate
   have been clarified, the redundant data framework has been modified, a
   new method of multi-channel data decomposition has been introduced, and
   the reasons for multiplexing have been solidified.  

   The timestamp now corresponds to the presentation time in milliseconds
   and the sample rate must now be identical for all streams. Steve Casner
   asked why milliseconds are used for the presentation time rather than
   the sample clock rate? Will revisit.

   The RFC 2198 redundancy format has a lot of overhead, because of the
   payload headers, and its block length field (10 bits) is too short, so
   redundancy is included in the payload format.

   Multi-channel decomposition is now included, allowing rate adaptation at
   the sender, which can drop the less important channels based on feedback
   from the receiver, and allowing for multi-channel presentations to be
   split across multiple ATRAC-X streams if they use more channels than are
   supported by the codec. 

   There was some confusion over the terminology, with Steve Casner and
   Colin Perkins asking questions for clarification whether the multiple
   streams are from one source, or multiple sources. The draft will be
   updated to clarify the distinction between ATRAC-X streams and RTP
   streams.

   Steve Casner expressed concern that the payload format is inflexible in
   the way streams are multiplexed: for example layered coding is typically
   implemented by layered data across several RTP sessions, but the payload 
   format sends the multiple layers within a single RTP stream. It would be 
   appropriate to consider scenarios other than unicast streaming to ensure 
   the design is does not unduely constrain future applications.

RTP Payload Format for iLBC Speech

   Alan Duric discussed iLBC speech (draft-ietf-avt-rtp-ilbc-01.txt and
   draft-ietf-avt-ilbc-codec-01.txt). Since the last meeting, the codec
   and payload format have been tested at the SIPit 12 interop event,
   with several interoperable implementations; a 20ms frame size mode
   has been introduced into the codec (with 4 blocks of 160 samples); 
   and the PLC has been enhanced.

   The SDP now includes a "mode=" attribute to indicate if 20ms or 30ms
   frame sizes is preferred (the default, if the attribute is not present,
   is 30ms for backwards compatibility). Steve Casner asked if the frame
   size can be distinguished in-band? Not without breaking the backwards
   compatibility. Steve asked how the zero value - support both - is used?
   Not clear, maybe just list it as "reserved". Colin Perkins asked if the
   "a=ptime" attribute can be used instead?  No due to ambiguity with 60ms
   packets (2x30ms or 3x20ms).

   The current open issue is VAD and comfort noise generation. No interest
   at present, so they plan to drop this. Steve Casner noted that comfort
   noise frames would need to be indicated in-band? Yes, this needs to be
   considered how they can be distinguished. May be possible to use the
   existing comfort noise payload format. 

   See also http://www.iLBCfreeware.org/ for source code and more info.

RTP Payload Format for the Speex Codec

   Greg Herlein was scheduled to discuss the Speex codec and payload format
   (draft-herlein-speex-rtp-profile-00.txt), but couldn't reach the meeting 
   due to traffic disruption caused by anti-war protests in the city. Steve 
   Casner briefly outlined the codec and payload format, and asked for any 
   comments to be sent to the mailing list.

RTP Payload Format for RGL Codec

   Michael Ramalho discussed the RGL codec (draft-ramalho-rgl-desc-01.txt)
   and payload format (draft-ramalho-rgl-rtpformat-01.txt). He started with
   a brief introduction to the codec, which provides lossless compression 
   for G.711. The output of the RGL codec is variable rate. Arbitrary input 
   can be compressed, and the output will be at most one octet larger than
   the input (and usually will be much smaller).

   The first octet of the compressed output will never match one of a set
   of reserved values (0x3e, 0x5e, 0x7e, 0x9e, 0xbe, 0xde and 0xfe). These
   reserved values are used in the payload format to indicate a particular
   bundled format: bundling of one, two or three RGL frames per packet is
   signalled by one of these reserved values, and a less efficient, but
   generic, format is signalled by another value, and allows an arbitrary
   number of frames per packet. 

   Steve Casner asked if the codec is likely to switch between single- and
   multi-frame modes? It may be possible to distinguish the bundling at
   session setup time, to avoid the need for reserved byte codes (as is
   done by the EVRC payload format). Magnus Westerlund also noted that one
   can always distinguish the format using different RTP payload types, so
   it may not be necessary to use the reserved codes.

   Magnus Westerlund asked if the generic format, or something similar with
   a count of the number of frames in the packet, can be used in all cases?
   This is possible, but the overhead may be too large. Magnus noted that,
   if the type octet was removed, the size would be the same in almost all
   cases, and the format would be simpler. 

   Steve Casner noted that this is rather complex, with several different
   formats.  That is a detraction, that could be fixed using two payload 
   types (one frame per packet; multiple frames per packet).

   Michael noted that it is desirable that the number of samples in the
   payload headers matches the "a=ptime" attribure in the SDP. Also, he
   noted that there is a missing parameter for a-law or u-law.

   More information is available at http://www.vovida.org/
    
RTP Payload Format for ETSI ES 202 050 DSR

   Qiaobing Xie discussed draft-xie-avt-dsr-es202050-00.txt, which is a new
   encoding scheme for distributed speech recognition. The draft uses exactly 
   the same format as that used in the previous DSR payload format RFC-to-be,
   although the contents of the frame pairs, and the MIME type, are different.
   Accept as a work item? Yes, subject to area director approval.

   Steve Casner asked if this draft cannot be simplified more, being just a
   reference to the previous DSR payload format and registration of the new
   MIME type? Qiaobing wished to clarify a couple of points that were unclear 
   in the previous draft, and so copied the text and added a diagram for 
   explanation. Steve said it would be appropriate to insert the change
   into the other draft, as an RFC editor note, to ensure that both are
   clear.

File format for EVRC/SMV vs QCP

   Steve Casner noted that there is a proposal to change the file format
   for the EVRC/SMV payload format (currently with the RFC Editor) to be
   that defined in draft-garudadri-qcp-00.txt. Steve asked for an "ego-free" 
   evaluation of which format makes most sense, or if we should accept both
   formats?

   It was asked if the QCP file format has IPR associated with it, and if
   the implementations are independent. Randall Gellens replied, noting
   that he is still researching these issues and cannot give a definitive
   answer, but that he thinks the QCP implementations are independent and
   is not aware of any IPR on QCP. 

   Randall also noted that he was unaware of the QCP format when working on
   the EVRC/SMV payload format.  The QCP format has been in use for several
   years, by other applications, but the MIME type has not been registered.
   It may be necessary to register this, for use with QCELP for example, 
   even if it is not used for the EVRC/SMV payload format. There may be
   implementations using this with EVRC/SMV though.

   It is unclear if the EVRC/SMV format is used as currently specified, 
   but 3GPP-2 may be considering this. Steve Casner asked if we can get
   clarification from 3GPP-2 on their plans for the file format? We may
   be able to get a liaison statement from 3GPP-2, stating their needs?

   No decision was made, but it is hoped that input from 3GPP can help
   to resolve the issues.

Other discussion

   Since there was time left at the end of the session - due to missing the 
   Speex discussion - Steve Casner asked if there were any other topics that 
   the group wished to discuss?

   It was asked if it is possible to add new items to the RTCP reporting
   extensions draft (e.g. for video quality assessment)? Not at this time,
   because that draft is complete and we don't wish to delay it. Extensions
   may be written as new drafts though.

   John Lazzaro asked when the AVT group might close, since the charter is
   open ended due to the addition of new payload formats? Steve noted that
   we have fairly broad license to work on payload formats, but need area
   director approval for other new work items. Colin Perkins noted that the
   current charter lists getting RTP to Draft Standard as the main goal of
   the group. This is now complete, so we should be considering future work
   items.

   Stephan Wenger asked if it is appropriate to consider moving existing
   payload formats to draft standard? Yes, or to historic if they are no
   longer useful. There was some discussion of how the H.263 format can
   move to draft standard. 

   Magnus Westerlund noted that NAT boxes present a challenge. Should the
   AVT group work to define "symmetric RTP"? Steve replied that it should
   probably be defined in the control protcols that use RTP, since they may
   have differing interpretations of the term. Magnus noted that there is 
   a lot of commonality between the protocols, so a separate draft may be 
   warrented, and there is the issue of address binding? Something to be
   discussed offline.

   Colin Perkins noted that the group may also wish to consider mapping RTP
   onto DCCP as a future work item. Transport over PR-SCTP was suggested as
   another possible work item.  These are long term goals, not something we
   should rush into, but something to consider as those other protocols are
   developed.

				  - * - 


--==_Exmh_4543271920--


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,

Back in the office, I have continued my work on updating the XR format 
text.

Colin Perkins wrote:

>We should certainly consider this, as one way of meeting the ITU deadline.
>Before we go down this path, though, I'd like to see some discussion of the
>use cases for RTCP XR packets: in which scenarios is the signalling needed?
>If typical scenarios require signalling, then we have no choice but to move
>forward with the SDP extensions and their integration with SIP/RTSP. On the
>other hand, if signalling is only required in unusual cases, it is much
>easier to separate into a new draft.
>
>  
>

I have one use case for the SDP signaling. This is the 3GPP streaming 
 service (PSS). There exist different releases which include different 
functionality and some of that functionality is optional. In Release 6 
of PSS there is a proposal for bit-rate adaptation based on RTCP 
feedback and server decision. It is proposed that as a part of the 
adaptation functionality some of the XR blocks should be used. In this 
case it is very useful for a server to tell the client that it desires 
the client to send XR blocks and which. In that way the client avoids 
having to try to guess or infer that it shall send the XR blocks or not.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Mon Apr 14 11:03:05 2003
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From: "Alan Clark" <alan@telchemy.com>
To: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        "'Colin Perkins'" <csp@csperkins.org>
Cc: "'IETF AVT WG'" <avt@ietf.org>
Subject: RE: [AVT] SDP signaling for RTCP XR format
Date: Mon, 14 Apr 2003 10:53:18 -0400
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My concern is that, whilst there may be some examples of scenarios which
would benefit from SDP signalling, is the identification of all such
scenarios and the discussion of their issues and ramifications for RTCP XR
going to be an unbounded and indefinite discussion?

Originally we agreed to put a simple SDP signalling definition in by way of
example - this seems to have been extended into a more general discussion.

Can we establish a firm goal for either completing the SDP text or deciding
that the number of scenarios warrants a separate discussion/draft?

Alan


-----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Magnus
Westerlund
Sent: Monday, April 14, 2003 4:46 AM
To: Colin Perkins
Cc: 'IETF AVT WG'
Subject: Re: [AVT] SDP signaling for RTCP XR format


Hi,

Back in the office, I have continued my work on updating the XR format
text.

Colin Perkins wrote:

>We should certainly consider this, as one way of meeting the ITU deadline.
>Before we go down this path, though, I'd like to see some discussion of the
>use cases for RTCP XR packets: in which scenarios is the signalling needed?
>If typical scenarios require signalling, then we have no choice but to move
>forward with the SDP extensions and their integration with SIP/RTSP. On the
>other hand, if signalling is only required in unusual cases, it is much
>easier to separate into a new draft.
>
>
>

I have one use case for the SDP signaling. This is the 3GPP streaming
 service (PSS). There exist different releases which include different
functionality and some of that functionality is optional. In Release 6
of PSS there is a proposal for bit-rate adaptation based on RTCP
feedback and server decision. It is proposed that as a part of the
adaptation functionality some of the XR blocks should be used. In this
case it is very useful for a server to tell the client that it desires
the client to send XR blocks and which. In that way the client avoids
having to try to guess or infer that it shall send the XR blocks or not.

Best Regards

Magnus Westerlund

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Mon Apr 14 11:35:09 2003
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cc: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        "'IETF AVT WG'" <avt@ietf.org>
Subject: Re: [AVT] SDP signaling for RTCP XR format 
In-Reply-To: Your message of "Mon, 14 Apr 2003 10:53:18 EDT."
             <007b01c30295$964e1c70$6801a8c0@telchemy.com> 
Date: Mon, 14 Apr 2003 11:35:25 -0400
From: Colin Perkins <csp@csperkins.org>
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Alan,

I'm not so concerned about the number of scenarios, but rather those that
we expect to be commonly used. How do you expect RTCP XR to be used? 

Colin



--> "Alan Clark" writes:
>
>My concern is that, whilst there may be some examples of scenarios which
>would benefit from SDP signalling, is the identification of all such
>scenarios and the discussion of their issues and ramifications for RTCP XR
>going to be an unbounded and indefinite discussion?
>
>Originally we agreed to put a simple SDP signalling definition in by way of
>example - this seems to have been extended into a more general discussion.
>
>Can we establish a firm goal for either completing the SDP text or deciding
>that the number of scenarios warrants a separate discussion/draft?
>
>Alan
>
>
>-----Original Message-----
>From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Magnus
>Westerlund
>Sent: Monday, April 14, 2003 4:46 AM
>To: Colin Perkins
>Cc: 'IETF AVT WG'
>Subject: Re: [AVT] SDP signaling for RTCP XR format
>
>
>Hi,
>
>Back in the office, I have continued my work on updating the XR format
>text.
>
>Colin Perkins wrote:
>
>>We should certainly consider this, as one way of meeting the ITU deadline.
>>Before we go down this path, though, I'd like to see some discussion of the
>>use cases for RTCP XR packets: in which scenarios is the signalling needed?
>>If typical scenarios require signalling, then we have no choice but to move
>>forward with the SDP extensions and their integration with SIP/RTSP. On the
>>other hand, if signalling is only required in unusual cases, it is much
>>easier to separate into a new draft.
>>
>>
>>
>
>I have one use case for the SDP signaling. This is the 3GPP streaming
> service (PSS). There exist different releases which include different
>functionality and some of that functionality is optional. In Release 6
>of PSS there is a proposal for bit-rate adaptation based on RTCP
>feedback and server decision. It is proposed that as a part of the
>adaptation functionality some of the XR blocks should be used. In this
>case it is very useful for a server to tell the client that it desires
>the client to send XR blocks and which. In that way the client avoids
>having to try to guess or infer that it shall send the XR blocks or not.
>
>Best Regards
>
>Magnus Westerlund
>
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
>Torshamsgatan 23           | Fax   +46 8 7575550
>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Mon Apr 14 12:03:00 2003
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From: "Alan Clark" <alan@telchemy.com>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: "'Magnus Westerlund'" <magnus.westerlund@era.ericsson.se>,
        "'IETF AVT WG'" <avt@ietf.org>
Subject: RE: [AVT] SDP signaling for RTCP XR format 
Date: Mon, 14 Apr 2003 11:52:14 -0400
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Colin

I expect that one of the major uses of RTCP XR would be to support the fault
and performance management of IP phones, addressing both enterprise and
carrier applications, and potentially residential gateways that support
VoIP.   There is a strong need within the industry for a low cost means of
monitoring user perceived quality - this can be very economically
implemented using RTCP XR.  There are specific cases, for example IP
Centrex, where the absence of an RTCP XR-like protocol is severely impacting
the ability of service providers to detect problems that affect user
perceived quality and hence is affecting the speed with which these services
are deployed.

I certainly don't regard RTCP XR as a research or experimental tool, but as
a protocol that has some very real and immediate applications that will help
to speed deployment of VoIP.  I am also the editor for several emerging TIA
standards related to VoIP CPE management and have already introduced the
RTCP XR draft to the group as forming a good basis for our work.

Regards

Alan Clark


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Monday, April 14, 2003 11:35 AM
To: alan@telchemy.com
Cc: 'Magnus Westerlund'; 'IETF AVT WG'
Subject: Re: [AVT] SDP signaling for RTCP XR format


Alan,

I'm not so concerned about the number of scenarios, but rather those that
we expect to be commonly used. How do you expect RTCP XR to be used?

Colin



--> "Alan Clark" writes:
>
>My concern is that, whilst there may be some examples of scenarios which
>would benefit from SDP signalling, is the identification of all such
>scenarios and the discussion of their issues and ramifications for RTCP XR
>going to be an unbounded and indefinite discussion?
>
>Originally we agreed to put a simple SDP signalling definition in by way of
>example - this seems to have been extended into a more general discussion.
>
>Can we establish a firm goal for either completing the SDP text or deciding
>that the number of scenarios warrants a separate discussion/draft?
>
>Alan
>
>
>-----Original Message-----
>From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Magnus
>Westerlund
>Sent: Monday, April 14, 2003 4:46 AM
>To: Colin Perkins
>Cc: 'IETF AVT WG'
>Subject: Re: [AVT] SDP signaling for RTCP XR format
>
>
>Hi,
>
>Back in the office, I have continued my work on updating the XR format
>text.
>
>Colin Perkins wrote:
>
>>We should certainly consider this, as one way of meeting the ITU deadline.
>>Before we go down this path, though, I'd like to see some discussion of
the
>>use cases for RTCP XR packets: in which scenarios is the signalling
needed?
>>If typical scenarios require signalling, then we have no choice but to
move
>>forward with the SDP extensions and their integration with SIP/RTSP. On
the
>>other hand, if signalling is only required in unusual cases, it is much
>>easier to separate into a new draft.
>>
>>
>>
>
>I have one use case for the SDP signaling. This is the 3GPP streaming
> service (PSS). There exist different releases which include different
>functionality and some of that functionality is optional. In Release 6
>of PSS there is a proposal for bit-rate adaptation based on RTCP
>feedback and server decision. It is proposed that as a part of the
>adaptation functionality some of the XR blocks should be used. In this
>case it is very useful for a server to tell the client that it desires
>the client to send XR blocks and which. In that way the client avoids
>having to try to guess or infer that it shall send the XR blocks or not.
>
>Best Regards
>
>Magnus Westerlund
>
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
>Torshamsgatan 23           | Fax   +46 8 7575550
>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Hi,

Here is an updated text of the SDP signaling of the RTCP XR format. 
Please read and comment. I think that I have addressed the comments I 
received. I am sorry that I has taken this time but I have been ill.

Best Regards

Magnus.

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


5. SDP Signaling

   This section defines an SDP attribute that can be used for signaling
   in applications utilizing SDP [s3].  This is signaling is defined to
   be used with Offer-Answer [s1] and for applications using SDP to
   describe media and transport configurations, like in SAP [s4] and
   RTSP [s5]. There exist other potential signaling methods, which are
   not defined here.

   The XR blocks MAY be used without prior signaling.  This is in line
   with other RTCP packet types and follows the normal rules as written
   in [7]. Reasons for this can be an application that requires that one
   or more XR block is always used.  This may require the application to
   define this, if all implementations of this application must work in
   the same way. However for applications that is configured at session
   initiation, the use of some type of signaling is RECOMMENDED.  If any
   future defined report block does not allow usage without signaling,
   so should be noted and motivated in its specification.

   If an application is using the SDP signaling defined here it needs to
   be done as defined below.  It is the usage of SDP for signaling the
   are optional, not if one is going to follow it or not, if used.  If
   SDP signaling is used in an environment where XR blocks are only
   implemented by some fraction of the participants, the one's not
   implementing the XR blocks will ignore the SDP attribute.

5.1 The SDP attribute

   This section defines one new SDP attribute "rtcp-xr" that can be used
   to signal the participants in a media session that it should utilize
   any specified XR block(s).  This attribute can in the future be
   easily extended with new parameters to cover any new reporting
   formats.



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Internet-Draft                  RTCP XR                     2 March 2003


   The RTCP XR format's SDP attribute is defined below in ABNF [s2].  It
   is an session and media level attribute.  When specified at session
   level it applies to all media blocks part of the session.  Any media
   level specification MUST replace an session level specification, if
   present, for that media block.

   rtcp-xr-attrib = "a=" "rtcp-xr" ":" [xr-format *(SP xr-format)] CRLF


   xr-format = pckt-loss-rle
             / dupp-rle
             / recv-ts
             / stat-sum
             / rcv-rtt
             / voip-stat
             / format-ext


   pckt-loss-rle = "rle-pckt" ["=" max-size]
   dupp-rle      = "rle-dup" ["=" max-size]
   recv-ts       = "recv-ts" ["=" max-size]
   stat-sum      = "stat-sum"
   recv-rtt      = "recv-rtt" "=" recv-rtt-mode [":" max-size]
   voip-stat     = "voip-sum"
   max-size      = 1*DIGIT ; Maximum block size in bytes
   recv-rtt-mode = "all"
                 / "sender"
   format-ext    = non-ws-string

   non-ws-string = 1*(%x21-FF)

   The "rtcp-xr" attribute contains zero, one or more XR block related
   parameter.  Each parameter defines its own interpretation as this
   will depend on the XR block(s).  The attribute is extensible so that
   any future XR block can define its own parameter(s).  There may be XR
   blocks where an "rtcp-xr" parameter is not applicable.  However, for
   any future defined formats the need of a parameter SHALL be
   evaluated.

   The defined "rtcp-xr" parameters belong to two categories.  The first
   category, the unilateral parameters,  corresponds to reporting blocks
   that simple reports on the RTP stream and related things.  The second
   category, collaborative parameters, represents XR block(s) that
   requires more than one party to perform an action to accomplish the
   functionality, like the RTT measuring functionality for a RTP
   receiver. These collaborative functionalities MAY use two or more
   different XR blocks, that together provide the functionality.




Friedman et al.             Standards Track            FORMFEED[Page 33]





Internet-Draft                  RTCP XR                     2 March 2003


   For the unilateral category, this document defines the following
   parameters.  The parameter names and their corresponding XR format is
   given in table 1.


   Parameter name    XR format (Nr and Name)
   --------------------------
   rle-pckt          1  Loss RLE Report Block
   rle-dup           2  Duplicate RLE Report Block
   recv-ts           3  Timestamp Report Block
   stat-sum          4  Statistics Summary Report Block
   voip-sum          7  VoIP Metrics Report Block

   Table 1. The unilateral parameters and their XR formats.


   The "rle-pckt", "rle-dup", and "recv-ts" parameters MAY specify an
   integer value.  This value indicates the largest size the whole
   report block SHOULD have in bytes.  This shall be seen as an
   indication that thinning shall be applied if necessary to meet the
   target size.

   Any functionality belongs to the collaborative category, if it
   consist of one initiator block, and one or more types of response
   blocks. The functionality SHOULD include signaling to indicate which
   receivers that are required to respond to the initiator block.

   The collaborative category, currently consist only of the RTT
   measuring for RTP receivers functionality, which uses the XR formats
   "Receiver Timestamp Report Block" (5) and "DLRR Report Block" (6).
   These two formats and their functionality is represented by the
   "recv-rtt" parameter.  This parameter takes first a mode value and
   optionally a maximum size for the DLRR report block.  The mode
   values: "all" indicates that both RTP senders and receivers MAY send
   DLRR blocks. "sender" indicates that only active RTP senders MAY send
   DLRR blocks, i.e. non RTP senders SHALL NOT send DLRR blocks. If a
   maximum size in bytes is included any sent DLRR block SHALL NOT
   exceed the given size. The DLRR block sender SHOULD report on
   different timestamp sources in a round robin fashion if not all
   sources can be reported on due to size limitations.

   Any XR-block not represented in the parameter list of the attribute
   SHALL NOT be sent.  This is required so that bandwidth is not wasted
   and may be specially important for collaborative parameters, where a
   initiator block otherwise could trigger responses from all
   participants in a large multicast group.

   The "rtcp-xr" attributes parameter list MAY be empty.  This is useful



Friedman et al.             Standards Track            FORMFEED[Page 34]





Internet-Draft                  RTCP XR                     2 March 2003


   in cases where one needs to signal that one understands the SDP
   signaling but do not want to use any of the XR functionality.  One
   such example could be a SIP controlled session where one want to stop
   using all XR blocks and therefore remove all parameters in the re-
   invite's SDP.  If one do not desire to use any XR blocks from the
   beginning it is RECOMMENDED that one does not include the "rtcp-xr"
   attribute at all.

5.2 Usage in Offer-Answer

   When having reporting blocks, it is in the interest of the sender of
   media, to request that the receiver sends reports on this media
   stream. However the interpretation will be dependent on any direction
   attribute. In the below definition, text applying for media stream
   with "sendrecv" attributes does also apply to media streams without
   any direction attribute.  The definitions apply only for media
   streams with a unicast destination, unless otherwise noted.

   For media streams in offers with a "sendonly" or a "sendrecv"
   attribute the inclusion of "rtcp-xr" attributes with parameters
   belonging to the unilateral category, has the interpretation that the
   receiver of the offer SHOULD send the XR formats corresponding to the
   parameter to the sender of these streams.  When doing the answer the
   answerer MAY include the "rtcp-xr" attribute with any unilateral
   parameter for media streams with the "sendrecv" attribute to request
   that the offerer SHOULD send the corresponding XR blocks from the
   offerer to the answerer.

   For media streams in offers with "recvonly" attribute the offerer MAY
   include any unilateral parameters, this declare its capability of
   sending the corresponding XR block.  If the answerer includes a
   "rtcp-xr" attribute with an unilateral parameter the offerer SHOULD
   send the corresponding XR block.

   For media streams with multicast destinations the inclusion of an
   "rtcp-xr attribute with any unilateral parameters means that any
   media recipient SHOULD send the corresponding XR block(s).

   For an SDP offer with media streams containing the "rtcp-xr"
   attribute with an collaborative parameter, like "recv-rtt", the
   offerer declares itself capable of receiving the indicated initiator
   ("Receiver Timestamp Report Block") and send responses using the
   responder block ("DLRR Report Block").  This means that the offerer
   has offered a situation where: The answerer MAY send the indicated
   initiator and the offerer SHOULD send the response block(s) if an
   initiator is received. The sending of a response can be preveted, as
   the offerer may not belong to the group that has been declared by the
   parameter to be allowed to send. The response may also be prevented



Friedman et al.             Standards Track            FORMFEED[Page 35]





Internet-Draft                  RTCP XR                     2 March 2003


   by other factors, like congestion control or MTU.  The Answerer MAY
   include a collaborative parameter, like "recv-rtt" in the answer, and
   thereby declare its capability to receive and respond according to
   the indicated functionality.  If a collaborative functionality is
   declared by the Answerer the Offerer MAY send the initiator block to
   the Answerer and the Answerer SHOULD receive the initiator block and
   send the corresponding response block(s).

   For an SDP offer with media streams with multicast destinations
   containing the "rtcp-xr" attribute with an collaborative parameter,
   the parameter needs to indicate which participants that should
   respond to a received initiator.  A SDP offerer according to above
   has the interpretation that the indicated participants SHOULD send a
   response if a initiator is received.  A participant MAY send a
   initiator block.  A participant not indicated to respond for a
   functionality SHALL NOT send any response.  The answerer to a session
   with multicast destination SHALL NOT include any collaborative
   parameters in the response.

   If one receives an SDP offer and understands the "rtcp-xr" attribute
   but does not want to use any XR functionality or it does not apply in
   the answer, the answerer SHOULD include an "rtcp-xr" attribute
   without parameters.  This allows a party to declare capability to
   understand the signaling, and not request any functionality.

5.3 Usage out-side of Offer/Answer

   This applies to the usage of SDP that is currently defined in SAP and
   RTSP where SDP is used to declare what applies for a multi-media
   presentation.  This SDP applies to how a receiver of these media
   streams shall behave to receive the media and under what rules.
   Therefore the usage of the "rtcp-xr" format is simpler in this case
   then compared to offer/answer, however the functionality goal is the
   same.

   For media streams having the "rtcp-xr" attribute with any unilateral
   parameter, the receiver of this media stream SHOULD send the
   corresponding XR block.  If the attribute contains a collaborative
   parameter, the participants indicated by the collaborative parameter
   SHOULD send the response blocks, if the functionalities initiator
   block is received. A session participant MAY send initiator block if
   the collaborative parameter indicates that any participants will send
   responses.

   When the "rtcp-xr" attribute is present no other XR blocks than the
   ones indicated by parameters to the attribute SHALL be used.  This
   means that inclusion of a "rtcp-xr" attribute without any parameter,
   tells a participant that he SHALL NOT send any XR blocks at all.



Friedman et al.             Standards Track            FORMFEED[Page 36]





Internet-Draft                  RTCP XR                     2 March 2003


6. IANA Considerations

   This document defines a new RTCP packet type, the extended report
   (XR) type, within the existing Internet Assigned Numbers Authority
   (IANA) registry of RTP RTCP Control Packet Types.  This document also
   defines a new IANA registry: the registry of RTP XR Block Types.
   Within this new registry, this document defines an initial set of
   seven block types and describes how the remaining types are to be
   allocated.  Further this document defines a new SDP attribute
   "a=rtcp-xr" to be registered, and for its parameters a IANA registry
   is defined.

6.1 XR Packet Type

   The IANA SHALL register the RTP extended report (XR) packet defined
   by this document as packet type 207 in the registry of RTP RTCP
   Control Packet types (PT).

6.2 RTP XR Block Type Registry

   The IANA SHALL create the RTP XR Block Type Registry to cover the
   name space of the extended report block type (BT) field specified in
   Section 3 of this document.  The BT field contains eight bits,
   allowing 256 values.  The IANA SHALL manage the RTP XR Block Type
   Registry according to the Specification Required policy of RFC 2434
   [6].  Future specifications SHOULD attribute block type values in
   strict numeric order following the values attributed in this
   document:


      BT  name
      --  ----
       1  Loss RLE Report Block
       2  Duplicate RLE Report Block
       3  Timestamp Report Block
       4  Statistics Summary Report Block
       5  Receiver Timestamp Report Block
       6  DLRR Report Block
       7  VoIP Metrics Report Block


   The BT value 255 is reserved for future extensions.

   Furthermore, future specifications SHOULD avoid the value 0.  Doing
   so facilitates packet validity checking, since an all-zeros field
   might commonly be found in an ill-formed packet.

6.3 The "rtcp-xr" SDP attribute



Friedman et al.             Standards Track            FORMFEED[Page 37]





Internet-Draft                  RTCP XR                     2 March 2003


   One SDP attribute is to be registered using the below information:

   SDP Attribute ("att-field"):

     Attribute name:     rtcp-xr
     Long form:          RTP Extended Report Parameters
     Type of name:       att-field
     Type of attribute:  Session and Media level
     Subject to charset: No
     Purpose:            See this document
     Reference:          This document
     Values:             See this document and registrations below

   The attribute has an extensible parameter field and therefore a
   registry for these parameters are needed.  IANA is requested to
   create one "rtcp-xr" SDP parameter registry and register the defined
   values. This registry is managed according to the Specification
   Required policy of RFC 2434.  Any registration MUST contain the
   following information:

   - Contact information of the one doing the registration, including at
     least name, address, and email.

   - An ABNF definition of the parameter, in accordance with the
     "format-ext" definition.

   - A description of what the parameter represents and how it shall be
     interpreted, both normally and in Offer/Answer.

   This document registers the six parameters defined in chapter 5.1,
   "rle-pckt", "rle-dup", "recv-ts", "stat-sum", "voip-sum", and "recv-
   rtt".

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: Colin Perkins <csp@csperkins.org>
CC: alan@telchemy.com, "'IETF AVT WG'" <avt@ietf.org>
Subject: Re: [AVT] SDP signaling for RTCP XR format
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Hi,

When I wrote the SDP section I have considered the following use cases:

- RTSP controlled streaming.
- Mbone type lecture session with extra feedback.
- SIP controlled conversational sessions. Two parties.

Are there any more that we really need to consider? I would imagine that 
some more obscure cases may also very well work. The SIP case may be the 
weakest as I don't have practical experience of SIP/SDP. So I would 
really appreciate review by someone that has that background.

Best Regards

Magnus

Colin Perkins wrote:

>Alan,
>
>I'm not so concerned about the number of scenarios, but rather those that
>we expect to be commonly used. How do you expect RTCP XR to be used? 
>
>Colin
>  
>
>  
>

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
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To: Julian.chesterfield@cl.cam.ac.uk, schooler@research.att.com,
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Julian et al.,

As you know, section 10.2 of your "RTCP SSM" Internet Draft references the 
"SDP source filter" draft - the current version of which can be found at 
<http://www.live.com/draft-mmusic-sdp-srcfilter-04.txt>.

Note that the syntax of SDP source filters has changed somewhat since you 
wrote section 10.2.  So, please change both occurrences of
	a=incl:
to
	a=source-filter: incl

     Ross.

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Subject: [AVT] CFP: 11th IEEE International Conference on Network Protocols (ICNP'03)
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                       CALL FOR PAPERS

11th IEEE International Conference on Network Protocols (ICNP'03)

                      Atlanta, Georgia
                     November 4-7, 2003

                  http://icnp03.cc.gatech.edu

ICNP is a highly selective conference dealing with all aspects of
communication protocols including design, specification, analysis,
verification, implementation, and performance.  The program consists
of a full day of tutorials followed by a single three-day track of
peer-reviewed papers, including a limited number of invited talks and
panels.  The web page will contain up-to-date information on the
conference program and tutorials.

High-quality papers dealing with any aspect of network protocols are
solicited for submission to the conference.  Topics of particular
interest include, but are not limited to:

protocol testing and analysis        protocol design and implementation
network measurement and monitoring           ad-hoc and sensor networks
peer-to-peer/overlay protocols                        routing protocols
wireless and mobile networks                                   security
multimedia                                           distributed gaming
QoS                                                           signaling
flow and congestion control                active/programmable networks

IMPORTANT DATES:

Register title, abstract and author affiliations:   May 9, 2003
Paper submission (hard deadline):                   May 16, 2003
Acceptance:                                         July 25, 2003
Camera ready:                                       August 25, 2003

GENERAL CO-CHAIRS:

Kevin Almeroth, UC Santa Barbara
Ken Calvert, U. of Kentucky

PROGRAM CO-CHAIRS:

Jorg Liebeherr, U. of Virginia
Ellen Zegura, Georgia Tech

PROGRAM COMMITTEE:

Kevin Almeroth, UC Santa Barbara
Mostafa Ammar, Georgia Tech
Paul Barford, U. of Wisconsin
Bobby Bhattacharjee, U. of Maryland
Ernst Biersack, Eurecom
John Byers, Boston University
Andrew Campbell, Columbia University
Reuven Cohen, Technion
John Crowcroft, U. of Cambridge
Constantinos Dovrolis, Georgia Tech
Wolfgang Effelsberg, U. of Mannheim
Magda El Zarki, UC Irvine
Michalis Faloutsos, UC Riverside
Serge Fdida, U. of Pierre and Marie Curie
Lixin Gao, U. of Massachusetts
Timothy G. Griffin, AT&T Research
Katie Guo, Bell Labs
Teruo Higashino, Osaka University
Sugih Jamin, U. of Michigan
Ed Knightly, Rice University
Jim Kurose, U. of Massachusetts
Tom Laporta, Penn State University
Brian Levine, U. of Massachusetts
Steven Low, CalTech
Ibrahim Matta, Boston University
Sue Moon, Sprint ATL
Klara Nahrstedt, U. of Illinois
Venkat Padmanabhan, Microsoft Research
Kihong Park, Purdue University
Sanjoy Paul, Bell Labs
Jennifer Rexford, AT&T Research
Dan Rubenstein, Columbia University
Srini Seshan, CMU
R. Srikant, U. of Illinois
Rolf Stadler, Royal Institute of Technology
Ioannis Stavrakakis, University of Athens
Peter Steenkiste, CMU
Joe Touch, ISI
Don Towsley, U. of Massachusetts
Hassan Ural, U. of Ottawa
Jim Xu, Georgia Tech
David Yau, Purdue University

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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--------------Boundary-00=_0FWG70E3FSRIR5YRS3YC
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AVT colleagues,

We have just submitted version 4 of the RTP Control Protocol Extended Rep=
orts=20
(RTCP XR) draft to Internet-Drafts@ietf.org (also attached here).  At thi=
s=20
point, as discussed in San Francisco, we hope to proceed to working group=
=20
last call.

Below is a list of changes from version 3.  At the following URL is a PDF=
=20
letter-format version of the draft:

http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-repor=
t-extns-04.pdf

Regards,

Timur Friedman


List of changes:

* Addition of an applicability statement section.

* Addition of an SDP signalling section.

* Fixed a major item that was overlooked: the definition of the "SSRC
  of source" field was missing from the blocks that used this field.
  Also, the field was entirely missing from the VoIP Metrics Report
  Block, which ordinarily should use it.  The length of the VoIP
  Metrics Report Block was adjusted accordingly.

* Clarified the different roles performed by the report blocks that
  were known as the Timestamp Report Block and Receiver Timestamp
  Report Block.  The names were too similar, and did not reflect the
  completely different roles played by the blocks.  The blocks have
  been renamed the Packet Receipt Times Report Block and the Receiver
  Reference Time Report Block, respectively.  Some explanatory text
  has been added, and in particular the RTP timestamp fields have been
  renamed Receipt time of packet, and their correspondence with RTP
  data packet timestamps discussed.

* Reorganized the discussion of the block types to fall into three
  categories: detailed packet-by-packet reports, reference time
  reports, and statistical reports.  To accomodate this
  categorization, the Statistics Summary Report Block was moved to
  accompany the VoIP Metrics Report Block.  As the description of the
  Statistics Summary Report Block now follows that of the Packet
  Receipt Times Report Block and the Receiver Reference Time Report
  Block, these three have had their block type numbers changed so as
  to remain consecutive in the document.

* Generalized the TTL reporting in the Statistics Summary Report Block
  to cover IPv6 Hop Limit values as well.  This necessitated the
  introduction of a two bit flag, which is 0 if no values are
  reported, 1 if IPv4 TTL values are reported, and 2 if IPv6 Hop Limit
  values are reported.  A flag value of 3 is undefined and disallowed.

* Additional explanatory text has been provided in the Introduction,
  and to introduce the XR packet format.

* Renamed the type-specific data field of the block framework to
  "type-specific block contents".  Just to be clear that what might be
  considered metadata, such as SSRC values, can perfectly well go
  here.

* For the VoIP Metrics Report Block, eliminated the following
  language: "An implementation that sends these blocks SHOULD send at
  least one every ten seconds for the duration of the call, SHOULD
  send one whenever a codec type change or other significant change
  occurs, SHOULD send one when significant call quality degradation is
  detected and SHOULD send one upon call termination."  We cannot
  specify RTCP packet sending intervals that do not correspond with
  the underlying RTCP rules.

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INTERNET-DRAFT                                             16 April 2003
Internet Engineering Task Force                 Expires: 16 October 2003
Audio/Video Transport Working Group

                                                 Timur Friedman, Paris 6
                                                 Ramon Caceres, ShieldIP
                                                 Alan Clark, Telchemy
                                                 Editors

            RTP Control Protocol Extended Reports (RTCP XR)

                draft-ietf-avt-rtcp-report-extns-04.txt


Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/1id-abstracts.html

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

   This document defines the Extended Report (XR) packet type for the
   RTP Control Protocol (RTCP), and defines how the use of XR packets
   can be signaled by an application if it employs the Session
   Description Protocol (SDP).  XR packets are composed of report
   blocks, and seven block types are defined here.  The purpose of the
   extended reporting format is to convey information that supplements
   the six statistics that are contained in the report blocks used by



Friedman et al.              Standards Track                    [Page 1]
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Internet-Draft                   RTCP XR                      April 2003


   RTCP's Sender Report (SR) and Receiver Report (RR) packets.  Some
   applications, such as multicast inference of network characteristics
   (MINC) or voice over IP (VoIP) monitoring, require other and more
   detailed statistics.  In addition to the block types defined here,
   additional block types may be defined in the future by adhering to
   the framework that this document provides.

Table of Contents

   1.     Introduction ..............................................  3
   1.1    Applicability .............................................  4
   1.2    Terminology ...............................................  6
   2.     XR Packet Format ..........................................  6
   3.     Extended Report Block Framework ...........................  8
   4.     Extended Report Blocks ....................................  9
   4.1    Loss RLE Report Block .....................................  9
   4.1.1  Run Length Chunk .......................................... 15
   4.1.2  Bit Vector Chunk .......................................... 15
   4.1.3  Terminating Null Chunk .................................... 15
   4.2    Duplicate RLE Report Block ................................ 16
   4.3    Packet Receipt Times Report Block ......................... 17
   4.4    Receiver Reference Time Report Block ...................... 20
   4.5    DLRR Report Block ......................................... 21
   4.6    Statistics Summary Report Block ........................... 22
   4.7    VoIP Metrics Report Block ................................. 25
   4.7.1  Packet Loss and Discard Metrics ........................... 26
   4.7.2  Burst Metrics ............................................. 27
   4.7.3  Delay Metrics ............................................. 29
   4.7.4  Signal Related Metrics .................................... 30
   4.7.5  Call Quality or Transmission Quality Metrics .............. 33
   4.7.6  Configuration Parameters .................................. 34
   4.7.7  Jitter Buffer Parameters .................................. 35
   5.     SDP Signaling ............................................. 36
   5.1    The SDP Attribute ......................................... 36
   5.2    Usage in Offer/Answer ..................................... 39
   5.3    Usage Outside of Offer/Answer ............................. 40
   6.     IANA Considerations ....................................... 41
   6.1    XR Packet Type ............................................ 41
   6.2    RTCP XR Block Type Registry ............................... 41
   6.3    The "rtcp-xr" SDP Attribute ............................... 42
   7.     Security Considerations ................................... 43
   A.     Algorithms ................................................ 44
   A.1    Sequence Number Interpretation ............................ 44
   A.2    Example Burst Packet Loss Calculation ..................... 45
          Intellectual Property ..................................... 47
          Full Copyright Statement .................................. 48
          Acknowledgments ........................................... 48
          Contributors .............................................. 48



Friedman et al.              Standards Track                    [Page 2]
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Internet-Draft                   RTCP XR                      April 2003


          Authors' Addresses ........................................ 49
          References ................................................ 51
          Normative References ...................................... 51
          Non-Normative References .................................. 52

1. Introduction

   This document defines the Extended Report (XR) packet type for the
   RTP Control Protocol (RTCP) [10], and defines how the use of XR
   packets can be signaled by an application if it employs the Session
   Description Protocol (SDP) [4].  XR packets convey information beyond
   that already contained in the reception report blocks of RTCP's
   sender report (SR) or Receiver Report (RR) packets.  The information
   is of use across RTP profiles, and so is not appropriately carried in
   SR or RR profile-specific extensions.  Information used for network
   management falls into this category, for instance.

   The definition is broken out over the three sections that follow the
   Introduction.  Section 2 defines the XR packet as consisting of an
   eight octet header followed by a series of components called report
   blocks.  Section 3 defines the common format, or framework,
   consisting of a type and a length field, required for all report
   blocks.  Section 4 defines several specific report block types.
   Other block types can be defined in future documents as the need
   arises.

   The report block types defined in this document fall into three
   categories.  The first category consists of packet-by-packet reports
   on received or lost RTP packets.  Reports in the second category
   convey reference time information between RTP participants.  In the
   third category, reports convey metrics relating to packet receipts,
   that are summary in nature but that are more detailed, or of a
   different type, than that conveyed in existing RTCP packets.

   All told, seven report block formats are defined by this document.
   Of these, three are packet-by-packet block types:

   - Loss RLE Report Block (Section 4.1): Run length encoding of reports
   concerning the losses and receipts of RTP packets.

   - Duplicate RLE Report Block (Section 4.2): Run length encoding of
   reports concerning duplicates of received RTP packets.

   - Packet Receipt Times Report Block (Section 4.3): A list of
   reception timestamps of RTP packets.

   There are two reference time related block types:




Friedman et al.              Standards Track                    [Page 3]
=0C
Internet-Draft                   RTCP XR                      April 2003


   - Receiver Reference Time Report Block (Section 4.4): Receiver-end
   wallclock timestamps.  Together with the DLRR Report Block mentioned
   next, these allow non-senders to calculate round-trip times.

   - DLRR Report Block (Section 4.5): The delay since the last Receiver
   Reference Time Report Block was received.  An RTP data sender that
   receives a Receiver Reference Time Report Block can respond with a
   DLRR Report Block, in much the same way as, in the mechanism already
   defined for RTCP [10, Section 6.3.1], an RTP data receiver that
   receives a sender's NTP timestamp can respond by filling in the DLSR
   field of an RTCP reception report block.

   Finally, this document defines two summary metric block types:

   - Statistics Summary Report Block (Section 4.6): Statistics on RTP
   packet sequence numbers, losses, duplicates, jitter, and TTL or Hop
   Limit values.

   - VoIP Metrics Report Block (Section 4.7): Metrics for monitoring
   Voice over IP (VoIP) calls.

   Before proceeding to the XR packet and report block definitions, this
   document provides an applicability statement (Section 1.1) that
   describes the contexts in which these report blocks can be used.  It
   also defines (Section 1.2) the normative use of key words, such as
   MUST and SHOULD, as they are employed in this document.

   Following the definitions of the various report blocks, this document
   describes how applications that employ SDP can signal their use
   (Section 5).  The document concludes with a discussion (Section 6) of
   numbering considerations for the Internet Assigned Numbers Authority
   (IANA), of security considerations (Section 7), and with appendices
   that provide examples of how to implement algorithms discussed in the
   text.

1.1 Applicability

   The XR packets are useful across multiple applications, and for that
   reason are not defined as profile-specific extensions to RTCP sender
   or Receiver Reports [10, Section 6.4.3].  Nonetheless, they are not
   of use in all contexts.  In particular, the VoIP metrics report block
   (Section 4.7) is specific to voice applications, though it can be
   employed over a wide variety of such applications.

   The VoIP metrics report block can be applied to any conversational,
   multicast or broadcast application for which the use of RTP and RTCP
   is specified.  The use of conversational metrics (Section 4.7.5),
   including the R factor (as described by the E Model defined in [3])



Friedman et al.              Standards Track                    [Page 4]
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Internet-Draft                   RTCP XR                      April 2003


   and the mean opinion score for conversational quality (MOS-CQ), in
   applications other than simple two party calls is not defined, and
   hence these metrics should be identified as unavailable in
   conferencing, multicast and broadcast applications.

   The packet-by-packet report block types, Loss RLE (Section 4.1),
   Duplicate RLE (Section 4.2), and Packet Receipt Times (Section 4.3),
   have been defined with network tomography applications, such as
   multicast inference of network characteristics (MINC) [11], in mind.
   MINC requires detailed packet receipt traces from multicast session
   receivers in order to infer the gross structure of the multicast
   distribution tree and the parameters, such as loss rates and delays,
   that apply to paths between the branching points of that tree.

   Any real time multicast multimedia application can use the packet-by-
   packet report block types.  Such an application could employ a MINC
   inference subsystem that would provide it with multicast tree
   topology information.  One potential use of such a subsystem would be
   for the identification of high loss regions in the multicast tree and
   the identification of multicast session participants well situated to
   provide retransmissions of lost packets.

   Detailed packet-by-packet reports do not necessarily have to consume
   disproportionate bandwidth with respect to other RTCP packets.  An
   application can cap the size of these blocks.  A mechanism called
   "thinning" is provided for these report blocks, and can be used to
   ensure that they adhere to a size limit by restricting the number of
   packets reported upon within any sequence number interval.  The
   rationale for, and use of this mechanism is described in [13].
   Furthermore, applications might not require report blocks from all
   receivers in order to answer such important questions as where in the
   multicast tree there are paths that exceed a defined loss rate
   threshold.  Intelligent decisions regarding which receivers send
   these report blocks can further restrict the portion of RTCP
   bandwidth that they consume.

   The packet-by-packet report blocks can also be used by dedicated
   network monitoring applications.  For such an application, it might
   be appropriate to allow more than 5% of RTP data bandwidth to be used
   for RTCP packets, thus allowing proportionately larger and more
   detailed report blocks.

   Nothing in the packet-by-packet block types restricts their use to
   multicast applications.  In particular, they could be used for
   network tomography similar to MINC, but using striped unicast
   packets.  In addition, if it were found useful, they could be used
   for applications limited to two participants.




Friedman et al.              Standards Track                    [Page 5]
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Internet-Draft                   RTCP XR                      April 2003


   One use to which the packet-by-packet reports are not immediately
   suited is for data packet acknowledgments as part of a packet
   retransmission mechanism.  The reason is that the packet accounting
   technique suggested for these blocks differs from the packet
   accounting normally employed by RTP.  In order to favor measurement
   applications, an effort is made to interpret as little as possible at
   the data receiver, and leave the interpretation as much as possible
   to participants that receive the report blocks.  Thus, for example, a
   packet with an anomalous SSRC ID or an anomalous sequence number
   might be excluded by normal RTP accounting, but would be reported
   upon for network monitoring purposes.

   The Statistics Summary Report Block (Section 4.6) has also been
   defined with network monitoring in mind.  This block type can be used
   equally well for reporting on unicast and multicast packet reception.

   The reference time related block types were conceived for receiver-
   based TCP-friendly multicast congestion control [17].  By allowing
   data receivers to calculate their round trip times to senders, they
   help the receivers estimate the downstream bandwidth they should
   request.  Note that if every receiver is to send Receiver Reference
   Time Report Blocks (Section 4.4), a sender might potentially send a
   number of DLRR Report Blocks (Section 4.5) equal to the number of
   receivers whose RTCP packets have arrived at the sender within its
   reporting interval.  As the number of participants in a multicast
   session increases, an application should use discretion regarding
   which participants send these blocks, and how frequently.

   The SDP signaling defined for XR packets in this document (Section 5)
   was done so with three use scenarios in mind: a Real Time Streaming
   Protocol (RTSP) controlled streaming application, a one-to-many
   multicast multimedia application such as a course lecture with
   enhanced feedback, and a Session Initiation Protocol (SIP) controlled
   conversational session involving two parties.  Applications that
   employ SDP are free to use additional SDP signaling for cases not
   covered here.  In addition, applications are free to use signaling
   mechanisms other than SDP.

1.2 Terminology

   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 [1] and
   indicate requirement levels for compliance with this specification.

2. XR Packet Format

   An XR packet consists of a header of two 32-bit words, followed by a



Friedman et al.              Standards Track                    [Page 6]
=0C
Internet-Draft                   RTCP XR                      April 2003


   number, possibly zero, of extended report blocks.  This type of
   packet is laid out in a manner consistent with other RTCP packets, as
   concerns the essential version, packet type, and length information.
   XR packets are thus backwards compatibility with RTCP receiver
   implementations that do not recognize them, but that ought to be able
   to parse past them using the length information.  A padding field and
   an SSRC/CSRC field are also provided in the same locations that they
   appear in other RTCP packets, for simplicity.  The format is as
   follows:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=3D2|P|reserved |   PT=3DXR=3D207   |             length            =
|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                           SSRC/CSRC                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                         report blocks                         :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   version (V): 2 bits
        Identifies the version of RTP.  This specification applies to
        RTP version two.

   padding (P): 1 bit
        If the padding bit is set, this XR packet contains some
        additional padding octets at the end.  The semantics of this
        field are identical to the semantics of the padding field in the
        SR packet, as defined by the RTP specification.

   reserved: 5 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   packet type (PT): 8 bits
        Contains the constant 207 to identify this as an RTCP XR packet.
        This value is registered with the Internet Assigned Numbers
        Authority (IANA), as described in Section 5.1.

   length: 16 bits
        As described for the RTCP Sender Report (SR) packet (see Section
        6.3.1 of the RTP specification [10]).  Briefly, the length of
        this XR packet in 32-bit words minus one, including the header
        and any padding.




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   SSRC/CSRC: 32 bits
        The synchronization source identifier for the originator of this
        XR packet.

   report blocks: variable length.
        Zero or more extended report blocks.  In keeping with the
        extended report block framework defined below, each block MUST
        consist of one or more 32-bit words.

3. Extended Report Block Framework

   Extended report blocks are stacked, one after the other, at the end
   of an XR packet.  An individual block's length is a multiple of 4
   octets.  The XR header's length field describes the total length of
   the packet, including these extended report blocks.

   Each block has block type and length fields that facilitate parsing.
   A receiving application can demultiplex the blocks based upon their
   type, and can use the length information to locate each successive
   block, even in the presence of block types it does not recognize.

   An extended report block has the following format:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      BT       | type-specific |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :             type-specific block contents                      :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        Identifies the block format.  Seven block types are defined in
        Section 4.  Additional block types may be defined in future
        specifications.  This field's name space is managed by the
        Internet Assigned Numbers Authority (IANA), as described in
        Section 5.2.

   type-specific: 8 bits
        The use of these bits is determined by the block type
        definition.

   block length: 16 bits
        The length of this report block including the header, in 32-bit
        words minus one.  If the block type definition permits, zero is
        an acceptable value, signifying a block that consists of only



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        the BT, type-specific, and block length fields, with a null
        type-specific block contents field.

   type-specific block contents: variable length
        The use of this field is defined by the particular block type,
        subject to the constraint that it MUST be a multiple of 32 bits
        long.  If the block type definition permits, It MAY be zero bits
        long.

4. Extended Report Blocks

   This section defines seven extended report blocks: block types for
   reporting upon received packet losses and duplicates, packet
   reception times, receiver reference time information, receiver inter-
   report delays, detailed reception statistics, and voice over IP
   (VoIP) metrics.  An implementation SHOULD ignore incoming blocks with
   types either not relevant or unknown to it. Additional block types
   MUST be registered with the Internet Assigned Numbers Authority
   (IANA) [7], as described in Section 5.2.

4.1 Loss RLE Report Block

   This block type permits detailed reporting upon individual packet
   receipt and loss events.  Such reports can be used, for example, for
   multicast inference of network characteristics (MINC) [11].  With
   MINC, one can discover the topology of the multicast tree used for
   distributing a source's RTP packets, and of the loss rates along
   links within that tree.  Or they could be used to provide raw data to
   a network management application.

   Since a Boolean trace of lost and received RTP packets is potentially
   lengthy, this block type permits the trace to be compressed through
   run length encoding.  To further reduce block size, loss event
   reports can be systematically dropped from the trace in a mechanism
   called thinning that is described below and that is studied in [13].

   A participant that generates a Loss RLE Report Block should favor
   accuracy in reporting on observed events over interpretation of those
   events whenever possible.  Interpretation should be left to those who
   observe the report blocks.  Following this approach implies that
   accounting for Loss RLE Report Blocks will differ from the accounting
   for the generation of the SR and RR packets described in the RTP
   specification [10] in the following two areas: per-sender accounting
   and per-packet accounting.

   In its per-sender accounting, an RTP session participant SHOULD NOT
   make the receipt of a threshold minimum number of RTP packets a
   condition for reporting upon the sender of those packets.  This



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   accounting technique differs from the technique described in Section
   6.2.1 and Appendix A.1 of the RTP specification that allows a
   threshold to determine whether a sender is considered valid.

   In its per-packet accounting, an RTP session participant SHOULD treat
   all sequence numbers as valid.  This accounting technique differs
   from the technique described in Appendix A.1 of the RTP specification
   that suggests ruling a sequence number valid or invalid on the basis
   of its contiguity with the sequence numbers of previously received
   packets.

   Sender validity and sequence number validity are interpretations of
   the raw data.  Such interpretations are justified in the interest,
   for example, of excluding the stray old packet from an unrelated
   session from having an effect upon the calculation of the RTCP
   transmission interval.  The presence of stray packets might, on the
   other hand, be of interest to a network monitoring application.

   One accounting interpretation that is still necessary is for a
   participant to decide whether the 16 bit sequence number has rolled
   over.  Under ordinary circumstances this is not a difficult task.
   For example, if packet number 65,535 (the highest possible sequence
   number) is followed shortly by packet number 0, it is reasonable to
   assume that there has been a rollover.  However it is possible that
   the packet is an earlier one (from 65,535 packets earlier).  It is
   also possible that the sequence numbers have rolled over multiple
   times, either forward or backward.  The interpretation becomes more
   difficult when there are large gaps between the sequence numbers,
   even accounting for rollover, and when there are long intervals
   between received packets.

   The per-packet accounting technique mandated here is for a
   participant to keep track of the sequence number of the packet most
   recently received from a sender.  For the next packet that arrives
   from that sender, the sequence number MUST be judged to fall no more
   than 32,768 packets ahead or behind the most recent one, whichever
   choice places it closer.  In the event that both choices are equally
   distant (only possible when the distance is 32,768), the choice MUST
   be the one that does not require a rollover.  Appendix A.1 presents
   an algorithm that implements this technique.

   Each block reports on a single RTP data packet source, identified by
   its SSRC.  The receiver that is supplying the report is identified in
   the header of the RTCP packet.

   Choice of beginning and ending RTP packet sequence numbers for the
   trace is left to the application.  These values are reported in the
   block.  The last sequence number in the trace MAY differ from the



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   sequence number reported on in any accompanying SR or RR report.

   Note that because of sequence number wraparound the ending sequence
   number MAY be less than the beginning sequence number.  A Loss RLE
   Report Block MUST NOT be used to report upon a range of 65,534 or
   greater in the sequence number space, as there is no means to
   identify multiple wraparounds.

   The trace described by a Loss RLE report consists of a sequence of
   Boolean values, one for each sequence number of the trace.  A value
   of one represents a packet receipt, meaning that one or more packets
   having that sequence number have been received since the most recent
   wraparound of sequence numbers (or since the beginning of the RTP
   session if no wraparound has been judged to have occurred).  A value
   of zero represents a packet loss, meaning that there has been no
   packet receipt for that sequence number as of the time of the report.
   If a packet with a given sequence number is received after a report
   of a loss for that sequence number, a later Loss RLE report MAY
   report a packet receipt for that sequence number.

   The encoding itself consists of a series of 16 bit units called
   chunks that describe sequences of packet receipts or losses in the
   trace.  Each chunk either specifies a run length or a bit vector, or
   is a null chunk.  A run length describes between 1 and 16,383 events
   that are all the same (either all receipts or all losses).  A bit
   vector describes 15 events that may be mixed receipts and losses.  A
   null chunk describes no events, and is used to round out the block to
   a 32 bit word boundary.

   The mapping from a sequence of lost and received packets into a
   sequence of chunks is not necessarily unique.  For example, the
   following trace covers 45 packets, of which the 22nd and 24th have
   been lost and the others received:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1111 1


   One way to encode this would be:


   bit vector 1111 1111 1111 111
   bit vector 1111 1101 0111 111
   bit vector 1111 1111 1111 111
   null chunk


   Another way to encode this would be:



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   run of 21 receipts
   bit vector 0101 1111 1111 111
   run of 9 receipts
   null chunk


   The choice of encoding is left to the application.  As part of this
   freedom of choice, applications MAY terminate a series of run length
   and bit vector chunks with a bit vector chunk that runs beyond the
   sequence number space described by the report block.  For example, if
   the 44th packet in the same sequence were lost:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1110 1


   This could be encoded as:


   run of 21 receipts
   bit vector 0101 1111 1111 111
   bit vector 1111 1110 1000 000
   null chunk


   In this example, the last five bits of the second bit vector describe
   a part of the sequence number space that extends beyond the last
   sequence number in the trace.  These bits have been set to zero.

   All bits in a bit vector chunk that describe a part of the sequence
   number space that extends beyond the last sequence number in the
   trace MUST be set to zero, and MUST be ignored by the receiver.

   A null packet MUST appear at the end of a Loss RLE Report Block if
   the number of run length plus bit vector chunks is odd.  The null
   chunk MUST NOT appear in any other context.

   Caution should be used in sending Loss RLE Report Blocks because,
   even with the compression provided by run length encoding, they can
   easily consume bandwidth out of proportion with normal RTCP packets.
   The block type includes a mechanism, called thinning, that allows an
   application to limit report sizes.

   A thinning value, T, selects a subset of packets within the sequence
   number space: those with sequence numbers that are multiples of 2^T.
   Packet reception and loss reports apply only to those packets.  T can
   vary between 0 and 15.  If T is zero then every packet in the
   sequence number space is reported upon.  If T is fifteen then one in



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   every 32,768 packets is reported upon.

   Suppose that the trace just described begins at sequence number
   13,821.  The last sequence number in the trace is 13,865.  If the
   trace were to be thinned with a thinning value of T=3D2, then the
   following sequence numbers would be reported upon: 13,824, 13,828,
   13,832, 13,836, 13,840, 13,844, 13,848, 13,852, 13,856, 13,860,
   13,864.  The thinned trace would be as follows:


      1    1    1    1    1    0    1    1    1    1    0


   This could be encoded as follows:


   bit vector 1111 1011 1100 000
   null chunk


   The last four bits in the bit vector, representing sequence numbers
   13,868, 13,872, 13,876, and 13,880, extend beyond the trace and are
   thus set to zero and are ignored by the receiver.  With thinning, the
   loss of the 22nd packet goes unreported because its sequence number,
   13,842, is not a multiple of four.  Packet receipts for all sequence
   numbers that are not multiples of four also go unreported.  However,
   in this example thinning has permitted the Loss RLE Report Block to
   be shortened by one 32 bit word.

   Choice of the thinning value is left to the application.

   The Loss RLE Report Block has the following format:



















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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D1      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Loss RLE Report Block is identified by the constant 1.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        The amount of thinning performed on the sequence number space.
        Only those packets with sequence numbers 0 mod 2^T are reported
        on by this block.  A value of 0 indicates that there is no
        thinning, and all packets are reported on.  The maximum thinning
        is one packet in every 32,768 (amounting to two packets within
        each 16-bit sequence space).

   block length: 16 bits
        Defined in Section 3.

   SSRC of source: 32 bits
        The SSRC of the RTP data packet source being reported upon by
        this report block.

   begin_seq: 16 bits
        The first sequence number that this block reports on.

   end_seq: 16 bits
        The last sequence number that this block reports on plus one.

   chunk i: 16 bits
        There are three chunk types: run length, bit vector, and



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        terminating null, defined in the following sections.  If the
        chunk is all zeroes then it is a terminating null chunk.
        Otherwise, the leftmost bit of the chunk determines its type: 0
        for run length and 1 for bit vector.

4.1.1 Run Length Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|R|        run length         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A zero identifies this as a run length chunk.

   run type (R): 1 bit
        Zero indicates a run of 0s.  One indicates a run of 1s.

   run length: 14 bits
        A value between 1 and 16,383.  The value MUST not be zero for a
        run length chunk (zeroes in both the run type and run length
        fields would make the chunk a terminating null chunk).  Run
        lengths of 15 or less MAY be described with a run length chunk
        despite the fact that they could also be described as part of a
        bit vector chunk.

4.1.2 Bit Vector Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|        bit vector           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A one identifies this as a bit vector chunk.

   bit vector: 15 bits
        The vector is read from left to right, in order of increasing
        sequence number (with the appropriate allowance for wraparound).

4.1.3 Terminating Null Chunk




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   This chunk is all zeroes.


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


4.2 Duplicate RLE Report Block

   This block type permits per-sequence-number reports on duplicates in
   a source's RTP packet stream.  Such information can be used for
   network diagnosis, and provide an alternative to packet losses as a
   basis for multicast tree topology inference.

   The Duplicate RLE Report Block format is identical to the Loss RLE
   Report Block format.  Only the interpretation is different, in that
   the information concerns packet duplicates rather than packet losses.
   The trace to be encoded in this case also consists of zeros and ones,
   but a zero here indicates the presence of duplicate packets for a
   given sequence number, whereas a one indicates that no duplicates
   were received.

   The existence of a duplicate for a given sequence number is
   determined over the entire reporting period.  For example, if packet
   number 12,593 arrives, followed by other packets with other sequence
   numbers, the arrival later in the reporting period of another packet
   numbered 12,593 counts as a duplicate for that sequence number.  The
   duplicate does not need to follow immediately upon the first packet
   of that number.  Care must be taken that a report does not cover a
   range of 65,534 or greater in the sequence number space.

   No distinction is made between the existence of a single duplicate
   packet and multiple duplicate packets for a given sequence number.
   Note also that since there is no duplicate for a lost packet, a loss
   is encoded as a one in a Duplicate RLE Report Block.

   The Duplicate RLE Report Block has the following format:











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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D2      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Duplicate RLE Report Block is identified by the constant 2.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.

   SSRC of source: 32 bits
        As defined in Section 4.1.

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   chunk i: 16 bits
        As defined in Section 4.1.

4.3 Packet Receipt Times Report Block

   This block type permits per-sequence-number reports on packet receipt
   times for a given source's RTP packet stream.  Such information can
   be used for MINC inference of the topology of the multicast tree used



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   to distribute the source's RTP packets, and of the delays along the
   links within that tree.  It can also be used to measure partial path
   characteristics and to model distributions for packet jitter.

   Packet receipt times are expressed in the same units as used in the
   RTP timestamps of data packets.  This is so that, for each packet,
   one can establish both the send time and the receipt time in
   comparable terms.  Note, however, that as an RTP sender ordinarily
   initializes its time to a value chosen at random, there can be no
   expectation that reported send and receipt times will differ by an
   amount equal to the one-way delay between sender and receiver.  The
   reported times can nonetheless be useful for the purposes mentioned
   above.

   At least one packet MUST have been received for each sequence number
   reported upon in this block.  If this block type is used to report
   receipt times for a series of sequence numbers that includes lost
   packets, several blocks are required.  If duplicate packets have been
   received for a given sequence number, and those packets differ in
   their receipt times, any time other than the earliest MUST NOT be
   reported.  This is to ensure consistency among reports.

   Times reported in RTP timestamp format consume more bits than loss or
   duplicate information, and do not lend themselves to run length
   encoding.  The use of thinning is encouraged to limit the size of
   Packet Receipt Times Report Blocks.

   The Packet Receipt Times Report Block has the following format:























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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D3      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet begin_seq                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet (begin_seq + 1) mod 65536        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet (end_seq - 1) mod 65536          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Packet Receipt Times Report Block is identified by the
        constant 3.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.

   SSRC of source: 32 bits
        As defined in Section 4.1.

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   Receipt time of packet i: 32 bits
        The receipt time of the packet with sequence number i at the
        receiver.  The modular arithmetic shown in the packet format
        diagram is to allow for sequence number rollover.  It is
        preferable for the time value to be established at the link



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        layer interface, or in any case as close as possible to the wire
        arrival time.  Units and format are the same as for the
        timestamp in RTP data packets.  As opposed to RTP data packet
        timestamps, in which nominal values may be used instead of
        system clock values in order to convey information useful for
        periodic playout, the receipt times should reflect the actual
        time as closely as possible.  For a session, if the RTP
        timestamp is chosen at random, the first receipt time value
        SHOULD also be chosen at random, and subsequent timestamps
        offset from this value.  On the other hand, if the RTP timestamp
        is meant to reflect the reference time at the sender, then the
        receipt time SHOULD be as close as possible to the reference
        time at the receiver.

4.4 Receiver Reference Time Report Block

   This block extends RTCP's timestamp reporting so that non-senders may
   also send timestamps.  It recapitulates the NTP timestamp fields from
   the RTCP Sender Report [10, Sec. 6.3.1].  A non-sender may estimate
   its round trip time (RTT) to other participants, as proposed in [17],
   by sending this report block and receiving DLRR Report Blocks (see
   next section) in reply.


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D4      |   reserved    |       block length =3D 2        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              NTP timestamp, most significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |             NTP timestamp, least significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Receiver Reference Time Report Block is identified by the
        constant 4.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 2, in accordance with the definition of this field
        in Section 3.




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   NTP timestamp: 64 bits
        Indicates the wallclock time when this block was sent so that it
        may be used in combination with timestamps returned in DLRR
        Report Blocks (see next section) from other receivers to measure
        round-trip propagation to those receivers.  Receivers should
        expect that the measurement accuracy of the timestamp may be
        limited to far less than the resolution of the NTP timestamp.
        The measurement uncertainty of the timestamp is not indicated as
        it may not be known. A report block sender that can keep track
        of elapsed time but has no notion of wallclock time may use the
        elapsed time since joining the session instead. This is assumed
        to be less than 68 years, so the high bit will be zero.  It is
        permissible to use the sampling clock to estimate elapsed
        wallclock time. A report sender that has no notion of wallclock
        or elapsed time may set the NTP timestamp to zero.

4.5 DLRR Report Block

   This block extends RTCP's delay since last Sender Report (DLSR)
   mechanism [10, Sec. 6.3.1] so that non-senders may also calculate
   round trip times, as proposed in [17].  It is termed DLRR for delay
   since last Receiver Report, and may be sent in response to a Receiver
   Timestamp Report Block (see previous section) from a receiver to
   allow that receiver to calculate its round trip time to the
   respondent.  The report consists of one or more 3 word sub-blocks:
   one sub-block per Receiver Report.


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D5      |   reserved    |         block length          |
   +=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+
   |                 SSRC_1 (SSRC of first receiver)               | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ bloc=
k
   |                         last RR (LRR)                         |   1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   delay since last RR (DLRR)                  |
   +=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+
   |                 SSRC_2 (SSRC of second receiver)              | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ bloc=
k
   :                               ...                             :   2
   +=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+


   block type (BT): 8 bits
        A DLRR Report Block is identified by the constant 5.




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   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        Defined in Section 3.

   last RR timestamp (LRR): 32 bits
        The middle 32 bits out of 64 in the NTP timestamp (as explained
        in the previous section) received as part of a Receiver
        Reference Time Report Block from participant SSRC_n. If no such
        block has been received, the field is set to zero.

   delay since last RR (DLRR): 32 bits
        The delay, expressed in units of 1/65536 seconds, between
        receiving the last Receiver Reference Time Report Block from
        participant SSRC_n and sending this DLRR Report Block.  If no
        Receiver Reference Time Report Block has been received yet from
        SSRC_n, the DLRR field is set to zero (or the DLRR is omitted
        entirely). Let SSRC_r denote the receiver issuing this DLRR
        Report Block. Participant SSRC_n can compute the round-trip
        propagation delay to SSRC_r by recording the time A when this
        Receiver Timestamp Report Block is received.  It calculates the
        total round-trip time A-LSR using the last SR timestamp (LSR)
        field, and then subtracting this field to leave the round-trip
        propagation delay as A-LSR-DLSR. This is illustrated in [10,
        Fig. 2].

4.6 Statistics Summary Report Block

   This block reports statistics beyond the information carried in the
   standard RTCP packet format, but not as fine grained as that carried
   in the report blocks previously described.  Information is recorded
   about lost packets, duplicate packets, jitter measurements, and TTL
   or Hop Limit values.  Such information can be useful for network
   management.

   The report block contents are dependent upon a series of flags bit
   carried in the first part of the header.  Not all parameters need to
   be reported in each block.  Flags indicate which are and which are
   not reported.  The fields corresponding to unreported parameters MUST
   be set to zero. The receiver MUST ignore any Statistics Summary
   Report Block with a non-zero value in any field flagged as
   unreported.

   The Statistics Summary Report Block has the following format:




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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D6      |L|D|J|ToH|rsvd.|       block length =3D 9        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        lost_packets                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        dup_packets                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         min_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         max_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         avg_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         dev_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | min_ttl_or_hl | max_ttl_or_hl | avg_ttl_or_hl | dev_ttl_or_hl |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Statistics Summary Report Block is identified by the constant
        6.

   loss report flag (L): 1 bit
        Bit set to 1 if the lost_packets field contains a report, 0
        otherwise.

   duplicate report flag (D): 1 bit
        Bit set to 1 if the dup_packets field contains a report, 0
        otherwise.

   jitter flag (J): 1 bit
        Bit set to 1 if the min_jitter, max_jitter, avg_jitter, and
        dev_jitter fields all contain reports, 0 if none of them do.

   TTL or Hop Limit flag (ToH): 2 bits
        This field is set to 0 if none of the fields min_ttl_or_hl,
        max_ttl_or_hl, avg_ttl_or_hl, or dev_ttl_or_hl contain reports.
        If the field is non-zero then all of these fields contain
        reports.  The value 1 signifies that they report on IPv4 TTL
        values.  The value 2 signifies that they report on IPv6 Hop
        Limit values.  This value 3 is undefined and MUST NOT be used.



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   rsvd.: 3 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 9, in accordance with the definition of this field
        in Section 3.

   SSRC of source: 32 bits
        As defined in Section 4.1.

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   lost_packets: 32 bits
        Number of lost packets in the above sequence number interval.

   dup_packets: 32 bits
        Number of duplicate packets in the above sequence number
        interval.

   min_jitter: 32 bits
        The minimum relative transit time between two packets in the
        above sequence number interval.  All jitter values are measured
        as the difference between a packet's RTP timestamp and the
        reporter's clock at the time of arrival, measured in the same
        units.

   max_jitter: 32 bits
        The maximum relative transit time between two packets in the
        above sequence number interval.

   avg_jitter: 32 bits
        The average relative transit time between each two packet series
        in the above sequence number interval.

   dev_jitter: 32 bits
        The standard deviation of the relative transit time between each
        two packet series in the above sequence number interval.

   min_ttl_or_hl: 8 bits
        The minimum TTL or Hop Limit value of data packets in the
        sequence number range.




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   max_ttl_or_hl: 8 bits
        The maximum TTL or Hop Limit value of data packets in the
        sequence number range.

   avg_ttl_or_hl: 8 bits
        The average TTL or Hop Limit value of data packets in the
        sequence number range.

   dev_ttl_or_hl: 8 bits
        The standard deviation of TTL or Hop Limit values of data
        packets in the sequence number range.

4.7 VoIP Metrics Report Block

   The VoIP Metrics Report Block provides metrics for monitoring voice
   over IP (VoIP) calls.  These metrics include packet loss and discard
   metrics, delay metrics, analog metrics, and voice quality metrics.
   The block reports separately on packets lost on the IP channel, and
   those that have been received but then discarded by the receiving
   jitter buffer.  It also reports on the combined effect of losses and
   discards, as both have equal effect on call quality.

   In order to properly assess the quality of a Voice over IP call it is
   desirable to consider the degree of burstiness of packet loss [14].
   Following a Gilbert-Elliott model [3], a period of time, bounded by
   lost and/or discarded packets, with a high rate of losses and/or
   discards is a "burst," and a period of time between two bursts is a
   "gap."  Bursts correspond to periods of time during which the packet
   loss rate is high enough to produce noticeable degradation in audio
   quality.  Gaps correspond to periods of time during which only
   isolated lost packets may occur, and in general these can be masked
   by packet loss concealment.  Delay reports include the transit delay
   between RTCP end points and the VoIP end system processing delays,
   both of which contribute to the user perceived delay.  Additional
   metrics include signal, echo, noise, and distortion levels.  Call
   quality metrics include R factors (as described by the E Model
   defined in [3]) and mean opinion scores (MOS scores).

   Implementations MUST provide values for all the fields defined here.
   For certain metrics, if the value is undefined or unknown, then the
   specified default or unknown field value MUST be provided.

   The block is encoded as seven 32-bit words:








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    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=3D7      |   reserved    |       block length =3D 8        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   loss rate   | discard rate  | burst density |  gap density  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       burst duration          |         gap duration          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     round trip delay          |       end system delay        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | signal level  |  noise level  |     RERL      |     Gmin      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   RX config   |   reserved    |          JB nominal           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          JB maximum           |          JB abs max           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A VoIP Metrics Report Block is identified by the constant 7.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   block length: 16 bits
        The constant 8, in accordance with the definition of this field
        in Section 3.

   SSRC of source: 32 bits
        As defined in Section 4.1.

   The remaining fields are described in the following six sections:
   Packet Loss and Discard Metrics, Delay Metrics, Signal Related
   Metrics, Call Quality or Transmission Quality Metrics, Configuration
   Metrics, and Jitter Buffer Parameters.

4.7.1 Packet Loss and Discard Metrics

   It is very useful to distinguish between packets lost by the network
   and those discarded due to jitter. Both have equal effect on the
   quality of the voice stream however having separate counts helps



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   identify the source of quality degradation. These fields MUST be
   populated.

   loss rate: 8 bits
        The fraction of RTP data packets from the source lost since the
        beginning of reception, expressed as a fixed point number with
        the binary point at the left edge of the field.  This value is
        calculated by dividing the total number of packets lost (after
        the effects of applying any error protection such as FEC) by the
        total number of packets expected, multiplying the result of the
        division by 256, limiting the maximum value to 255 (to avoid
        overflow), and taking the integer part.  The numbers of
        duplicated packets and discarded packets do not enter into this
        calculation.  Since receivers cannot be required to maintain
        unlimited buffers, a receiver MAY categorize late-arriving
        packets as lost.  The degree of lateness that triggers a loss
        SHOULD be significantly greater than that which triggers a
        discard.

   discard rate: 8 bits
        The fraction of RTP data packets from the source that have been
        discarded since the beginning of reception, due to late or early
        arrival, under-run or overflow at the receiving jitter buffer.
        This value is expressed as a fixed point number with the binary
        point at the left edge of the field.  It is calculated by
        dividing the total number of packets discarded (excluding
        duplicate packet discards) by the total number of packets
        expected, multiplying the result of the division by 256,
        limiting the maximum value to 255 (to avoid overflow), and
        taking the integer part.

4.7.2 Burst Metrics

   A burst, informally, is a period of high packet losses and/or
   discards.  Formally, a burst is defined as a longest sequence of
   packets bounded by lost or discarded packets with the constraint that
   within a burst any sequence of successive packets that were received,
   and not discarded due to delay variation, is of length less than a
   value Gmin.

   A gap, informally, is a period of low packet losses and/or discards.
   Formally, a gap is defined as any of the following: (a) the period
   from the start of an RTP session to the receipt time of the last
   received packet before the first burst, (b) the period from the end
   of the last burst to either the time of the report or the end of the
   RTP session, whichever comes first, or (c) the period of time between
   two bursts.




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   For the purpose of determining if a lost or discarded packet near the
   start or end of an RTP session is within a gap or a burst it is
   assumed that the RTP session is preceded and followed by at least
   Gmin received packets, and that the time of the report is followed by
   at least Gmin received packets.

   A gap has the property that any lost or discarded packets within the
   gap must be preceded and followed by at least Gmin packets that were
   received and not discarded.  This gives a maximum loss/discard
   density within a gap of: 1 / (Gmin + 1).

   A Gmin value of 16 is RECOMMENDED as it results in gap
   characteristics that correspond to good quality (i.e. low packet loss
   rate, a minimum distance of 16 received packets between lost packets)
   and hence differentiates nicely between good and poor quality
   periods.

   For example, a 1 denotes a received, 0 a lost, and X a discarded
   packet in the following pattern covering 64 packets:


   11110111111111111111111X111X1011110111111111111111111X111111111
   |---------gap----------|--burst---|------------gap------------|


   The burst consists of the twelve packets indicated above, starting at
   a discarded packet and ending at a lost packet.  The first gap starts
   at the beginning of the session and the second gap ends at the time
   of the report.

   If the packet spacing is 10 ms and the Gmin value is the recommended
   value of 16, the burst duration is 120 ms, the burst density 0.33,
   the gap duration 230 ms + 290 ms =3D 520 ms, and the gap density 0.04.

   This would result in reported values as follows (see field
   descriptions for semantics and details on how these are calculated):

   loss density          12, which corresponds to 5%
   discard density       12, which corresponds to 5%
   burst density         84, which corresponds to 33%
   gap density           10, which corresponds to 4%
   burst duration       120, value in milliseconds
   gap duration         520, value in milliseconds

   burst density: 8 bits
        The fraction of RTP data packets within burst periods since the
        beginning of reception that were either lost or discarded.  This
        value is expressed as a fixed point number with the binary point



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        at the left edge of the field.  It is calculated by dividing the
        total number of packets lost or discarded (excluding duplicate
        packet discards) within burst periods by the total number of
        packets expected within the burst periods, multiplying the
        result of the division by 256, limiting the maximum value to 255
        (to avoid overflow), and taking the integer part.

   gap density: 8 bits
        The fraction of RTP data packets within inter-burst gaps since
        the beginning of reception that were either lost or discarded.
        The value is expressed as a fixed point number with the binary
        point at the left edge of the field.  It is calculated by
        dividing the total number of packets lost or discarded
        (excluding duplicate packet discards) within gap periods by the
        total number of packets expected within the gap periods,
        multiplying the result of the division by 256, limiting the
        maximum value to 255 (to avoid overflow), and taking the integer
        part.

   burst duration: 16 bits
        The mean duration, expressed in milliseconds, of the burst
        periods that have occurred since the beginning of reception.
        The duration of each period is calculated based upon the packets
        that mark the beginning and end of that period.  It is equal to
        the timestamp of the end packet, plus the duration of the end
        packet, minus the timestamp of the beginning packet.  If the
        actual values are not available, estimated values MUST be used.
        If there have been no burst periods, the burst duration value
        MUST be zero.

   gap duration: 16 bits
        The mean duration, expressed in milliseconds, of the gap periods
        that have occurred since the beginning of reception.  The
        duration of each period is calculated based upon the packet that
        marks the end of the prior burst and the packet that marks the
        beginning of the subsequent burst. It is equal to the timestamp
        of the subsequent burst packet, minus the timestamp of the prior
        burst packet, plus the duration of the prior burst packet.  If
        the actual values are not available, estimated values MUST be
        used.  In the case of a gap that occurs at the beginning of
        reception, the sum of the timestamp of the prior burst packet
        and the duration of the prior burst packet are replaced by the
        reception start time.  In the case of a gap that occurs at the
        end of reception, the timestamp of the subsequent burst packet
        is replaced by the reception end time.  If there have been no
        gap periods, the gap duration value MUST be zero.

4.7.3 Delay Metrics



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   For the purpose of the following definitions, the RTP interface is
   the interface between the RTP instance and the voice application
   (i.e.  FEC, de-interleaving, de-multiplexing, jitter buffer). For
   example, the time delay due to RTP payload multiplexing would be
   considered to be part of the voice application or end-system delay
   whereas delay due to multiplexing RTP frames within a UDP frame would
   be considered part of the RTP reported delay.  This distinction is
   consistent with the use of RTCP for delay measurements.

   round trip delay: 16 bits
        The most recently calculated round trip time between RTP
        interfaces, expressed in milliseconds. This value MAY be
        measured using RTCP, the DLRR method defined in Section 4.5 of
        this document or other approaches.  If RTCP is used then the
        reported delay value is the time of receipt of the most recent
        RTCP packet from source SSRC, minus the LSR (last SR) time
        reported in its SR (Sender Report), minus the DLSR (delay since
        last SR) reported in its SR.  A non-zero LSR value is required
        in order to calculate round trip delay. A value of 0 is
        permissible, however this field MUST be populated as soon as a
        delay estimate is available.

   end system delay: 16 bits
        The most recently estimated end system delay, expressed in
        milliseconds.  End system delay is defined as the total
        encoding, decoding and jitter buffer delay determined at the
        reporting endpoint.  This is the time required for an RTP frame
        to be buffered, decoded, converted to analog form, looped back
        at the local analog interface, encoded, and made available for
        transmission as an RTP frame.  The manner in which this value is
        estimated is implementation dependent.  This parameter MUST be
        provided in all VoIP metrics reports.

   Note that the one way symmetric VoIP segment delay may be calculated
   from the round trip and end system delays as follows.  If the round
   trip delay is denoted RTD and the end system delays associated with
   the two endpoints are ESD(A) and ESD(B) then:


   one way symmetric voice path delay  =3D  ( RTD + ESD(A) + ESD(B) ) / 2


4.7.4 Signal Related Metrics

   The following metrics are intended to provide real time information
   related to the non-packet elements of the voice over IP system to
   assist with the identification of problems affecting call quality.
   The values identified below must be determined for the received audio



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   signal. The information required to populate these fields may not be
   available in all systems, although it is strongly recommended that
   this data SHOULD be provided to support problem diagnosis.

   signal level: 8 bits
        The voice signal relative level is defined as the ratio of the
        signal level to a reference digital milliwatt, expressed in
        decibels as a signed integer in two's complement form.  This is
        measured only for packets containing speech energy.  The intent
        of this metric is not to provide a precise measurement of the
        signal level but to provide a real time indication that the
        signal level may be excessively high or low.

        signal level =3D 10 Log10 ( rms talkspurt power (mW) )

        A value of 127 indicates that this parameter is unavailable.
        Typical values should be generally in the -15 to -20 dBm range.

   noise level: 8 bits
        The noise level is defined as the ratio of the silent period
        background noise level to a reference digital milliwatt,
        expressed in decibels as a signed integer in two's complement
        form.

        noise level =3D 10 Log10 ( rms silence power (mW) )

        A value of 127 indicates that this parameter is unavailable.

   residual echo return loss (RERL): 8 bits
        The residual echo return loss is defined as the sum of the
        measured echo return loss (ERL) and the echo return loss
        enhancement (ERLE) expressed in dB as a signed integer in two's
        complement form.  It defines the ratio of a transmitted voice
        signal that is reflected back to the talker.  A low level of
        echo return loss (say less than 20 dB) in conjunction with some
        delay can lead to hollowness or audible echo.  A high level of
        echo return loss (say over 40 dB) is preferable.

   The ERL and ERLE parameters are often available directly from the
   echo canceler contained within the VoIP end system. They relate to
   the echo on the external network attached to the end point.

   In the case of a VoIP gateway this would be line echo that typically
   occurs at 2-4 wire conversion points in the network.  Echo return
   loss from typical 2-4 wire conversions can be in the 8-12 dB range.
   A line echo canceler can provide an ERLE of 30 dB or more and hence
   reduce this to 40-50 dB.  In the case of an IP phone this could be
   residual acoustic echo from speakerphone operation, and may more



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   correctly be termed terminal coupling loss (TCL).  A typical handset
   would result in 40-50 dB of echo due to acoustic feedback.

   Typical values for RERL are as follows:

   (i) IP gateway connected to circuit switched network with 2 wire loop
   Without echo cancellation, typical 2-4 wire converter ERL of 12 dB
   RERL =3D ERL + ERLE =3D 12 + 0 =3D 12 dB

   (ii) IP gateway connected to circuit switched network with 2 wire loop
   With echo canceler that improves echo by 30 dB
   RERL =3D ERL + ERLE =3D 12 + 30 =3D 42 dB

   (iii) IP phone with conventional handset
   Acoustic coupling from handset speaker to microphone 40 dB
   Residual echo return loss =3D TCL =3D 40 dB

   If we denote the "local" end of the VoIP path as A and the remote end
   as B and if the sender loudness rating (SLR) and receiver loudness
   rating (RLR) are known for A (default values 8 dB and 2 dB
   respectively), then the echo loudness level at end A (talker echo
   loudness rating or TELR) is given by:

   TELR(A) =3D SRL(A) + ERL(B) + ERLE(B) + RLR(A)

   TELR(B) =3D SRL(B) + ERL(A) + ERLE(A) + RLR(B)

   Hence in order to incorporate echo into a voice quality estimate at
   the A end of a VoIP connection it is desirable to send the ERL + ERLE
   value from B to A.

   For an IP phone with handset this metric MUST be set to the designed
   or measured terminal coupling loss, which would typically be 40-50
   dB.

   For a PC softphone or speakerphone this metric MUST be set to either
   the value reported by the acoustic echo canceler or to 127 to
   indicate an undefined value.

   For an IP gateway with ERL and ERLE measurements this metric MUST be
   reported as ERL + ERLE.

   For an IP gateway without ERL and ERLE measurement capability then
   this metric MUST be reported as 12 dB if line echo cancellation is
   disabled and 40 dB if line echo cancellation is enabled.

   Gmin
        See Configuration Parameters (Section 4.7.6, below).



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4.7.5 Call Quality or Transmission Quality Metrics

   The following metrics are direct measures of the call quality or
   transmission quality, and incorporate the effects of codec type,
   packet loss, discard, burstiness, delay etc.  These metrics may not
   be available in all systems however SHOULD be provided in order to
   support problem diagnosis.

   R factor: 8 bits
        The R factor is a voice quality metric describing the segment of
        the call that is carried over this RTP session.  It is expressed
        as an integer in the range 0 to 100, with a value of 94
        corresponding to "toll quality" and values of 50 or less
        regarded as unusable.  This metric is defined as including the
        effects of delay, consistent with ITU-T G.107 [6] and ETSI TS
        101 329-5 [3].

        A value of 127 indicates that this parameter is unavailable.

   ext. R factor: 8 bits
        The external R factor is a voice quality metric describing the
        segment of the call that is carried over a network segment
        external to the RTP segment, for example a cellular network. Its
        values are interpreted in the same manner as for the RTP R
        factor.  This metric is defined as including the effects of
        delay, consistent with ITU-T G.107 [6] and ETSI TS 101 329-5
        [3], and relates to the outward voice path from the Voice over
        IP termination for which this metrics block applies.

        A value of 127 indicates that this parameter is unavailable.

   Note that an overall R factor may be estimated from the RTP segment R
   factor and the external R factor, as follows:

   R total =3D RTP R factor + ext. R factor - 94

   MOS-LQ: 8 bits
        The estimated mean opinion score for listening quality (MOS-LQ)
        is a voice quality metric on a scale from 1 to 5, in which 5
        represents excellent and 1 represents unacceptable.  This metric
        is defined as not including the effects of delay and can be
        compared to MOS scores obtained from listening quality (ACR)
        tests. It is expressed as an integer in the range 10 to 50,
        corresponding to MOS x 10.  For example, a value of 35 would
        correspond to an estimated MOS score of 3.5.

        A value of 127 indicates that this parameter is unavailable.




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   MOS-CQ: 8 bits
        The estimated mean opinion score for conversational quality
        (MOS-CQ) is defined as including the effects of delay and other
        effects that would affect conversational quality.  The metric
        may be calculated by converting an R factor determined according
        to ITU-T G.107 [6] or ETSI TS 101 329-5 [3] into an estimated
        MOS using the equation specified in G.107.  It is expressed as
        an integer in the range 10 to 50, corresponding to MOS x 10, as
        for MOS-LQ.

        A value of 127 indicates that this parameter is unavailable.

4.7.6 Configuration Parameters

   Gmin: 8 bits
        The gap threshold.  This field contains the value used for this
        report block to determine if a gap exists.  The recommended
        value of 16 corresponds to a burst period having a minimum
        density of 6.25% of lost or discarded packets, which may cause
        noticeable degradation in call quality; during gap periods, if
        packet loss or discard occurs, each lost or discarded packet
        would be preceded by and followed by a sequence of at least 16
        received non-discarded packets.  Note that lost or discarded
        packets that occur within Gmin packets of a report being
        generated may be reclassified as being part of a burst or gap in
        later reports.  ETSI TS 101 329-5 [3] defines a computationally
        efficient algorithm for measuring burst and gap density using a
        packet loss/discard event driven approach.  This algorithm is
        reproduced in Appendix A.2 of the present document.  Gmin MUST
        not be zero and MUST be provided.

   receiver configuration byte (RX config): 8 bits
        This byte consists of the following fields:


         0 1 2 3 4 5 6 7
        +-+-+-+-+-+-+-+-+
        |PLC|JBA|JB rate|
        +-+-+-+-+-+-+-+-+


        packet loss concealment (PLC): 2 bits
             Standard (11) / enhanced (10) / disabled (01) / unspecified
             (00).  When PLC =3D 11 then a simple replay or interpolation
             algorithm is being used to fill-in the missing packet.
             This is typically able to conceal isolated lost packets
             with loss rates under 3%.  When PLC =3D 10 then an enhanced
             interpolation algorithm is being used.  This would



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             typically be able to conceal lost packets for loss rates of
             10% or more.  When PLC =3D 01 then silence is inserted in
             place of lost packets.  When PLC =3D 00 then no information
             is available concerning the use of PLC however for some
             codecs this may be inferred.

        jitter buffer adaptive (JBA): 2 bits
             Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown
             (00).  When the jitter buffer is adaptive then its size is
             being dynamically adjusted to deal with varying levels of
             jitter.  When non-adaptive, the jitter buffer size is
             maintained at a fixed level.  When either adaptive or non-
             adaptive modes are specified then the jitter buffer size
             parameters below MUST be specified.

        jitter buffer rate (JB rate): 4 bits
             J =3D adjustment rate (0-15). This represents the
             implementation specific adjustment rate of a jitter buffer
             in adaptive mode. This parameter is defined in terms of the
             approximate time taken to fully adjust to a step change in
             peak to peak jitter from 30 ms to 100 ms such that:

             adjustment time =3D 2 * J * frame size (ms)

             This parameter is intended only to provide a guide to the
             degree of "aggressiveness" of a an adaptive jitter buffer
             and may be estimated. A value of 0 indicates that the
             adjustment time is unknown for this implementation.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

4.7.7 Jitter Buffer Parameters

   jitter buffer nominal delay (JB nominal): 16 bits
        This is the current nominal jitter buffer delay in milliseconds,
        which corresponds to the nominal jitter buffer delay for packets
        that arrive exactly on time.  This parameter MUST be provided
        for both fixed and adaptive jitter buffer implementations.

   jitter buffer maximum delay (JB maximum): 16 bits
        This is the current maximum jitter buffer delay corresponding to
        the earliest arriving packet that would not be discarded.  In
        simple queue implementations this may correspond to the nominal
        size. In adaptive jitter buffer implementations this value may
        dynamically vary up to JB abs max (see below).  This parameter



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        MUST be provided for both fixed and adaptive jitter buffer
        implementations.

   jitter buffer absolute maximum delay (JB abs max): 16 bits
        This is the absolute maximum delay that the adaptive jitter
        buffer can reach under worst case jitter conditions.  This
        parameter MUST be provided for adaptive jitter buffer
        implementations and its value MUST be set to JB maximum for
        fixed jitter buffer implementations.

5. SDP Signaling

   This section defines Session Description Protocol (SDP) [4] signaling
   for XR blocks that can be employed by applications that utilize SDP.
   This signaling is defined to be used either by applications that
   implement the SDP Offer/Answer model [9] or by applications that use
   SDP to describe media and transport configurations in connection with
   such protocols as the Session Announcement Protocol (SAP) [15] or the
   Real Time Streaming Protocol (RTSP) [16].  There exist other
   potential signaling methods, which are not defined here.

   The XR blocks MAY be used without prior signaling.  This is
   consistent with the rules governing other RTCP packet types, as
   described in [10].  An example in which signaling would not be used
   is an application that always requires the use of one or more XR
   blocks.  However for applications that are configured at session
   initiation, the use of some type of signaling is recommended.

   Note that, although the use of SDP signaling for XR blocks may be
   optional, if used it MUST be used as defined here.  If SDP signaling
   is used in an environment where XR blocks are only implemented by
   some fraction of the participants, the ones not implementing the XR
   blocks will ignore the SDP attribute.

5.1 The SDP Attribute

   This section defines one new SDP attribute "rtcp-xr" that can be used
   to signal participants in a media session that they should use the
   specified XR blocks.  This attribute can be easily extended in the
   future with new parameters to cover any new report blocks.

   The RTCP XR blocks SDP attribute is defined below in Augmented
   Backus-Naur Form (ABNF) [2].  It is both a session and a media level
   attribute.  When specified at session level, it applies to all media
   level blocks in the session.  Any media level specification MUST
   replace a session level specification, if one is present, for that
   media block.




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   rtcp-xr-attrib =3D "a=3D" "rtcp-xr" ":" [xr-format *(SP xr-format)] CR=
LF


   xr-format =3D pkt-loss-rle
             / pkt-dup-rle
             / pkt-rcpt-times
             / rcvr-rtt
             / stat-summary
             / voip-metrics
             / format-ext


   pkt-loss-rle   =3D "pkt-loss-rle" ["=3D" max-size]
   pkt-dup-rle    =3D "pkt-dup-rle" ["=3D" max-size]
   pkt-rcpt-times =3D "pkt-rcpt-times" ["=3D" max-size]
   rcvr-rtt       =3D "rcvr-rtt" "=3D" rcvr-rtt-mode [":" max-size]
   stat-summary   =3D "stat-summary"
   voip-metrics   =3D "voip-metrics"
   max-size       =3D 1*DIGIT ; maximum block size in octets
   rcvr-rtt-mode  =3D "all"
                  / "sender"
   format-ext     =3D non-ws-string

   non-ws-string  =3D 1*(%x21-FF)

   The "rtcp-xr" attribute contains zero, one, or more XR block related
   parameters.  Each parameter signals functionality for an XR block, or
   a group of XR blocks.  The attribute is extensible so that parameters
   can be defined for any future XR block (and a parameter should be
   defined for every future block).

   Each "rtcp-xr" parameter belongs to one of two categories.  The first
   category, the unilateral parameters, are for report blocks that
   simply report on the RTP stream and related metrics.  The second
   category, collaborative parameters, are for XR blocks that involve
   actions by more than one party in order to carry out their functions.

   Round trip time (RTT) measurement is an example of collaborative
   functionality.  An RTP data packet receiver sends a Receiver
   Reference Time Report Block (Section 4.4).  A participant that
   receives this block sends a DLRR Report Block (Section 4.5) in
   response, allowing the receiver to calculate its RTT to that
   participant.  As this example illustrates, collaborative
   functionality may be implemented by two or more different XR blocks.
   The collaborative functionality of several XR blocks may be governed
   by a single "rtcp-xr" parameter.

   For the unilateral category, this document defines the following



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   parameters.  The parameter names and their corresponding XR formats
   are as follows:

   Parameter name    XR block (block type and name)
   --------------    ------------------------------------
   pkt-loss-rle      1  Loss RLE Report Block
   pkt-dup-rle       2  Duplicate RLE Report Block
   pkt-rcpt-times    3  Packet Receipt Times Report Block
   stat-summary      6  Statistics Summary Report Block
   voip-metrics      7  VoIP Metrics Report Block


   The "pkt-loss-rle", "pkt-dup-rle", and "pkt-rcpt-times" parameters
   MAY specify an integer value.  This value indicates the largest size
   the whole report block SHOULD have in octets.  This shall be seen as
   an indication that thinning shall be applied if necessary to meet the
   target size.

   Blocks in the collaborative category are classified as initiator
   blocks or response blocks.  Signaling SHOULD indicate which
   participants are required to respond to the initiator block.  A party
   that wishes to receive response blocks from those participants can
   trigger this by sending an initiator block.

   The collaborative category currently consists only of one
   functionality, namely the RTT measurement mechanism for RTP data
   receivers.  The collective functionality of the Receiver Reference
   Time Report Block and DLRR Report Block is represented by the "rcvr-
   rtt" parameter.  This parameter takes as its arguments a mode value
   and, optionally, a maximum size for the DLRR report block.  The mode
   value "all" indicates that both RTP data senders and data receivers
   MAY send DLRR blocks, while the mode value "sender" indicates that
   only active RTP senders MAY send DLRR blocks, i.e. non RTP senders
   SHALL NOT send DLRR blocks.  If a maximum size in octets is included,
   any DLRR Report Blocks that are sent SHALL NOT exceed the specified
   size.  If size limitations mean that a DLRR Report Block sender
   cannot report in one block upon all participants from which it has
   received a Receiver Reference Time Report Block then it SHOULD report
   on participants in a round robin fashion across several report
   intervals.

   The "rtcp-xr" attributes parameter list MAY be empty.  This is useful
   in cases in which an application needs to signal that it understands
   the SDP signaling but does not wish to avail itself of XR
   functionality.  For example, an application in a SIP controlled
   session could signal that it wishes to stop using all XR blocks by
   removing all applicable SDP parameters in a re-INVITE message that it
   sends.  If XR blocks are not to be used at all from the beginning of



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   a session, it is RECOMMENDED that the "rtcp-xr" attribute not be
   supplied at all.

   When the "rtcp-xr" attribute is present, participants SHOULD NOT send
   XR blocks other than the ones indicated by the parameters.  This
   means that inclusion of a "rtcp-xr" attribute without any parameters
   tells a participant that it SHOULD NOT send any XR blocks at all.
   The purpose is to conserve bandwidth.  This is especially important
   when collaborative parameters are applied to a large multicast group:
   the sending of an initiator block could potentially trigger responses
   from all participants.  There are, however, contexts in which it
   makes sense to send an XR block in the absence of a parameter
   signaling its use.  For instance, an application might be designed so
   as to send certain report blocks without negotiation, while using SDP
   signaling to negotiate the use of other blocks.

5.2 Usage in Offer/Answer

   In the Offer/Answer context [9], the interpretation of SDP signaling
   for XR packets depends upon the direction attribute that is signaled:
   "recvonly", "sendrecv", or "sendonly" [4].  If no direction attribute
   is supplied then "sendrecv" is assumed.  This section applies only to
   unicast media streams, except where noted.  Discussion of unilateral
   parameters is followed by discussion of collaborative parameters in
   this section.

   For "sendonly" and "sendrecv" media stream offers that specify
   unilateral "rtcp-xr" attribute parameters, the answerer SHOULD send
   the corresponding XR blocks.  For "sendrecv" offers, the answerer MAY
   include the "rtcp-xr" attribute in its response, and specify any
   unilateral parameters in order to request that the offerer send the
   corresponding XR blocks.  The offerer SHOULD send these blocks.

   For "recvonly" media stream offers, the offerer's use of the "rtcp-
   xr" attribute in connection with unilateral parameters indicates that
   the offerer is capable of sending the corresponding XR blocks.  If
   the answerer responds with an "rtcp-xr" attribute, the offerer SHOULD
   send XR blocks for each specified unilateral parameter that was in
   its offer.

   For multicast media streams, the inclusion of an "rtcp-xr" attribute
   with unilateral parameters means that every media recipient SHOULD
   send the corresponding XR blocks.

   An SDP offer with a collaborative parameter declares the offerer
   capable of receiving the corresponding initiator and replying with
   the appropriate responses.  For example, an offer that specifies the
   "rcvr-rtt" parameter means that the offerer is prepared to receive



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   Receiver Reference Time Report Blocks and to send DLRR Report Blocks.
   An offer of a collaborative parameter means that the answerer MAY
   send the initiator, and, having received the initiator, the offerer
   SHOULD send the responses.

   There are exceptions to the rule that an offerer of a collaborative
   parameter should send responses.  For instance, the collaborative
   parameter might specify a mode that excludes the offerer.  Or
   congestion control or maximum transmission unit considerations might
   militate against the offerer's response.

   By including a collaborative parameter in its answer, the answerer
   declares its ability to receive initiators and to send responses.
   The offerer MAY then send initiators, to which the answerer SHOULD
   reply with responses.  As for the offer of a collaborative parameter,
   there are exceptions to the rule that the answerer should reply.

   When making an SDP offer of a collaborative parameter for a multicast
   media stream, the offerer SHOULD specify which participants are to
   respond to a received initiator.  A participant that is not specified
   SHOULD NOT send responses.  Otherwise, undue bandwidth might be
   consumed.  The offer indicates that each participant that is
   specified SHOULD respond if it receives an initiator.  It also
   indicates that a specified participant MAY send an initiator block.

   An SDP answer for a multicast media stream SHOULD include all
   collaborative parameters that are present in the offer and that are
   supported by the answerer.  It SHOULD NOT include any collaborative
   parameter that is absent from the offer.

   If a participant receives an SDP offer and understands the "rtcp-xr"
   attribute but does not wish to implement XR functionality offered,
   its answer SHOULD include an "rtcp-xr" attribute without parameters.
   By doing so, the party declares that at a minimum that it is capable
   of understanding the signaling.

5.3 Usage Outside of Offer/Answer

   SDP can be employed outside of the Offer/Answer context, for instance
   for multimedia sessions that are announced through the Session
   Announcement Protocol (SAP) [15], or streamed through the Real Time
   Streaming Protocol (RTSP) [16].  The signaling model is simpler, as
   the sender does not negotiate parameters, but the functionality that
   is expected from specifying the "rtcp-xr" attribute is the same as in
   Offer/Answer.

   When a unilateral parameter is specified for the "rtcp-xr" attribute
   associated with a media stream, the receiver of that stream SHOULD



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   send the corresponding XR block.  When a collaborative parameter is
   specified, only the participants indicated by the mode value in the
   collaborative parameter are concerned.  Each such participant that
   receives an initiator block SHOULD send the corresponding response
   block.  Each such participant MAY also send initiator blocks.

6. IANA Considerations

   This document defines a new RTCP packet type, the Extended Report
   (XR) type, within the existing Internet Assigned Numbers Authority
   (IANA) registry of RTP RTCP Control Packet Types.  This document also
   defines a new IANA registry: the registry of RTCP XR Block Types.
   Within this new registry, this document defines an initial set of
   seven block types and describes how the remaining types are to be
   allocated.

   Further, this document defines a new SDP attribute, "rtcp-xr", within
   the existing IANA registry of SDP Parameters.  It defines a new IANA
   registry, the registry of RTCP XR SDP Parameters, and an initial set
   of six parameters, and describes how additional parameters are to be
   allocated.

6.1 XR Packet Type

   The XR packet type defined by this document is registered with the
   IANA as packet type 207 in the registry of RTP RTCP Control Packet
   types (PT).

6.2 RTCP XR Block Type Registry

   This document creates an IANA registry called the RTCP XR Block Type
   Registry to cover the name space of the Extended Report block type
   (BT) field specified in Section 3. The BT field contains eight bits,
   allowing 256 values.  The RTCP XR Block Type Registry is to be
   managed by the IANA according to the Specification Required policy of
   RFC 2434 [8].  Future specifications SHOULD attribute block type
   values in strict numeric order following the values attributed in
   this document:













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   BT  name
   --  ----
    1  Loss RLE Report Block
    2  Duplicate RLE Report Block
    3  Packet Receipt Times Report Block
    4  Receiver Reference Time Report Block
    5  DLRR Report Block
    6  Statistics Summary Report Block
    7  VoIP Metrics Report Block


   The BT value 255 is reserved for future extensions.

   Furthermore, future specifications SHOULD avoid the value 0.  Doing
   so facilitates packet validity checking, since an all-zeros field
   might commonly be found in an ill-formed packet.

6.3 The "rtcp-xr" SDP Attribute

   This SDP attribute "rtcp-xr" defined by this document is registered
   with the IANA registry of SDP Parameters as follows:

   SDP Attribute ("att-field"):

     Attribute name:     rtcp-xr
     Long form:          RTP Control Protocol Extended Report Parameters
     Type of name:       att-field
     Type of attribute:  session and media level
     Subject to charset: no
     Purpose:            see Section 5 of this document
     Reference:          this document
     Values:             see this document and registrations below

   The attribute has an extensible parameter field and therefore a
   registry for these parameters is required.  This document creates an
   IANA registry called the RTCP XR SDP Parameters Registry.  It
   contains the six parameters defined in Section 5.1: "pkt-loss-rle",
   "pkt-dup-rle", "pkt-rcpt-times", "stat-summary", "voip-metrics", and
   "recv-rtt".

   Additional parameters are to be added to this registry in accordance
   with the Specification Required policy of RFC 2434 [8].  Any
   registration MUST contain the following information:

   - Contact information of the one doing the registration, including at
     least name, address, and email.

   - An Augmented Backus-Naur Form (ABNF) [2] definition of the



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     parameter, in accordance with the "format-ext" definition.

   - A description of what the parameter represents and how it shall be
     interpreted, both normally and in Offer/Answer.

7. Security Considerations

   This document extends the RTCP reporting mechanism.  The security
   considerations that apply to RTCP reports also apply to XR reports.
   This section details the additional security considerations that
   apply to the extensions.

   The extensions introduce heightened confidentiality concerns.
   Standard RTCP reports contain a limited number of summary statistics.
   The information contained in XR reports is both more detailed and
   more extensive (covering a larger number of parameters).  The per-
   packet report blocks and the VoIP Metrics Report Block provide
   examples.

   The per-packet information contained in Loss RLE, Duplicate RLE, and
   Packet Receipt Times Report Blocks facilitates multicast inference of
   network characteristics (MINC) [11].  Such inference can reveal the
   gross topology of a multicast distribution tree, as well as
   parameters, such as the loss rates and delays, along paths between
   branching points in that tree.  Such information might be considered
   sensitive to autonomous system administrators.

   The VoIP Metrics Report Block provides information on the quality of
   ongoing voice calls.  Though such information might be carried in
   application specific format in standard RTP sessions, making it
   available in a standard format here makes it more available to
   potential eavesdroppers.

   No new mechanisms are introduced in this document to ensure
   confidentiality.  Encryption procedures, such as those being
   suggested for a Secure RTCP (SRTCP) [12] at the time that this
   document was written, can be used when confidentiality is a concern
   to end hosts.  Given that RTCP traffic can be encrypted by the end
   hosts, autonomous systems must be prepared for the fact that certain
   aspects of their network topology can be revealed.

   Any encryption or filtering of XR report blocks entails a loss of
   monitoring information to third parties.  For example, a network that
   establishes a tunnel to encrypt VoIP Report Blocks denies that
   information to the service providers traversed by the tunnel.  The
   service providers cannot then monitor or respond to the quality of
   the VoIP calls that they carry, potentially creating problems for the
   network's users.  As a default, XR packets SHOULD NOT be encrypted or



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   filtered.

   The extensions also make certain denial of service attacks easier.
   This is because of the potential to create RTCP packets much larger
   than average with the per packet reporting capabilities of the Loss
   RLE, Duplicate RLE, and Timestamp Report Blocks.  Because of the
   automatic bandwidth adjustment mechanisms in RTCP, if some session
   participants are sending large RTCP packets, all participants will
   see their RTCP reporting intervals lengthened, meaning they will be
   able to report less frequently.  To limit the effects of large
   packets, even in the absence of denial of service attacks,
   applications SHOULD place an upper limit on the size of the XR report
   blocks they employ.  The "thinning" techniques described in Section
   4.1 permit the packet-by-packet report blocks to adhere to a
   predefined size limit.

A. Algorithms

A.1 Sequence Number Interpretation

   This the algorithm suggested by Section 4.1 for keeping track of the
   sequence numbers from a given sender.  It implements the accounting
   practice required for the generation of Loss RLE Report Blocks.

   This algorithm keeps track of 16 bit sequence numbers by translating
   them into a 32 bit sequence number space.  The first packet received
   from a source is considered to have arrived roughly in the middle of
   that space.  Each packet that follows is placed either ahead or
   behind the prior one in this 32 bit space, depending upon which
   choice would place it closer (or, in the event of a tie, which choice
   would not require a rollover in the 16 bit sequence number).

   // The reference sequence number is an extended sequence number
   // that serves as the basis for determining whether a new 16 bit
   // sequence number comes earlier or later in the 32 bit sequence
   // space.
   u_int32 _src_ref_seq;
   bool    _uninitialized_src_ref_seq;

   // Place seq into a 32-bit sequence number space based upon a
   // heuristic for its most likely location.
   u_int32 extend_seq(const u_int16 seq) {

           u_int32 extended_seq, seq_a, seq_b, diff_a, diff_b;
           if(_uninitialized_src_ref_seq) {

                   // This is the first sequence number received.  Place
                   // it in the middle of the extended sequence number



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                   // space.
                   _src_ref_seq                =3D seq | 0x80000000u;
                   _uninitialized_src_ref_seq  =3D false;
                   extended_seq                =3D _src_ref_seq;
           }
           else {

                   // Prior sequence numbers have been received.
                   // Propose two candidates for the extended sequence
                   // number: seq_a is without wraparound, seq_b with
                   // wraparound.
                   seq_a =3D seq | (_src_ref_seq & 0xFFFF0000u);
                   if(_src_ref_seq < seq_a) {
                           seq_b  =3D seq_a - 0x00010000u;
                           diff_a =3D seq_a - _src_ref_seq;
                           diff_b =3D _src_ref_seq - seq_b;
                   }
                   else {
                           seq_b  =3D seq_a + 0x00010000u;
                           diff_a =3D _src_ref_seq - seq_a;
                           diff_b =3D seq_b - _src_ref_seq;
                   }

                   // Choose the closer candidate.  If they are equally
                   // close, the choice is somewhat arbitrary: we choose
                   // the candidate for which no rollover is necessary.
                   if(diff_a < diff_b) {
                           extended_seq =3D seq_a;
                   }
                   else {
                           extended_seq =3D seq_b;
                   }

                   // Set the reference sequence number to be this most
                   // recently-received sequence number.
                   _src_ref_seq =3D extended_seq;
           }

           // Return our best guess for a 32-bit sequence number that
           // corresponds to the 16-bit number we were given.
           return extended_seq;
   }

   A.2 Example Burst Packet Loss Calculation.

   This is an algorithm for measuring the burst characteristics for the
   VoIP Metrics Report Block (Section 4.7).  It is reproduced from ETSI
   TS 101 329-5 [3].  The algorithm is event driven and hence extremely



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   computationally efficient.

   Given the following definition of states:


   state 1 =3D received a packet during a gap
   state 2 =3D received a packet during a burst
   state 3 =3D lost a packet during a burst
   state 4 =3D lost an isolated packet during a gap


   The "c" variables below correspond to state transition counts, i.e.
   c14 is the transition from state 1 to state 4. It is possible to
   infer one of a pair of state transition counts to an accuracy of 1
   which is generally sufficient for this application.

   "pkt" is the count of packets received since the last packet was
   declared lost or discarded and "lost" is the number of packets lost
   within the current burst.  "packet_lost" and "packet_discarded" are
   Boolean variables that indicate if the event that resulted in this
   function being invoked was a lost or discarded packet.

   if(packet_lost) {
           loss_count++;
   }
   if(packet_discarded) {
           discard_count++;
   }
   if(!packet_lost && !packet_discarded) {
           pkt++;
   }
   else {
           if(pkt >=3D gmin) {
                   if(lost =3D=3D 1) {
                           c14++;
                   }
                   else {
                           c13++;
                   }
                   lost =3D 1;
                   c11 +=3D pkt;
           }
           else {
                   lost++;
                   if(pkt =3D=3D 0) {
                           c33++;
                   }
                   else {



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Internet-Draft                   RTCP XR                      April 2003


                           c23++;
                           c22 +=3D (pkt - 1);
                   }
           }
           pkt =3D 0;
   }

   At each reporting interval the burst and gap metrics can be
   calculated as follows.

   // Calculate additional transition counts.
   c31 =3D c13;
   c32 =3D c23;
   ctotal =3D c11 + c14 + c13 + c22 + c23 + c31 + c32 + c33;

   // Calculate burst and densities.
   p32 =3D c32 / (c31 + c32 + c33);
   if((c22 + c23) < 1) {
           p23 =3D 1;
   }
   else {
           p23 =3D 1 - c22/(c22 + c23);
   }
   burst_density =3D 256 * p23 / (p23 + p32);
   gap_density =3D 256 * c14 / (c11 + c14);

   // Calculate burst and gap durations in ms
   m =3D frameDuration_in_ms * framesPerRTPPkt;
   gap_length =3D (c11 + c14 + c13) * m / c13;
   burst_length =3D ctotal * m / c13 - lgap;

   /* calculate loss and discard densities */
   loss_density =3D 256 * loss_count / ctotal;
   discard_density =3D 256 * discard_count / ctotal;

Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   intellectual property or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; neither does it represent that it
   has made any effort to identify any such rights.  Information on the
   IETF's procedures with respect to rights in standards-track and
   standards-related documentation can be found in BCP 11 [5].  Copies
   of claims of rights made available for publication and any assurances
   of licenses to be made available, or the result of an attempt made to
   obtain a general license or permission for the use of such



Friedman et al.              Standards Track                   [Page 47]
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Internet-Draft                   RTCP XR                      April 2003


   proprietary rights by implementors or users of this specification can
   be obtained from the IETF Secretariat.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights which may cover technology that may be required to practice
   this standard.  Please address the information to the IETF Executive
   Director.

Full Copyright Statement

   Copyright (C) The Internet Society (2003). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgments

   We thank the following people: Colin Perkins, Steve Casner, and
   Henning Schulzrinne for their considered guidance; Sue Moon for
   helping foster collaboration between the authors; Mounir Benzaid for
   drawing our attention to the reporting needs of MLDA; Dorgham Sisalem
   and Adam Wolisz for encouraging us to incorporate MLDA block types;
   and Jose Rey for valuable review of the SDP Signaling section.

Contributors




Friedman et al.              Standards Track                   [Page 48]
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Internet-Draft                   RTCP XR                      April 2003


   The following people are the authors of this document:

   Kevin Almeroth, UCSB
   Ramon Caceres, ShieldIP
   Alan Clark, Telchemy
   Robert Cole, AT&T Labs
   Nick Duffield, AT&T Labs-Research
   Timur Friedman, Paris 6
   Kaynam Hedayat, Brix Networks
   Kamil Sarac, UT Dallas
   Magnus Westerlund, Ericsson

   The principal people to contact regarding the individual report
   blocks described in this document are as follows:


   sec. report block                          principal contributors
   ---- ------------                          ----------------------
   4.1  Loss RLE Report Block                 Friedman, Caceres, Duffield
   4.2  Duplicate RLE Report Block            Friedman, Caceres, Duffield
   4.3  Packet Receipt Times Report Block     Friedman, Caceres, Duffield
   4.4  Receiver Reference Time Report Block  Friedman
   4.5  DLRR Report Block                     Friedman
   4.6  Statistics Summary Report Block       Almeroth, Sarac
   4.7  VoIP Metrics Report Block             Clark, Cole, Hedayat


   The principal person to contact regarding the SDP signaling described
   in this document is Magnus Westerlund.

Authors' Addresses

      Kevin Almeroth <almeroth@cs.ucsb.edu>
      Department of Computer Science
      University of California
      Santa Barbara, CA 93106 USA

      Ramon Caceres <ramon@shieldip.com>
      ShieldIP, Inc.
      11 West 42nd Street, 31st Floor
      New York, NY 10036 USA










Friedman et al.              Standards Track                   [Page 49]
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Internet-Draft                   RTCP XR                      April 2003


      Alan Clark <alan@telchemy.com>
      Telchemy Incorporated
      3360 Martins Farm Road, Suite 200
      Suwanee, GA 30024 USA
      Tel: +1 770 614 6944
      Fax: +1 770 614 3951

      Robert Cole <rgcole@att.com>
      AT&T Labs
      330 Saint Johns Street,
      2nd Floor
      Havre de Grace, MD 21078 USA
      Tel: +1 410 939 8732
      Fax: +1 410 939 8732

      Nick Duffield <duffield@research.att.com>
      AT&T Labs-Research
      180 Park Avenue, P.O. Box 971
      Florham Park, NJ 07932-0971 USA
      Tel: +1 973 360 8726
      Fax: +1 973 360 8050

      Timur Friedman <timur.friedman@lip6.fr>
      Universite Pierre et Marie Curie (Paris 6)
      Laboratoire LiP6-CNRS
      8, rue du Capitaine Scott
      75015 PARIS France
      Tel: +33 1 44 27 71 06
      Fax: +33 1 44 27 74 95

      Kaynam Hedayat <khedayat@brixnet.com>
      Brix Networks
      285 Mill Road
      Chelmsford, MA 01824 USA
      Tel: +1 978 367 5600
      Fax: +1 978 367 5700

      Kamil Sarac <ksarac@utdallas.edu>
      Department of Computer Science (ES 4.207)
      Eric Jonsson School of Engineering & Computer Science
      University of Texas at Dallas
      Richardson, TX 75083-0688 USA
      Tel: +1 972 883 2337
      Fax: +1 972 883 2349







Friedman et al.              Standards Track                   [Page 50]
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Internet-Draft                   RTCP XR                      April 2003


      Magnus Westerlund <magnus.westerlund@era.ericsson.se>
      Ericsson Research, Corporate Unit
      Ericsson Radio Systems AB
      SE-164 80 Stockholm
      Sweden
      Tel: +46 8 404 82 87
      Fax: +46 8 757 55 50

References

   The references are divided into normative references and non-
   normative references.

Normative References

   [1] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," BCP 14, RFC 2119, IETF, March 1997.

   [2] D. Crocker, P. Overell, "Augmented BNF for Syntax Specifications:
   ABNF", RFC 2234, Internet Engineering Task Force, November 1997.

   [3] ETSI, "Quality of Service (QoS) measurement methodologies," ETSI
   TS 101 329-5 V1.1.1 (2000-11), November 2000.

   [4] M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC
   2327, Internet Engineering Task Force, April 1998.

   [5] R. Hovey and S. Bradner, "The Organizations Involved in the IETF
   Standards Process," best current practice BCP 11, RFC 2028, IETF,
   October 1996.

   [6] ITU-T, "The E-Model, a computational model for use in
   transmission planning," Recommendation G.107 (05/00), May 2000.

   [7] J. Reynolds and J. Postel, "Assigned Numbers," standard STD 2,
   RFC 1700, IETF, October 1994.

   [8] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
   Considerations Section in RFCs," best current practice BCP 26, RFC
   2434, IETF, October 1998.

   [9] J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
   Session Description Protocol (SDP)", RFC 3264, Internet Engineering
   Task Force, June 2002.

   [10] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   A transport protocol for real-time applications," RFC 1889, IETF,
   February 1996.



Friedman et al.              Standards Track                   [Page 51]
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Internet-Draft                   RTCP XR                      April 2003


Non-Normative References

   [11] A. Adams, T. Bu, R. Caceres, N. G. Duffield, T. Friedman, J.
   Horowitz, F. Lo Presti, S. B. Moon, V. Paxson, and D. Towsley, "The
   Use of End-to-End Multicast Measurements for Characterizing Internal
   Network Behavior," IEEE Communications Magazine, May 2000.

   [12] Baugher, McGrew, Oran, Blom, Carrara, Naslund, and Norrman, "The
   Secure Real-time Transport Protocol," Internet-Draft draft-ietf-avt-
   srtp-05.txt, June 2002.  Note: this is is a work in progress.

   [13] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
   measurement infrastructures using RTP," Proc. IEEE Infocom 2002.

   [14] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
   Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
   2001.

   [15] M. Handley, C. Perkins, E. Whelan, "Session Announcement
   Protocol", RFC 2974, Internet Engineering Task Force, October 2000.

   [16] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [17] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
   Control Framework for Heterogeneous Multicast Environments", Proc.
   IWQoS 2000.























Friedman et al.              Standards Track                   [Page 52]
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--------------Boundary-00=_0FWG70E3FSRIR5YRS3YC--

_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Does anybody have experience with using FireWire for real-time video, 
e.g., to feed video into vic on Windows from a digital video camera?

Thanks.

Henning

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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Thu, 17 Apr 2003 16:04:08 -0700 (PDT)
From: Greg Herlein <gherlein@herlein.com>
To: Henning Schulzrinne <hgs@cs.columbia.edu>
cc: avt@ietf.org
Subject: Re: [AVT] Firewire for real-time video
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> Does anybody have experience with using FireWire for real-time video, 
> e.g., to feed video into vic on Windows from a digital video camera?

I looked at this last year and found that Firewire-capable
cameras that could be controlled (ie, focus.aperture, etc) were
*really* expensive (multiple thousands of dollars) and that the
average consumer DV cam could not support anything other than
playback features.  If that has changed, I'd like to know about
it too!

There are several libraries on the linux side for pulling
Firewire streams right off the camera in playback mode... there
must be some for Win32.

To ask a related question, does anyone know if there is a tape
drive for miniDV tapes that can be used directly in a PC (ideally
PCI bus or USB interface?).  Once a tape is created, it's crazy
to have to use the camera to play it so it can be ripped onto
spinning storage.  

Greg



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Date: Fri, 18 Apr 2003 11:29:49 +0900
Subject: Re: [AVT] Firewire for real-time video
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See:

http://www.sfc.wide.ad.jp/DVTS/

If you can accept Japanese page,

http://www.fatware.co.jp/

On 2003.Apr.18, at 07:12  AM, Henning Schulzrinne wrote:

> Does anybody have experience with using FireWire for real-time video, 
> e.g., to feed video into vic on Windows from a digital video camera?
>
> Thanks.
>
> Henning
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
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https://www1.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 Control Protocol Extended Reports (RTCP XR)
	Author(s)	: T. Friedman et al.
	Filename	: draft-ietf-avt-rtcp-report-extns-04.txt
	Pages		: 52
	Date		: 2003-4-17
	
This document defines the Extended Report (XR) packet type for the
RTP Control Protocol (RTCP), and defines how the use of XR packets
can be signaled by an application if it employs the Session
Description Protocol (SDP).  XR packets are composed of report
blocks, and seven block types are defined here.  The purpose of the
extended reporting format is to convey information that supplements
the six statistics that are contained in the report blocks used by
RTCP's Sender Report (SR) and Receiver Report (RR) packets.  Some
applications, such as multicast inference of network characteristics
(MINC) or voice over IP (VoIP) monitoring, require other and more
detailed statistics.  In addition to the block types defined here,
additional block types may be defined in the future by adhering to
the framework that this document provides.

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_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Fri Apr 18 09:36:33 2003
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Subject: RE: [AVT] Firewire for real-time video
Date: Fri, 18 Apr 2003 09:34:42 -0400
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Thread-Topic: [AVT] Firewire for real-time video
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Hi Henning,

I did some experiments some years ago (when Firewire 400 was still experimental). At that time, I used the card's SDK directly, which allowed for receiving the synchronous data directly in a block of memory. 

VFW-compatible integration (which is probably what you need if you want to feed it directly into vic without changing the vic code) was not possible at this time. I don't know whether or not this has changed though. However, opening a small window in Windows that displays a VFW-compatible source, is quite easy to do. If your Firewire card allows for VFW integration, it should display the video right away.

Regards,

Dirk

>-----Original Message-----
>From: ext Henning Schulzrinne [mailto:hgs@cs.columbia.edu]
>Sent: Thursday, April 17, 2003 6:12 PM
>To: avt@ietf.org
>Subject: [AVT] Firewire for real-time video
>
>
>Does anybody have experience with using FireWire for real-time video, 
>e.g., to feed video into vic on Windows from a digital video camera?
>
>Thanks.
>
>Henning
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
>
_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Fri Apr 18 10:14:52 2003
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Colin, Steve

The new (and hopefully) final RTCP XR draft has emerged, complete with SDP
signaling, I would like to propose that this be considered for last call.

Regards

Alan Clark
Telchemy


_______________________________________________
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From mailnull@www1.ietf.org  Fri Apr 18 19:47:21 2003
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Subject: [AVT] WG Last call: draft-ietf-avt-rtcp-report-extns-04.txt
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This is to announce a working group last call on the RTP Control Protocol
Extended Reports (draft-ietf-avt-rtcp-report-extns-04.txt). Please send any
final comments on this draft to the mailing list by Monday 28th April 2003.
At that time, we will consider if it is appropriate to request publication
of this draft as a proposed standard RFC.

I have a number of concerns regarding the clarity of specification of the
Voice-over-IP metrics, which I will send as last call comments in the next
couple of days. 

-- 
Colin Perkins
http://csperkins.org/
_______________________________________________
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From mailnull@www1.ietf.org  Mon Apr 21 19:04:16 2003
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Hello,

 

I was wondering if this draft is still being considered as an action item
for the Working Group. We are in need of an assignment for the "Clear Mode"
(bit exact data with no encoding) codec to be used in our products.

 

I could not find this draft in the IETF draft index and in the AVT WG home
page as well.

 

Please let me know the status of the draft.

 

Thanks,

Bala

______________________

Bala Pitchandi

UTStarcom Inc.


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<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
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<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
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<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>I was wondering if this draft is still being =
considered as
an action item for the Working Group. We are in need of an assignment =
for the &#8220;Clear
Mode&#8221; (bit exact data with no encoding) codec to be used in our =
products.</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>I could not find this draft in the IETF draft index =
and in
the AVT WG home page as well.</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>&nbsp;</span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DVerdana><span =
style=3D'font-size:9.0pt;
font-family:Verdana'>Please let me know the status of the =
draft.</span></font></p>

<p class=3DMsoAutoSig><font size=3D2 face=3DTahoma><span =
style=3D'font-size:11.0pt;
font-family:Tahoma'>&nbsp;</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
font-family:Verdana'>Thanks,</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
font-family:Verdana'>Bala</span></font></p>

<p class=3DMsoAutoSig><font size=3D1 face=3DVerdana><span =
style=3D'font-size:7.5pt;
font-family:Verdana'>______________________</span></font></p>

<p class=3DMsoAutoSig><strong><b><font size=3D1 face=3DVerdana><span
style=3D'font-size:7.5pt;font-family:Verdana'>Bala =
Pitchandi</span></font></b></strong></p>

<p class=3DMsoAutoSig><strong><b><font size=3D1 face=3DVerdana><span
style=3D'font-size:7.5pt;font-family:Verdana'>UTStarcom =
Inc.</span></font></b></strong></p>

</div>

</body>

</html>

------=_NextPart_000_0062_01C30838.6C83E5C0--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: bs@utstar.com
cc: avt@ietf.org, ruediger.kreuter@siemens.com
Subject: Re: [AVT] Status of draft-kreuter-avt-rtp-clearmode-02.txt 
In-Reply-To: Your message of "Mon, 21 Apr 2003 19:01:32 EDT."
             <006101c30859$f39585c0$2b2810ac@balanew> 
Date: Mon, 21 Apr 2003 19:22:57 -0400
From: Colin Perkins <csp@csperkins.org>
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--> "Balasubramanian Pitchandi" writes:
>I was wondering if this draft is still being considered as an action item
>for the Working Group. We are in need of an assignment for the "Clear Mode"
>(bit exact data with no encoding) codec to be used in our products.
>
>I could not find this draft in the IETF draft index and in the AVT WG home
>page as well.
>
>Please let me know the status of the draft.

The draft filename changed to draft-ietf-avt-rtp-clearmode-00.txt when it
was accepted as an AVT work item.  It is still under consideration by the
group (and may change before the final RFC is issued).

Colin
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Tue Apr 22 03:33:58 2003
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From: "Adam Li" <adamli@icsl.ucla.edu>
To: <avt@ietf.org>
Cc: "'Scott Bradner'" <sob@harvard.edu>, "'Allison Mankin'" <mankin@psg.com>,
        <craig.greer@nokia.com>, <jlee@nextreaming.com>,
        <sakazawa@kddilabs.jp>, <keith.miller@nokia.com>
Date: Tue, 22 Apr 2003 00:31:29 -0700
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Subject: [AVT] The storage format of EVRC/SMV vocoder (resent)
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Dear Steve, Colin, AVT'ers,

The issue of the storage format of the EVRC and SMV vocoders in the MIME
registration has came up a while ago, and has caused the last minute
holding of the MIME registration for these two vocoders. The issue has
been discussed thoroughly here on the mailing list in the San Francisco
meeting. The reasons of each side of the argument have been well
presented. Now it may be time to make a decision?

Like many other issues in our AVT WG, the decision here is ultimately
relying on the judgment of our chairs. During the two and half years
that this draft is being developed in the AVT, the course of the draft
has benefited greatly from your guidance. After the open and ego-free
discuss that our chairs called upon at the meeting session, it may be
another good time to hear the opinions from the chairs and get some help
on the decision from their experience and vision on which one will
reflect the consensus of the AVT and would be most beneficial to the
whole Internet and telecom industry who will use our specification.

Below is the summary of the rationales for each of the choices:

  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
------------------------------------+--------------------------------
(1) Mature specification, been      | (1) Brand new ID
    developed for 2.5 years in AVT  |
------------------------------------+--------------------------------
(2) Covers both EVRC and SMV        | (2) Only cover EVRC
------------------------------------+--------------------------------
(3) Referred to by 3GPP2(*) FFMS    | (3) 
    specifications                  |
------------------------------------+--------------------------------
(4) Implemented and used by many 3G | (4) Supported in Eudora
    telecom operators (e.g., SKT,   |
    KTF, LGT). The interoperability |
    has been tested among them.     |
------------------------------------+--------------------------------
(5) Simple and efficient design,    | (5)
    quick to implement              |
------------------------------------+--------------------------------
(6) Registration already in IANA,   | (6)
------------------------------------+--------------------------------
(7) Supported and developed by 7    | (7) Supported and developed by
    companies and universities      |     Qualcomm
    (including Qualcomm)            |
------------------------------------+--------------------------------
(8) No IP related to this format    | (8) IP situation unclear
------------------------------------+--------------------------------

(*) ERVC and SMV are both vocoders developed and defined in the
international consortium 3GPP2 (http://www.3gpp2.org) for the next
generation wireless network cdma2000.

Thank you very much.

Adam

PS. This is a resent of the previous message, which did not seem to go
through maybe because of the long cc list. If you think anyone is missed
out, please let me know. Thanks.

> -----Original Message-----
> From: Adam Li [mailto:adamli@icsl.ucla.edu]
> Sent: Saturday, March 15, 2003 9:20 PM
> To: 'Ietf-Avt'
> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
> 'villa@icsl.ucla.edu'; 'yllee@samsung.com'; 'jeonghoon@samsung.com';
> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
> 'nleung@qualcomm.com'; 'dgal@lucent.com'; 'ajayrajkumar@lucent.com';
> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
> Subject: Issues for the file format for EVRC/SMV vocoder
> 
> Hi folks,
> 
> The topic of the file format for EVRC/SMV vocoders hopefully will be
> discussed in this meeting at San Francisco. Below is a list of the
> issues that might be related to this topic. They are listed here as
> potential issues for you to consider before the discussion at the
> meeting.
> 
> (1) Technically differences. Is there much difference in efficiency
> and performance between the format in draft-ietf-avt-evrc-smv and
> draft-garudadri-qcp? Are the differences simply on the file syntax?
> 
> (2) Completeness. For EVRC codec data, formats defined in both
> document can handle it. For SMV codec data, the format defined in
> draft-ietf-avt-evrc-smv is currently the only file format that
> handles SMV data. draft-garudadri-qcp does not handle SMV at this
> time.
> 
> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv has
> its first version submitted to AVT on November 2000. It has been
> actively worked on all these years, and is co-authored by people
> from seven companies and universities. It is currently on the RFC
> editor's queue. draft-garudadri-qcp is submitted in February 2003.
> Will there be concerns about the maturity of the drafts,
> particularly the handling of SMV codec hasn't been written in the
> later draft yet?
> 
> (4) The recognition. Since EVRC and SMV codec are designed in 3GPP2
> for their CDMA networks, formats recognized by them are likely to be
> the most widely used format for those codecs. Which of the formats,
> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has
> been considered by 3GPP2 for the format for storing EVRC and SMV
> data?
> 
> (5) Document organization. Even though this is a rather minor point,
> but would there be enough reasons to trade-off for the additional
> complexity for having the MIME registration of EVRC/SMV refering to
> a separate and yet to be complete draft?
> 
> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If
> we want to consider deleting the file format that is in the draft
> for almost two years in the last minute before it becomes an RFC and
> using another yet to be complete new draft instead, we should be
> providing ourselves with the clear justification for doing so.
> 
> I will not be there for the discussion unfortunately since I have
> not made the plan to attend in advance. However, I hope the list of
> potential issues above might be useful for us to discuss and
> consider.
> 
> Wish we have a fruitful meeting in San Francisco.
> 
> Adam


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Tue Apr 22 13:29:23 2003
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To: "Adam Li" <adamli@icsl.ucla.edu>
cc: avt@ietf.org, "'Scott Bradner'" <sob@harvard.edu>,
        "'Allison Mankin'" <mankin@psg.com>, craig.greer@nokia.com,
        jlee@nextreaming.com, sakazawa@kddilabs.jp, keith.miller@nokia.com
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent) 
In-Reply-To: Your message of "Tue, 22 Apr 2003 00:31:29 PDT."
             <000801c308a1$345d6a40$657ba8c0@divxnetworks.com> 
Date: Tue, 22 Apr 2003 13:27:23 -0400
From: Colin Perkins <csp@csperkins.org>
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Adam,

The format specified in draft-garudadri-qcp-00.txt clearly needs to be
published, and to have a MIME type registered, since it has been used by
several applications for some years now. 

What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should refer to
that format, or if it should continue to use the alternative format. The
usual arguments apply: having two formats is clearly worse than one, but
there may be technical or political reasons why the new format is to be
preferred for some uses. 

As the minutes from the recent AVT meeting note, the main users of the
format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is currently
unclear if they have implementations of the file format, and if they chose
it for technical reasons or because it it was the only format they knew. We
are hoping to get input from them, to clarify their position.

Colin




--> "Adam Li" writes:
>
>The issue of the storage format of the EVRC and SMV vocoders in the MIME
>registration has came up a while ago, and has caused the last minute
>holding of the MIME registration for these two vocoders. The issue has
>been discussed thoroughly here on the mailing list in the San Francisco
>meeting. The reasons of each side of the argument have been well
>presented. Now it may be time to make a decision?
>
>Like many other issues in our AVT WG, the decision here is ultimately
>relying on the judgment of our chairs. During the two and half years
>that this draft is being developed in the AVT, the course of the draft
>has benefited greatly from your guidance. After the open and ego-free
>discuss that our chairs called upon at the meeting session, it may be
>another good time to hear the opinions from the chairs and get some help
>on the decision from their experience and vision on which one will
>reflect the consensus of the AVT and would be most beneficial to the
>whole Internet and telecom industry who will use our specification.
>
>Below is the summary of the rationales for each of the choices:
>
>  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
>------------------------------------+--------------------------------
>(1) Mature specification, been      | (1) Brand new ID
>    developed for 2.5 years in AVT  |
>------------------------------------+--------------------------------
>(2) Covers both EVRC and SMV        | (2) Only cover EVRC
>------------------------------------+--------------------------------
>(3) Referred to by 3GPP2(*) FFMS    | (3) 
>    specifications                  |
>------------------------------------+--------------------------------
>(4) Implemented and used by many 3G | (4) Supported in Eudora
>    telecom operators (e.g., SKT,   |
>    KTF, LGT). The interoperability |
>    has been tested among them.     |
>------------------------------------+--------------------------------
>(5) Simple and efficient design,    | (5)
>    quick to implement              |
>------------------------------------+--------------------------------
>(6) Registration already in IANA,   | (6)
>------------------------------------+--------------------------------
>(7) Supported and developed by 7    | (7) Supported and developed by
>    companies and universities      |     Qualcomm
>    (including Qualcomm)            |
>------------------------------------+--------------------------------
>(8) No IP related to this format    | (8) IP situation unclear
>------------------------------------+--------------------------------
>
>(*) ERVC and SMV are both vocoders developed and defined in the
>international consortium 3GPP2 (http://www.3gpp2.org) for the next
>generation wireless network cdma2000.
>
>Thank you very much.
>
>Adam
>
>PS. This is a resent of the previous message, which did not seem to go
>through maybe because of the long cc list. If you think anyone is missed
>out, please let me know. Thanks.
>
>> -----Original Message-----
>> From: Adam Li [mailto:adamli@icsl.ucla.edu]
>> Sent: Saturday, March 15, 2003 9:20 PM
>> To: 'Ietf-Avt'
>> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
>> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
>> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
>> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
>> 'villa@icsl.ucla.edu'; 'yllee@samsung.com'; 'jeonghoon@samsung.com';
>> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
>> 'nleung@qualcomm.com'; 'dgal@lucent.com'; 'ajayrajkumar@lucent.com';
>> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
>> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
>> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
>> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
>> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
>> Subject: Issues for the file format for EVRC/SMV vocoder
>> 
>> Hi folks,
>> 
>> The topic of the file format for EVRC/SMV vocoders hopefully will be
>> discussed in this meeting at San Francisco. Below is a list of the
>> issues that might be related to this topic. They are listed here as
>> potential issues for you to consider before the discussion at the
>> meeting.
>> 
>> (1) Technically differences. Is there much difference in efficiency
>> and performance between the format in draft-ietf-avt-evrc-smv and
>> draft-garudadri-qcp? Are the differences simply on the file syntax?
>> 
>> (2) Completeness. For EVRC codec data, formats defined in both
>> document can handle it. For SMV codec data, the format defined in
>> draft-ietf-avt-evrc-smv is currently the only file format that
>> handles SMV data. draft-garudadri-qcp does not handle SMV at this
>> time.
>> 
>> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv has
>> its first version submitted to AVT on November 2000. It has been
>> actively worked on all these years, and is co-authored by people
>> from seven companies and universities. It is currently on the RFC
>> editor's queue. draft-garudadri-qcp is submitted in February 2003.
>> Will there be concerns about the maturity of the drafts,
>> particularly the handling of SMV codec hasn't been written in the
>> later draft yet?
>> 
>> (4) The recognition. Since EVRC and SMV codec are designed in 3GPP2
>> for their CDMA networks, formats recognized by them are likely to be
>> the most widely used format for those codecs. Which of the formats,
>> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has
>> been considered by 3GPP2 for the format for storing EVRC and SMV
>> data?
>> 
>> (5) Document organization. Even though this is a rather minor point,
>> but would there be enough reasons to trade-off for the additional
>> complexity for having the MIME registration of EVRC/SMV refering to
>> a separate and yet to be complete draft?
>> 
>> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If
>> we want to consider deleting the file format that is in the draft
>> for almost two years in the last minute before it becomes an RFC and
>> using another yet to be complete new draft instead, we should be
>> providing ourselves with the clear justification for doing so.
> 
>> I will not be there for the discussion unfortunately since I have
>> not made the plan to attend in advance. However, I hope the list of
>> potential issues above might be useful for us to discuss and
>> consider.
>> 
>> Wish we have a fruitful meeting in San Francisco.
>> 
>> Adam
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Subject: RE: [AVT] The storage format of EVRC/SMV vocoder (resent) 
Date: Tue, 22 Apr 2003 12:38:56 -0500
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All,

This issue was discussed at the last 3GPP2 (April 14-18) meeting by the group developing the 3GPP2
file format specification. It was decided that both formats should be published, but
that Qualcomm would request a different MIME type for the EVRC storage in .QCP
(maybe EVRC_QCP). Hari Garudadri is expected to submit an updated
draft for ".QCP". Of course we need Qualcomm to verify this.

regards,
KEith Miller
Chair 3GPP2 TSG-C 1.2 WG

-----Original Message-----
From: ext Colin Perkins [mailto:csp@csperkins.org]
Sent: 22 April, 2003 12:27 PM
To: Adam Li
Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin'; Greer Craig
(Nokia/Dallas); jlee@nextreaming.com; sakazawa@kddilabs.jp; Miller Keith
(NRC/Dallas)
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent) 


Adam,

The format specified in draft-garudadri-qcp-00.txt clearly needs to be
published, and to have a MIME type registered, since it has been used by
several applications for some years now. 

What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should refer to
that format, or if it should continue to use the alternative format. The
usual arguments apply: having two formats is clearly worse than one, but
there may be technical or political reasons why the new format is to be
preferred for some uses. 

As the minutes from the recent AVT meeting note, the main users of the
format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is currently
unclear if they have implementations of the file format, and if they chose
it for technical reasons or because it it was the only format they knew. We
are hoping to get input from them, to clarify their position.

Colin




--> "Adam Li" writes:
>
>The issue of the storage format of the EVRC and SMV vocoders in the MIME
>registration has came up a while ago, and has caused the last minute
>holding of the MIME registration for these two vocoders. The issue has
>been discussed thoroughly here on the mailing list in the San Francisco
>meeting. The reasons of each side of the argument have been well
>presented. Now it may be time to make a decision?
>
>Like many other issues in our AVT WG, the decision here is ultimately
>relying on the judgment of our chairs. During the two and half years
>that this draft is being developed in the AVT, the course of the draft
>has benefited greatly from your guidance. After the open and ego-free
>discuss that our chairs called upon at the meeting session, it may be
>another good time to hear the opinions from the chairs and get some help
>on the decision from their experience and vision on which one will
>reflect the consensus of the AVT and would be most beneficial to the
>whole Internet and telecom industry who will use our specification.
>
>Below is the summary of the rationales for each of the choices:
>
>  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
>------------------------------------+--------------------------------
>(1) Mature specification, been      | (1) Brand new ID
>    developed for 2.5 years in AVT  |
>------------------------------------+--------------------------------
>(2) Covers both EVRC and SMV        | (2) Only cover EVRC
>------------------------------------+--------------------------------
>(3) Referred to by 3GPP2(*) FFMS    | (3) 
>    specifications                  |
>------------------------------------+--------------------------------
>(4) Implemented and used by many 3G | (4) Supported in Eudora
>    telecom operators (e.g., SKT,   |
>    KTF, LGT). The interoperability |
>    has been tested among them.     |
>------------------------------------+--------------------------------
>(5) Simple and efficient design,    | (5)
>    quick to implement              |
>------------------------------------+--------------------------------
>(6) Registration already in IANA,   | (6)
>------------------------------------+--------------------------------
>(7) Supported and developed by 7    | (7) Supported and developed by
>    companies and universities      |     Qualcomm
>    (including Qualcomm)            |
>------------------------------------+--------------------------------
>(8) No IP related to this format    | (8) IP situation unclear
>------------------------------------+--------------------------------
>
>(*) ERVC and SMV are both vocoders developed and defined in the
>international consortium 3GPP2 (http://www.3gpp2.org) for the next
>generation wireless network cdma2000.
>
>Thank you very much.
>
>Adam
>
>PS. This is a resent of the previous message, which did not seem to go
>through maybe because of the long cc list. If you think anyone is missed
>out, please let me know. Thanks.
>
>> -----Original Message-----
>> From: Adam Li [mailto:adamli@icsl.ucla.edu]
>> Sent: Saturday, March 15, 2003 9:20 PM
>> To: 'Ietf-Avt'
>> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
>> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
>> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
>> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
>> 'villa@icsl.ucla.edu'; 'yllee@samsung.com'; 'jeonghoon@samsung.com';
>> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
>> 'nleung@qualcomm.com'; 'dgal@lucent.com'; 'ajayrajkumar@lucent.com';
>> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
>> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
>> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
>> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
>> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
>> Subject: Issues for the file format for EVRC/SMV vocoder
>> 
>> Hi folks,
>> 
>> The topic of the file format for EVRC/SMV vocoders hopefully will be
>> discussed in this meeting at San Francisco. Below is a list of the
>> issues that might be related to this topic. They are listed here as
>> potential issues for you to consider before the discussion at the
>> meeting.
>> 
>> (1) Technically differences. Is there much difference in efficiency
>> and performance between the format in draft-ietf-avt-evrc-smv and
>> draft-garudadri-qcp? Are the differences simply on the file syntax?
>> 
>> (2) Completeness. For EVRC codec data, formats defined in both
>> document can handle it. For SMV codec data, the format defined in
>> draft-ietf-avt-evrc-smv is currently the only file format that
>> handles SMV data. draft-garudadri-qcp does not handle SMV at this
>> time.
>> 
>> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv has
>> its first version submitted to AVT on November 2000. It has been
>> actively worked on all these years, and is co-authored by people
>> from seven companies and universities. It is currently on the RFC
>> editor's queue. draft-garudadri-qcp is submitted in February 2003.
>> Will there be concerns about the maturity of the drafts,
>> particularly the handling of SMV codec hasn't been written in the
>> later draft yet?
>> 
>> (4) The recognition. Since EVRC and SMV codec are designed in 3GPP2
>> for their CDMA networks, formats recognized by them are likely to be
>> the most widely used format for those codecs. Which of the formats,
>> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has
>> been considered by 3GPP2 for the format for storing EVRC and SMV
>> data?
>> 
>> (5) Document organization. Even though this is a rather minor point,
>> but would there be enough reasons to trade-off for the additional
>> complexity for having the MIME registration of EVRC/SMV refering to
>> a separate and yet to be complete draft?
>> 
>> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If
>> we want to consider deleting the file format that is in the draft
>> for almost two years in the last minute before it becomes an RFC and
>> using another yet to be complete new draft instead, we should be
>> providing ourselves with the clear justification for doing so.
> 
>> I will not be there for the discussion unfortunately since I have
>> not made the plan to attend in advance. However, I hope the list of
>> potential issues above might be useful for us to discuss and
>> consider.
>> 
>> Wish we have a fruitful meeting in San Francisco.
>> 
>> Adam
>
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Tue Apr 22 14:12:41 2003
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To: keith.miller@nokia.com
cc: adamli@icsl.ucla.edu, avt@ietf.org, sob@harvard.edu, mankin@psg.com,
        craig.greer@nokia.com, jlee@nextreaming.com, sakazawa@kddilabs.jp
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent) 
In-Reply-To: Your message of "Tue, 22 Apr 2003 12:38:56 CDT."
             <8146199999A6384A8231B07B6E51D0BB10B3C5@daebe003.americas.nokia.com> 
Date: Tue, 22 Apr 2003 14:04:57 -0400
From: Colin Perkins <csp@csperkins.org>
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Keith,

Are there minutes of this meeting available? I'd like to understand the
rationale behind the choice (especially the choice of changing the MIME 
type, since "audio/qcp" seems appropriate).

Colin

--> keith.miller@nokia.com writes:
>All,
>
>This issue was discussed at the last 3GPP2 (April 14-18) meeting by the group developing the 3GPP2
>file format specification. It was decided that both formats should be published, but
>that Qualcomm would request a different MIME type for the EVRC storage in .QCP
>(maybe EVRC_QCP). Hari Garudadri is expected to submit an updated
>draft for ".QCP". Of course we need Qualcomm to verify this.
>
>regards,
>KEith Miller
>Chair 3GPP2 TSG-C 1.2 WG
>
>-----Original Message-----
>From: ext Colin Perkins [mailto:csp@csperkins.org]
>Sent: 22 April, 2003 12:27 PM
>To: Adam Li
>Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin'; Greer Craig
>(Nokia/Dallas); jlee@nextreaming.com; sakazawa@kddilabs.jp; Miller Keith
>(NRC/Dallas)
>Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent) 
>
>
>Adam,
>
>The format specified in draft-garudadri-qcp-00.txt clearly needs to be
>published, and to have a MIME type registered, since it has been used by
>several applications for some years now. 
>
>What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should refer to
>that format, or if it should continue to use the alternative format. The
>usual arguments apply: having two formats is clearly worse than one, but
>there may be technical or political reasons why the new format is to be
>preferred for some uses. 
>
>As the minutes from the recent AVT meeting note, the main users of the
>format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is currently
>unclear if they have implementations of the file format, and if they chose
>it for technical reasons or because it it was the only format they knew. We
>are hoping to get input from them, to clarify their position.
>
>Colin
>
>
>
>
>--> "Adam Li" writes:
>>
>>The issue of the storage format of the EVRC and SMV vocoders in the MIME
>>registration has came up a while ago, and has caused the last minute
>>holding of the MIME registration for these two vocoders. The issue has
>>been discussed thoroughly here on the mailing list in the San Francisco
>>meeting. The reasons of each side of the argument have been well
>>presented. Now it may be time to make a decision?
>>
>>Like many other issues in our AVT WG, the decision here is ultimately
>>relying on the judgment of our chairs. During the two and half years
>>that this draft is being developed in the AVT, the course of the draft
>>has benefited greatly from your guidance. After the open and ego-free
>>discuss that our chairs called upon at the meeting session, it may be
>>another good time to hear the opinions from the chairs and get some help
>>on the decision from their experience and vision on which one will
>>reflect the consensus of the AVT and would be most beneficial to the
>>whole Internet and telecom industry who will use our specification.
>>
>>Below is the summary of the rationales for each of the choices:
>>
>>  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
>>------------------------------------+--------------------------------
>>(1) Mature specification, been      | (1) Brand new ID
>>    developed for 2.5 years in AVT  |
>>------------------------------------+--------------------------------
>>(2) Covers both EVRC and SMV        | (2) Only cover EVRC
>>------------------------------------+--------------------------------
>>(3) Referred to by 3GPP2(*) FFMS    | (3) 
>>    specifications                  |
>>------------------------------------+--------------------------------
>>(4) Implemented and used by many 3G | (4) Supported in Eudora
>>    telecom operators (e.g., SKT,   |
>>    KTF, LGT). The interoperability |
>>    has been tested among them.     |
>>------------------------------------+--------------------------------
>>(5) Simple and efficient design,    | (5)
>>    quick to implement              |
>>------------------------------------+--------------------------------
>>(6) Registration already in IANA,   | (6)
>>------------------------------------+--------------------------------
>>(7) Supported and developed by 7    | (7) Supported and developed by
>>    companies and universities      |     Qualcomm
>>    (including Qualcomm)            |
>>------------------------------------+--------------------------------
>>(8) No IP related to this format    | (8) IP situation unclear
>>------------------------------------+--------------------------------
>>
>>(*) ERVC and SMV are both vocoders developed and defined in the
>>international consortium 3GPP2 (http://www.3gpp2.org) for the next
>>generation wireless network cdma2000.
>>
>>Thank you very much.
>>
>>Adam
>>
>>PS. This is a resent of the previous message, which did not seem to go
>>through maybe because of the long cc list. If you think anyone is missed
>>out, please let me know. Thanks.
>>
>>> -----Original Message-----
>>> From: Adam Li [mailto:adamli@icsl.ucla.edu]
>>> Sent: Saturday, March 15, 2003 9:20 PM
>>> To: 'Ietf-Avt'
>>> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
>>> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
>>> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
>>> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
>>> 'villa@icsl.ucla.edu'; 'yllee@samsung.com'; 'jeonghoon@samsung.com';
>>> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
>>> 'nleung@qualcomm.com'; 'dgal@lucent.com'; 'ajayrajkumar@lucent.com';
>>> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
>>> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
>>> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
>>> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
>>> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
>>> Subject: Issues for the file format for EVRC/SMV vocoder
>>> 
>>> Hi folks,
>>> 
>>> The topic of the file format for EVRC/SMV vocoders hopefully will be
>>> discussed in this meeting at San Francisco. Below is a list of the
>>> issues that might be related to this topic. They are listed here as
>>> potential issues for you to consider before the discussion at the
>>> meeting.
>>> 
>>> (1) Technically differences. Is there much difference in efficiency
>>> and performance between the format in draft-ietf-avt-evrc-smv and
>>> draft-garudadri-qcp? Are the differences simply on the file syntax?
>>> 
>>> (2) Completeness. For EVRC codec data, formats defined in both
>>> document can handle it. For SMV codec data, the format defined in
>>> draft-ietf-avt-evrc-smv is currently the only file format that
>>> handles SMV data. draft-garudadri-qcp does not handle SMV at this
>>> time.
>>> 
>>> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv has
>>> its first version submitted to AVT on November 2000. It has been
>>> actively worked on all these years, and is co-authored by people
>>> from seven companies and universities. It is currently on the RFC
>>> editor's queue. draft-garudadri-qcp is submitted in February 2003.
>>> Will there be concerns about the maturity of the drafts,
>>> particularly the handling of SMV codec hasn't been written in the
>>> later draft yet?
>>> 
>>> (4) The recognition. Since EVRC and SMV codec are designed in 3GPP2
>>> for their CDMA networks, formats recognized by them are likely to be
>>> the most widely used format for those codecs. Which of the formats,
>>> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has
>>> been considered by 3GPP2 for the format for storing EVRC and SMV
>>> data?
>>> 
>>> (5) Document organization. Even though this is a rather minor point,
>>> but would there be enough reasons to trade-off for the additional
>>> complexity for having the MIME registration of EVRC/SMV refering to
>>> a separate and yet to be complete draft?
>>> 
>>> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If
>>> we want to consider deleting the file format that is in the draft
>>> for almost two years in the last minute before it becomes an RFC and
>>> using another yet to be complete new draft instead, we should be
>>> providing ourselves with the clear justification for doing so.
>> 
>>> I will not be there for the discussion unfortunately since I have
>>> not made the plan to attend in advance. However, I hope the list of
>>> potential issues above might be useful for us to discuss and
>>> consider.
>>> 
>>> Wish we have a fruitful meeting in San Francisco.
>>> 
>>> Adam
>>
>>
>>_______________________________________________
>>Audio/Video Transport Working Group
>>avt@ietf.org
>>https://www1.ietf.org/mailman/listinfo/avt
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Adam Li" <adamli@icsl.ucla.edu>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: <avt@ietf.org>, "'Scott Bradner'" <sob@harvard.edu>,
        "'Allison Mankin'" <mankin@psg.com>, <craig.greer@nokia.com>,
        <jlee@nextreaming.com>, <sakazawa@kddilabs.jp>,
        <keith.miller@nokia.com>
Subject: RE: [AVT] The storage format of EVRC/SMV vocoder (resent) 
Date: Tue, 22 Apr 2003 23:45:12 -0700
Organization: UCLA
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Dear Colin,

Thank you very much for your email. We all have agreed on the fact that
.QCP is used by Eudora while .EVC/.SMV are used by 3G service providers.

However, I have difficulty seeing the rationale for your opinion that
the registration of .QCP is clearly needed while the registration of
.EVC/.SMV is the opposite. Would you mind explain to us AVT'ers your
rationales and pointing out the factors that we missed in the discussion
so we can better understand your opinions?

With all the factors that have been expressed on this issue which are
summarized in the previous email, it does not seem that the major
consensus is inclining that way. Maybe we missed something?

Your opinion is very much appreciated.

Adam


> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of
> Colin Perkins
> Sent: Tuesday, April 22, 2003 10:27 AM
> To: Adam Li
> Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin';
> craig.greer@nokia.com; jlee@nextreaming.com; sakazawa@kddilabs.jp;
> keith.miller@nokia.com
> Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
> 
> Adam,
> 
> The format specified in draft-garudadri-qcp-00.txt clearly needs to
> be
> published, and to have a MIME type registered, since it has been
> used by
> several applications for some years now.
> 
> What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should
> refer to
> that format, or if it should continue to use the alternative format.
> The
> usual arguments apply: having two formats is clearly worse than one,
> but
> there may be technical or political reasons why the new format is to
> be
> preferred for some uses.
> 
> As the minutes from the recent AVT meeting note, the main users of
> the
> format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is
> currently
> unclear if they have implementations of the file format, and if they
> chose
> it for technical reasons or because it it was the only format they
> knew. We
> are hoping to get input from them, to clarify their position.
> 
> Colin
> 
> 
> 
> 
> --> "Adam Li" writes:
> >
> >The issue of the storage format of the EVRC and SMV vocoders in the
> MIME
> >registration has came up a while ago, and has caused the last
> minute
> >holding of the MIME registration for these two vocoders. The issue
> has
> >been discussed thoroughly here on the mailing list in the San
> Francisco
> >meeting. The reasons of each side of the argument have been well
> >presented. Now it may be time to make a decision?
> >
> >Like many other issues in our AVT WG, the decision here is
> ultimately
> >relying on the judgment of our chairs. During the two and half
> years
> >that this draft is being developed in the AVT, the course of the
> draft
> >has benefited greatly from your guidance. After the open and ego-
> free
> >discuss that our chairs called upon at the meeting session, it may
> be
> >another good time to hear the opinions from the chairs and get some
> help
> >on the decision from their experience and vision on which one will
> >reflect the consensus of the AVT and would be most beneficial to
> the
> >whole Internet and telecom industry who will use our specification.
> >
> >Below is the summary of the rationales for each of the choices:
> >
> >  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
> >------------------------------------+------------------------------
> --
> >(1) Mature specification, been      | (1) Brand new ID
> >    developed for 2.5 years in AVT  |
> >------------------------------------+------------------------------
> --
> >(2) Covers both EVRC and SMV        | (2) Only cover EVRC
> >------------------------------------+------------------------------
> --
> >(3) Referred to by 3GPP2(*) FFMS    | (3)
> >    specifications                  |
> >------------------------------------+------------------------------
> --
> >(4) Implemented and used by many 3G | (4) Supported in Eudora
> >    telecom operators (e.g., SKT,   |
> >    KTF, LGT). The interoperability |
> >    has been tested among them.     |
> >------------------------------------+------------------------------
> --
> >(5) Simple and efficient design,    | (5)
> >    quick to implement              |
> >------------------------------------+------------------------------
> --
> >(6) Registration already in IANA,   | (6)
> >------------------------------------+------------------------------
> --
> >(7) Supported and developed by 7    | (7) Supported and developed
> by
> >    companies and universities      |     Qualcomm
> >    (including Qualcomm)            |
> >------------------------------------+------------------------------
> --
> >(8) No IP related to this format    | (8) IP situation unclear
> >------------------------------------+------------------------------
> --
> >
> >(*) ERVC and SMV are both vocoders developed and defined in the
> >international consortium 3GPP2 (http://www.3gpp2.org) for the next
> >generation wireless network cdma2000.
> >
> >Thank you very much.
> >
> >Adam
> >
> >PS. This is a resent of the previous message, which did not seem to
> go
> >through maybe because of the long cc list. If you think anyone is
> missed
> >out, please let me know. Thanks.
> >
> >> -----Original Message-----
> >> From: Adam Li [mailto:adamli@icsl.ucla.edu]
> >> Sent: Saturday, March 15, 2003 9:20 PM
> >> To: 'Ietf-Avt'
> >> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
> >> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
> >> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
> >> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
> >> 'villa@icsl.ucla.edu'; 'yllee@samsung.com';
> 'jeonghoon@samsung.com';
> >> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
> >> 'nleung@qualcomm.com'; 'dgal@lucent.com';
> 'ajayrajkumar@lucent.com';
> >> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
> >> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
> >> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
> >> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
> >> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
> >> Subject: Issues for the file format for EVRC/SMV vocoder
> >>
> >> Hi folks,
> >>
> >> The topic of the file format for EVRC/SMV vocoders hopefully will
> be
> >> discussed in this meeting at San Francisco. Below is a list of
> the
> >> issues that might be related to this topic. They are listed here
> as
> >> potential issues for you to consider before the discussion at the
> >> meeting.
> >>
> >> (1) Technically differences. Is there much difference in
> efficiency
> >> and performance between the format in draft-ietf-avt-evrc-smv and
> >> draft-garudadri-qcp? Are the differences simply on the file
> syntax?
> >>
> >> (2) Completeness. For EVRC codec data, formats defined in both
> >> document can handle it. For SMV codec data, the format defined in
> >> draft-ietf-avt-evrc-smv is currently the only file format that
> >> handles SMV data. draft-garudadri-qcp does not handle SMV at this
> >> time.
> >>
> >> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv
> has
> >> its first version submitted to AVT on November 2000. It has been
> >> actively worked on all these years, and is co-authored by people
> >> from seven companies and universities. It is currently on the RFC
> >> editor's queue. draft-garudadri-qcp is submitted in February 2003.
> >> Will there be concerns about the maturity of the drafts,
> >> particularly the handling of SMV codec hasn't been written in the
> >> later draft yet?
> >>
> >> (4) The recognition. Since EVRC and SMV codec are designed in
> 3GPP2
> >> for their CDMA networks, formats recognized by them are likely to
> be
> >> the most widely used format for those codecs. Which of the
> formats,
> >> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp,
> has
> >> been considered by 3GPP2 for the format for storing EVRC and SMV
> >> data?
> >>
> >> (5) Document organization. Even though this is a rather minor
> point,
> >> but would there be enough reasons to trade-off for the additional
> >> complexity for having the MIME registration of EVRC/SMV refering
> to
> >> a separate and yet to be complete draft?
> >>
> >> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue.
> If
> >> we want to consider deleting the file format that is in the draft
> >> for almost two years in the last minute before it becomes an RFC
> and
> >> using another yet to be complete new draft instead, we should be
> >> providing ourselves with the clear justification for doing so.
> >
> >> I will not be there for the discussion unfortunately since I have
> >> not made the plan to attend in advance. However, I hope the list
> of
> >> potential issues above might be useful for us to discuss and
> >> consider.
> >>
> >> Wish we have a fruitful meeting in San Francisco.
> >>
> >> Adam
> >
> >
> >_______________________________________________
> >Audio/Video Transport Working Group
> >avt@ietf.org
> >https://www1.ietf.org/mailman/listinfo/avt
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Wed Apr 23 09:53:12 2003
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To: "Adam Li" <adamli@icsl.ucla.edu>
cc: avt@ietf.org, "'Scott Bradner'" <sob@harvard.edu>,
        "'Allison Mankin'" <mankin@psg.com>, craig.greer@nokia.com,
        jlee@nextreaming.com, sakazawa@kddilabs.jp, keith.miller@nokia.com
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent) 
In-Reply-To: Your message of "Tue, 22 Apr 2003 23:45:12 PDT."
             <000b01c30963$e88ccde0$657ba8c0@divxnetworks.com> 
Date: Wed, 23 Apr 2003 09:52:17 -0400
From: Colin Perkins <csp@csperkins.org>
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Adam,

My understanding is that there have been implementations of the QCP format
for several years now, but the MIME type has not been registered. The MIME
type should be registered, to document this use.

This is independent of whether the EVRC/SMV file format is also registered.

Colin



--> "Adam Li" writes:
>Dear Colin,
>
>Thank you very much for your email. We all have agreed on the fact that
>.QCP is used by Eudora while .EVC/.SMV are used by 3G service providers.
>
>However, I have difficulty seeing the rationale for your opinion that
>the registration of .QCP is clearly needed while the registration of
>.EVC/.SMV is the opposite. Would you mind explain to us AVT'ers your
>rationales and pointing out the factors that we missed in the discussion
>so we can better understand your opinions?
>
>With all the factors that have been expressed on this issue which are
>summarized in the previous email, it does not seem that the major
>consensus is inclining that way. Maybe we missed something?
>
>Your opinion is very much appreciated.
>
>Adam
>
>
>> -----Original Message-----
>> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of
>> Colin Perkins
>> Sent: Tuesday, April 22, 2003 10:27 AM
>> To: Adam Li
>> Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin';
>> craig.greer@nokia.com; jlee@nextreaming.com; sakazawa@kddilabs.jp;
>> keith.miller@nokia.com
>> Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
>> 
>> Adam,
>> 
>> The format specified in draft-garudadri-qcp-00.txt clearly needs to
>> be
>> published, and to have a MIME type registered, since it has been
>> used by
>> several applications for some years now.
>> 
>> What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should
>> refer to
>> that format, or if it should continue to use the alternative format.
>> The
>> usual arguments apply: having two formats is clearly worse than one,
>> but
>> there may be technical or political reasons why the new format is to
>> be
>> preferred for some uses.
>> 
>> As the minutes from the recent AVT meeting note, the main users of
>> the
>> format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is
>> currently
>> unclear if they have implementations of the file format, and if they
>> chose
>> it for technical reasons or because it it was the only format they
>> knew. We
>> are hoping to get input from them, to clarify their position.
>> 
>> Colin
>> 
>> 
>> 
>> 
>> --> "Adam Li" writes:
>> >
>> >The issue of the storage format of the EVRC and SMV vocoders in the
>> MIME
>> >registration has came up a while ago, and has caused the last
>> minute
>> >holding of the MIME registration for these two vocoders. The issue
>> has
>> >been discussed thoroughly here on the mailing list in the San
>> Francisco
>> >meeting. The reasons of each side of the argument have been well
>> >presented. Now it may be time to make a decision?
>> >
>> >Like many other issues in our AVT WG, the decision here is
>> ultimately
>> >relying on the judgment of our chairs. During the two and half
>> years
>> >that this draft is being developed in the AVT, the course of the
>> draft
>> >has benefited greatly from your guidance. After the open and ego-
>> free
>> >discuss that our chairs called upon at the meeting session, it may
>> be
>> >another good time to hear the opinions from the chairs and get some
>> help
>> >on the decision from their experience and vision on which one will
>> >reflect the consensus of the AVT and would be most beneficial to
>> the
>> >whole Internet and telecom industry who will use our specification.
>> >
>> >Below is the summary of the rationales for each of the choices:
>> >
>> >  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
>> >------------------------------------+------------------------------
>> --
>> >(1) Mature specification, been      | (1) Brand new ID
>> >    developed for 2.5 years in AVT  |
>> >------------------------------------+------------------------------
>> --
>> >(2) Covers both EVRC and SMV        | (2) Only cover EVRC
>> >------------------------------------+------------------------------
>> --
>> >(3) Referred to by 3GPP2(*) FFMS    | (3)
>> >    specifications                  |
>> >------------------------------------+------------------------------
>> --
>> >(4) Implemented and used by many 3G | (4) Supported in Eudora
>> >    telecom operators (e.g., SKT,   |
>> >    KTF, LGT). The interoperability |
>> >    has been tested among them.     |
>> >------------------------------------+------------------------------
>> --
>> >(5) Simple and efficient design,    | (5)
>> >    quick to implement              |
>> >------------------------------------+------------------------------
>> --
>> >(6) Registration already in IANA,   | (6)
>> >------------------------------------+------------------------------
>> --
>> >(7) Supported and developed by 7    | (7) Supported and developed
>> by
>> >    companies and universities      |     Qualcomm
>> >    (including Qualcomm)            |
>> >------------------------------------+------------------------------
>> --
>> >(8) No IP related to this format    | (8) IP situation unclear
>> >------------------------------------+------------------------------
>> --
>> >
>> >(*) ERVC and SMV are both vocoders developed and defined in the
>> >international consortium 3GPP2 (http://www.3gpp2.org) for the next
>> >generation wireless network cdma2000.
>> >
>> >Thank you very much.
>> >
>> >Adam
>> >
>> >PS. This is a resent of the previous message, which did not seem to
>> go
>> >through maybe because of the long cc list. If you think anyone is
>> missed
>> >out, please let me know. Thanks.
>> >
>> >> -----Original Message-----
>> >> From: Adam Li [mailto:adamli@icsl.ucla.edu]
>> >> Sent: Saturday, March 15, 2003 9:20 PM
>> >> To: 'Ietf-Avt'
>> >> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
>> >> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
>> >> 'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
>> >> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
>> >> 'villa@icsl.ucla.edu'; 'yllee@samsung.com';
>> 'jeonghoon@samsung.com';
>> >> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
>> >> 'nleung@qualcomm.com'; 'dgal@lucent.com';
>> 'ajayrajkumar@lucent.com';
>> >> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
>> >> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
>> >> 'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
>> >> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
>> >> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
>> >> Subject: Issues for the file format for EVRC/SMV vocoder
>> >>
>> >> Hi folks,
>> >>
>> >> The topic of the file format for EVRC/SMV vocoders hopefully will
>> be
>> >> discussed in this meeting at San Francisco. Below is a list of
>> the
>> >> issues that might be related to this topic. They are listed here
>> as
>> >> potential issues for you to consider before the discussion at the
>> >> meeting.
>> >>
>> >> (1) Technically differences. Is there much difference in
>> efficiency
>> >> and performance between the format in draft-ietf-avt-evrc-smv and
>> >> draft-garudadri-qcp? Are the differences simply on the file
>> syntax?
>> >>
>> >> (2) Completeness. For EVRC codec data, formats defined in both
>> >> document can handle it. For SMV codec data, the format defined in
>> >> draft-ietf-avt-evrc-smv is currently the only file format that
>> >> handles SMV data. draft-garudadri-qcp does not handle SMV at this
>> >> time.
>> >>
>> >> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv
>> has
>> >> its first version submitted to AVT on November 2000. It has been
>> >> actively worked on all these years, and is co-authored by people
>> >> from seven companies and universities. It is currently on the RFC
>> >> editor's queue. draft-garudadri-qcp is submitted in February 2003.
>> >> Will there be concerns about the maturity of the drafts,
>> >> particularly the handling of SMV codec hasn't been written in the
>> >> later draft yet?
>> >>
>> >> (4) The recognition. Since EVRC and SMV codec are designed in
>> 3GPP2
>> >> for their CDMA networks, formats recognized by them are likely to
>> be
>> >> the most widely used format for those codecs. Which of the
>> formats,
>> >> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp,
>> has
>> >> been considered by 3GPP2 for the format for storing EVRC and SMV
>> >> data?
>> >>
>> >> (5) Document organization. Even though this is a rather minor
>> point,
>> >> but would there be enough reasons to trade-off for the additional
>> >> complexity for having the MIME registration of EVRC/SMV refering
>> to
>> >> a separate and yet to be complete draft?
>> >>
>> >> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue.
>> If
>> >> we want to consider deleting the file format that is in the draft
>> >> for almost two years in the last minute before it becomes an RFC
>> and
>> >> using another yet to be complete new draft instead, we should be
>> >> providing ourselves with the clear justification for doing so.
>> >
>> >> I will not be there for the discussion unfortunately since I have
>> >> not made the plan to attend in advance. However, I hope the list
>> of
>> >> potential issues above might be useful for us to discuss and
>> >> consider.
>> >>
>> >> Wish we have a fruitful meeting in San Francisco.
>> >>
>> >> Adam
>> >
>> >
>> >_______________________________________________
>> >Audio/Video Transport Working Group
>> >avt@ietf.org
>> >https://www1.ietf.org/mailman/listinfo/avt
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Section "6 Payload Type Definitions" refers to Tables 4 and 5.  However, 
those tables actually appear in the middle of the following section: "7 RTP 
over TCP and Similar Byte Stream Protocols".  Confusing...

         Ross.


_______________________________________________
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avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: "Adam Li" <adamli@icsl.ucla.edu>, <avt@ietf.org>
From: Randall Gellens <randy@qualcomm.com>
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
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At 12:31 AM -0700 4/22/03, Adam Li wrote:

>  Below is the summary of the rationales for each of the choices:
>
>    draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-00.txt
>  ------------------------------------+--------------------------------
>  (1) Mature specification, been      | (1) Brand new ID
>      developed for 2.5 years in AVT  |

I'm not sure it makes sense to call a specification "mature" based on 
the amount of time between inception and standardization.  That is, I 
believe "mature" refers to the amount of time multiple interoperating 
implementations have been deployed.

I also do not believe it is accurate to imply that the qcp format is 
not mature simply because the Internet Draft is new.  In reality, the 
specification has been available for some years now, and there are a 
number of implementations which do interoperate.

>  ------------------------------------+--------------------------------
>  (2) Covers both EVRC and SMV        | (2) Only cover EVRC

The qcp format covers QCELP 13K as well as EVRC.  An updated version 
covers SMV as well.

>  ------------------------------------+--------------------------------
>  (3) Referred to by 3GPP2(*) FFMS    | (3)
>      specifications                  |

My understanding is that the 3GPP2 FFMS specifications refer to both 
(it is waiting on RFC publications for both).

>  ------------------------------------+--------------------------------
>  (4) Implemented and used by many 3G | (4) Supported in Eudora
>      telecom operators (e.g., SKT,   |
>      KTF, LGT). The interoperability |
>      has been tested among them.     |

The qcp format is supported by PureVoice and QuickTime, both 
available on Windows and Macs.   Also, a large number of CDMA 
handsets deployed today create and play qcp files and use this to 
interoperate.  A number of services and tools on the web use the qcp 
format (often calling it PureVoice).

My information on the implementation of the evrc-smv draft's format 
is that Nextreaming uses this as an internal intermediate format 
prior to generating ".3g2" files (as specified in the 3GPP2 FFMS 
specification, which include both formats).  That is, my 
understanding is that this implementation is not used to exchange 
data.  If my understanding is not correct, please let me know.  Can 
you please provide more information on the telecom operators (e.g., 
SKT,  KTF, LGT) implementations that you refer to, and if these 
implementations are used to exchange data in the format specified in 
the evrc-smv draft?

>  ------------------------------------+--------------------------------
>  (5) Simple and efficient design,    | (5)
>      quick to implement              |

The text in the qcp draft is more confusing than it should be.  An 
updated version is about to be published which replaces the pseudo-C 
description with clear ABNF, showing that the two formats are 
actually very close.  In essence, the qcp format is the same as the 
evrc-smv format, except that some additional data is present at the 
start of the file.  But in both formats, individual vocoder frames 
are stored with a preceding byte which describes the frame size.

>  ------------------------------------+--------------------------------
>  (6) Registration already in IANA,   | (6)
>  ------------------------------------+--------------------------------
>  (7) Supported and developed by 7    | (7) Supported and developed by
>      companies and universities      |     Qualcomm
>      (including Qualcomm)            |

The qcp format is supported/implemented by QuickTime, PureVoice, and 
numerous handset vendors and operators, plus a number of web services.

>  ------------------------------------+--------------------------------
>  (8) No IP related to this format    | (8) IP situation unclear

There is no IP associated with the the qcp format.  (The draft 
includes an explicit statement that it is in full compliance with all 
provisions of Section 10 of RFC2026.)

>  ------------------------------------+--------------------------------
>
>  (*) ERVC and SMV are both vocoders developed and defined in the
>  international consortium 3GPP2 (http://www.3gpp2.org) for the next
>  generation wireless network cdma2000.
>
>  Thank you very much.
>
>  Adam
>
>  PS. This is a resent of the previous message, which did not seem to go
>  through maybe because of the long cc list. If you think anyone is missed
>  out, please let me know. Thanks.
>
>>  -----Original Message-----
>>  From: Adam Li [mailto:adamli@icsl.ucla.edu]
>>  Sent: Saturday, March 15, 2003 9:20 PM
>>  To: 'Ietf-Avt'
>>  Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins'; 'Allison
>>  Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
>>  'mdturner@lucent.com'; 'smathai@lucent.com'; 'lioy@qualcomm.com';
>>  'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
>>  'villa@icsl.ucla.edu'; 'yllee@samsung.com'; 'jeonghoon@samsung.com';
>>  'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
>>  'nleung@qualcomm.com'; 'dgal@lucent.com'; 'ajayrajkumar@lucent.com';
>>  'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
>>  t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
>>  'magda@qualcomm.com'; 'casner@acm.org'; 'ned.freed@mrochek.com';
>>  'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
>>  'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp'; 'tsgc@3gpp2.org'
>>  Subject: Issues for the file format for EVRC/SMV vocoder
>>
>>  Hi folks,
>>
>>  The topic of the file format for EVRC/SMV vocoders hopefully will be
>>  discussed in this meeting at San Francisco. Below is a list of the
>>  issues that might be related to this topic. They are listed here as
>>  potential issues for you to consider before the discussion at the
>>  meeting.
>>
>>  (1) Technically differences. Is there much difference in efficiency
>>  and performance between the format in draft-ietf-avt-evrc-smv and
>>  draft-garudadri-qcp? Are the differences simply on the file syntax?
>>
>>  (2) Completeness. For EVRC codec data, formats defined in both
>>  document can handle it. For SMV codec data, the format defined in
>>  draft-ietf-avt-evrc-smv is currently the only file format that
>>  handles SMV data. draft-garudadri-qcp does not handle SMV at this
>>  time.
>>
>>  (3) Maturity of the format definition. draft-ietf-avt-evrc-smv has
>>  its first version submitted to AVT on November 2000. It has been
>>  actively worked on all these years, and is co-authored by people
>>  from seven companies and universities. It is currently on the RFC
>>  editor's queue. draft-garudadri-qcp is submitted in February 2003.
>>  Will there be concerns about the maturity of the drafts,
>>  particularly the handling of SMV codec hasn't been written in the
>>  later draft yet?
>>
>>  (4) The recognition. Since EVRC and SMV codec are designed in 3GPP2
>>  for their CDMA networks, formats recognized by them are likely to be
>>  the most widely used format for those codecs. Which of the formats,
>>  as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp, has
>>  been considered by 3GPP2 for the format for storing EVRC and SMV
>>  data?
>>
>>  (5) Document organization. Even though this is a rather minor point,
>>  but would there be enough reasons to trade-off for the additional
>>  complexity for having the MIME registration of EVRC/SMV refering to
>>  a separate and yet to be complete draft?
>>
>>  Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue. If
>>  we want to consider deleting the file format that is in the draft
>>  for almost two years in the last minute before it becomes an RFC and
>>  using another yet to be complete new draft instead, we should be
>>  providing ourselves with the clear justification for doing so.
>>
>>  I will not be there for the discussion unfortunately since I have
>>  not made the plan to attend in advance. However, I hope the list of
>>  potential issues above might be useful for us to discuss and
>>  consider.
>>
>>  Wish we have a fruitful meeting in San Francisco.
>>
>>  Adam
>
>
>  _______________________________________________
>  Audio/Video Transport Working Group
>  avt@ietf.org
>  https://www1.ietf.org/mailman/listinfo/avt


_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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To: Colin Perkins <csp@csperkins.org>, keith.miller@nokia.com,
        Ned Freed <ned.freed@mrochek.com>
From: Randall Gellens <randy@qualcomm.com>
Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
Cc: adamli@icsl.ucla.edu, avt@ietf.org, sob@harvard.edu, mankin@psg.com,
        craig.greer@nokia.com, jlee@nextreaming.com, sakazawa@kddilabs.jp
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At 2:04 PM -0400 4/22/03, Colin Perkins wrote:

>  Are there minutes of this meeting available? I'd like to understand the
>  rationale behind the choice (especially the choice of changing the MIME
>  type, since "audio/qcp" seems appropriate).

Regarding the MIME type, "audio/qcp" was the initial suggested MIME 
type, but there was pushback based on the principle that MIME types 
should describe the content, not the format.  In general, neglect of 
this principle reduces interoperability, since it then means that an 
implementation that supports a certain MIME type may not be able to 
handle content received in that MIME type.  So, in accordance with 
the principle, it makes sense to use "audio/qcelp" for QCELP 13K data 
in this format, "audio/evrc-qcp" for EVRC data stored in the QCP 
format, and "audio/smv-qcp" for SMV data in the QCP format.

I suppose "audio/qcelp" plus "audio/qcp" with a mandatory parameter 
for the codec included, which could only be SMV or EVRC, not QCELP, 
could also work.  I've added Ned Freed to the distribution list to 
get his views, as the IANA-designated MIME expert.
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Randall Gellens <randy@qualcomm.com>
Subject: RE: [AVT] The storage format of EVRC/SMV vocoder (resent)
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        "'Allison Mankin'" <mankin@psg.com>, <craig.greer@nokia.com>,
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At 11:45 PM -0700 4/22/03, Adam Li wrote:

>   We all have agreed on the fact that
>  .QCP is used by Eudora while .EVC/.SMV are used by 3G service providers.

This isn't accurate.  As I noted in my earlier message to the list, 
the QCP format is used by QuickTime and PureVoice, as well as a large 
number of CDMA handsets, and a number of web services.  (Please see 
my earlier message for more detailed comments.)
-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly-selected tag: ---------------
Misery loves company, but company does not reciprocate.
_______________________________________________
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To: keith.miller@nokia.com, <csp@csperkins.org>, <adamli@icsl.ucla.edu>
From: Randall Gellens <randy@qualcomm.com>
Subject: RE: [AVT] The storage format of EVRC/SMV vocoder (resent)
Cc: <avt@ietf.org>, <sob@harvard.edu>, <mankin@psg.com>,
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At 12:38 PM -0500 4/22/03, keith.miller@nokia.com wrote:

>  This issue was discussed at the last 3GPP2 (April 14-18) meeting by 
> the group developing the 3GPP2
>  file format specification. It was decided that both formats should 
> be published, but
>  that Qualcomm would request a different MIME type for the EVRC 
> storage in .QCP
>  (maybe EVRC_QCP). Hari Garudadri is expected to submit an updated
>  draft for ".QCP". Of course we need Qualcomm to verify this.

Yes, this was the agreed resolution; an updated qcp draft will be 
published.  See my earlier message to the list for the MIME type 
discussion.
-- 
Randall Gellens
Opinions are personal;    facts are suspect;    I speak for myself only
-------------- Randomly-selected tag: ---------------
I make a fortune from criticizing the policy of the government, and
then hand it over to the government in taxes to keep it going.
                                              --George Bernard Shaw
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From mailnull@www1.ietf.org  Thu Apr 24 01:14:35 2003
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Date: Wed, 23 Apr 2003 16:32:11 -0700
From: ramin <rrezaiif@san.rr.com>
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Subject: [AVT] RTP timestamp
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<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
  <meta http-equiv="Content-Type" content="text/html;charset=ISO-8859-1">
  <title></title>
</head>
<body text="#000000" bgcolor="#ffffff">
<font face=""><br>
Assuming that both sender and receiver have access to accurate time, is
there any reason not to let the RTP timestamp reflect the real time (w/
the resolution specified in the profile and with a start time agreed
upon by both sender and receiver)?&nbsp; <br>
<br>
Thank you,<br>
<br>
-Ramin<br>
</font>
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From: "Adam Li" <adamli@icsl.ucla.edu>
To: "'Colin Perkins'" <csp@csperkins.org>
Cc: <avt@ietf.org>, "'Scott Bradner'" <sob@harvard.edu>,
        "'Allison Mankin'" <mankin@psg.com>, <craig.greer@nokia.com>,
        <jlee@nextreaming.com>, <sakazawa@kddilabs.jp>,
        <keith.miller@nokia.com>
Subject: RE: [AVT] The storage format of EVRC/SMV vocoder (resent) 
Date: Thu, 24 Apr 2003 01:34:12 -0700
Organization: UCLA
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Dear Colin,

Thank you very much for your email. I completely agree with you that
these two drafts are rather independent drafts which are not necessarily
exclusive of each other. As you mentioned, both formats have been used
by one or more applications in the software and telecom industries, and
we all agree with you that it will be very helpful to listen to the
opinions of the industries which will rely upon IETF for the protocol
standards.

The registration MIME type of the EVRC and SMV codec has been halted at
IANA in the last minute because of the new draft on this storage format
issue. From the perspective of participants in IETF and as
engineers/researchers, we all wish to see the RTP payload format for
these codecs being used rather than proprietary formats being
implemented in the systems which will cause the interoperability
problems later. It will be particularly disturbing if this would happen
while the international standards are waiting to replace the "Work in
progress" with "RFC xxxx".

The RTP payload type specification for EVRC and SMV has been in working
in AVT for 2.5 years, and is now stalled at RFC editor's queue. Two and
a half years is a long time in the telecommunication industry, and it is
a long time for Internet community too. 

I believe whatever proposal one supports, we all hope to see a swift
ending to this issue with a fair conclusion while giving each of the
proposed format the objective consideration it deserves.

Thank you very much,

Adam


> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of
> Colin Perkins
> Sent: Wednesday, April 23, 2003 6:52 AM
> To: Adam Li
> Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin';
> craig.greer@nokia.com; jlee@nextreaming.com; sakazawa@kddilabs.jp;
> keith.miller@nokia.com
> Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
> 
> Adam,
> 
> My understanding is that there have been implementations of the QCP
> format
> for several years now, but the MIME type has not been registered.
> The MIME
> type should be registered, to document this use.
> 
> This is independent of whether the EVRC/SMV file format is also
> registered.
> 
> Colin
> 
> 
> 
> --> "Adam Li" writes:
> >Dear Colin,
> >
> >Thank you very much for your email. We all have agreed on the fact
> that
> >.QCP is used by Eudora while .EVC/.SMV are used by 3G service
> providers.
> >
> >However, I have difficulty seeing the rationale for your opinion
> that
> >the registration of .QCP is clearly needed while the registration
> of
> >.EVC/.SMV is the opposite. Would you mind explain to us AVT'ers
> your
> >rationales and pointing out the factors that we missed in the
> discussion
> >so we can better understand your opinions?
> >
> >With all the factors that have been expressed on this issue which
> are
> >summarized in the previous email, it does not seem that the major
> >consensus is inclining that way. Maybe we missed something?
> >
> >Your opinion is very much appreciated.
> >
> >Adam
> >
> >
> >> -----Original Message-----
> >> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of
> >> Colin Perkins
> >> Sent: Tuesday, April 22, 2003 10:27 AM
> >> To: Adam Li
> >> Cc: avt@ietf.org; 'Scott Bradner'; 'Allison Mankin';
> >> craig.greer@nokia.com; jlee@nextreaming.com; sakazawa@kddilabs.jp;
> >> keith.miller@nokia.com
> >> Subject: Re: [AVT] The storage format of EVRC/SMV vocoder (resent)
> >>
> >> Adam,
> >>
> >> The format specified in draft-garudadri-qcp-00.txt clearly needs
> to
> >> be
> >> published, and to have a MIME type registered, since it has been
> >> used by
> >> several applications for some years now.
> >>
> >> What is unclear is whether draft-ieft-avt-evrc-smv-03.txt should
> >> refer to
> >> that format, or if it should continue to use the alternative
> format.
> >> The
> >> usual arguments apply: having two formats is clearly worse than
> one,
> >> but
> >> there may be technical or political reasons why the new format is
> to
> >> be
> >> preferred for some uses.
> >>
> >> As the minutes from the recent AVT meeting note, the main users
> of
> >> the
> >> format in draft-ieft-avt-evrc-smv-03.txt seem to be 3GPP2. It is
> >> currently
> >> unclear if they have implementations of the file format, and if
> they
> >> chose
> >> it for technical reasons or because it it was the only format
> they
> >> knew. We
> >> are hoping to get input from them, to clarify their position.
> >>
> >> Colin
> >>
> >>
> >>
> >>
> >> --> "Adam Li" writes:
> >> >
> >> >The issue of the storage format of the EVRC and SMV vocoders in
> the
> >> MIME
> >> >registration has came up a while ago, and has caused the last
> >> minute
> >> >holding of the MIME registration for these two vocoders. The
> issue
> >> has
> >> >been discussed thoroughly here on the mailing list in the San
> >> Francisco
> >> >meeting. The reasons of each side of the argument have been well
> >> >presented. Now it may be time to make a decision?
> >> >
> >> >Like many other issues in our AVT WG, the decision here is
> >> ultimately
> >> >relying on the judgment of our chairs. During the two and half
> >> years
> >> >that this draft is being developed in the AVT, the course of the
> >> draft
> >> >has benefited greatly from your guidance. After the open and
> ego-
> >> free
> >> >discuss that our chairs called upon at the meeting session, it
> may
> >> be
> >> >another good time to hear the opinions from the chairs and get
> some
> >> help
> >> >on the decision from their experience and vision on which one
> will
> >> >reflect the consensus of the AVT and would be most beneficial to
> >> the
> >> >whole Internet and telecom industry who will use our
> specification.
> >> >
> >> >Below is the summary of the rationales for each of the choices:
> >> >
> >> >  draft-ieft-avt-evrc-smv-03.txt    |    draft-garudadri-qcp-
> 00.txt
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(1) Mature specification, been      | (1) Brand new ID
> >> >    developed for 2.5 years in AVT  |
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(2) Covers both EVRC and SMV        | (2) Only cover EVRC
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(3) Referred to by 3GPP2(*) FFMS    | (3)
> >> >    specifications                  |
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(4) Implemented and used by many 3G | (4) Supported in Eudora
> >> >    telecom operators (e.g., SKT,   |
> >> >    KTF, LGT). The interoperability |
> >> >    has been tested among them.     |
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(5) Simple and efficient design,    | (5)
> >> >    quick to implement              |
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(6) Registration already in IANA,   | (6)
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(7) Supported and developed by 7    | (7) Supported and
> developed
> >> by
> >> >    companies and universities      |     Qualcomm
> >> >    (including Qualcomm)            |
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >(8) No IP related to this format    | (8) IP situation unclear
> >> >------------------------------------+---------------------------
> ---
> >> --
> >> >
> >> >(*) ERVC and SMV are both vocoders developed and defined in the
> >> >international consortium 3GPP2 (http://www.3gpp2.org) for the
> next
> >> >generation wireless network cdma2000.
> >> >
> >> >Thank you very much.
> >> >
> >> >Adam
> >> >
> >> >PS. This is a resent of the previous message, which did not seem
> to
> >> go
> >> >through maybe because of the long cc list. If you think anyone
> is
> >> missed
> >> >out, please let me know. Thanks.
> >> >
> >> >> -----Original Message-----
> >> >> From: Adam Li [mailto:adamli@icsl.ucla.edu]
> >> >> Sent: Saturday, March 15, 2003 9:20 PM
> >> >> To: 'Ietf-Avt'
> >> >> Cc: 'casner@acm.org'; 'Scott Bradner'; 'Colin Perkins';
> 'Allison
> >> >> Mankin'; 'randy@qualcomm.com'; 'mccap@lucent.com';
> >> >> 'mdturner@lucent.com'; 'smathai@lucent.com';
> 'lioy@qualcomm.com';
> >> >> 'zeng@packetvideo.com'; 'sherwood@packetvideo.com';
> >> >> 'villa@icsl.ucla.edu'; 'yllee@samsung.com';
> >> 'jeonghoon@samsung.com';
> >> >> 'tom.hiller@lucent.com'; 'David.Leon@nokia.com';
> >> >> 'nleung@qualcomm.com'; 'dgal@lucent.com';
> >> 'ajayrajkumar@lucent.com';
> >> >> 'Lars-Erik.Jonsson@epl.ericsson.se'; 'magnus.westerlund@era-
> >> >> t.ericsson.se'; 'vbharga@cisco.com'; 'craig.greer@nokia.com';
> >> >> 'magda@qualcomm.com'; 'casner@acm.org';
> 'ned.freed@mrochek.com';
> >> >> 'mankin@psg.com'; 'hgarudad@qualcomm.com'; 'csp@isi.edu';
> >> >> 'jlee@nextreaming.com'; 'sakazawa@kddilabs.jp';
> 'tsgc@3gpp2.org'
> >> >> Subject: Issues for the file format for EVRC/SMV vocoder
> >> >>
> >> >> Hi folks,
> >> >>
> >> >> The topic of the file format for EVRC/SMV vocoders hopefully
> will
> >> be
> >> >> discussed in this meeting at San Francisco. Below is a list of
> >> the
> >> >> issues that might be related to this topic. They are listed
> here
> >> as
> >> >> potential issues for you to consider before the discussion at
> the
> >> >> meeting.
> >> >>
> >> >> (1) Technically differences. Is there much difference in
> >> efficiency
> >> >> and performance between the format in draft-ietf-avt-evrc-smv
> and
> >> >> draft-garudadri-qcp? Are the differences simply on the file
> >> syntax?
> >> >>
> >> >> (2) Completeness. For EVRC codec data, formats defined in both
> >> >> document can handle it. For SMV codec data, the format defined
> in
> >> >> draft-ietf-avt-evrc-smv is currently the only file format that
> >> >> handles SMV data. draft-garudadri-qcp does not handle SMV at
> this
> >> >> time.
> >> >>
> >> >> (3) Maturity of the format definition. draft-ietf-avt-evrc-smv
> >> has
> >> >> its first version submitted to AVT on November 2000. It has
> been
> >> >> actively worked on all these years, and is co-authored by
> people
> >> >> from seven companies and universities. It is currently on the
> RFC
> >> >> editor's queue. draft-garudadri-qcp is submitted in February
> 2003.
> >> >> Will there be concerns about the maturity of the drafts,
> >> >> particularly the handling of SMV codec hasn't been written in
> the
> >> >> later draft yet?
> >> >>
> >> >> (4) The recognition. Since EVRC and SMV codec are designed in
> >> 3GPP2
> >> >> for their CDMA networks, formats recognized by them are likely
> to
> >> be
> >> >> the most widely used format for those codecs. Which of the
> >> formats,
> >> >> as defined in draft-ietf-avt-evrc-smv and draft-garudadri-qcp,
> >> has
> >> >> been considered by 3GPP2 for the format for storing EVRC and
> SMV
> >> >> data?
> >> >>
> >> >> (5) Document organization. Even though this is a rather minor
> >> point,
> >> >> but would there be enough reasons to trade-off for the
> additional
> >> >> complexity for having the MIME registration of EVRC/SMV
> refering
> >> to
> >> >> a separate and yet to be complete draft?
> >> >>
> >> >> Draft-ietf-avt-evrc-smv is currently on the RFC editor's queue.
> >> If
> >> >> we want to consider deleting the file format that is in the
> draft
> >> >> for almost two years in the last minute before it becomes an
> RFC
> >> and
> >> >> using another yet to be complete new draft instead, we should
> be
> >> >> providing ourselves with the clear justification for doing so.
> >> >
> >> >> I will not be there for the discussion unfortunately since I
> have
> >> >> not made the plan to attend in advance. However, I hope the
> list
> >> of
> >> >> potential issues above might be useful for us to discuss and
> >> >> consider.
> >> >>
> >> >> Wish we have a fruitful meeting in San Francisco.
> >> >>
> >> >> Adam
> >> >
> >> >
> >> >_______________________________________________
> >> >Audio/Video Transport Working Group
> >> >avt@ietf.org
> >> >https://www1.ietf.org/mailman/listinfo/avt
> >> _______________________________________________
> >> Audio/Video Transport Working Group
> >> avt@ietf.org
> >> https://www1.ietf.org/mailman/listinfo/avt
> >
> >_______________________________________________
> >Audio/Video Transport Working Group
> >avt@ietf.org
> >https://www1.ietf.org/mailman/listinfo/avt
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Thu Apr 24 12:45:52 2003
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From: "Mark McFarland" <mark@its.bldrdoc.gov>
To: <avt@ietf.org>
Date: Thu, 24 Apr 2003 10:43:26 -0600
Organization: The Institute for Telecommunication Sciences
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Subject: [AVT] Most efficent way to send RTP voice packets?
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Guidance is requested for the TR-8 Project 25 ISSI Messages TSB (and =
other
interface TSBs yet to be drafted) about a preferred use of 3 voice =
messages
per RTP message:
=20
The draft states now that 3 voice messages will be used per RTP message, =
but
this has been chosen rather arbitrarly. Does anybody have any experience =
in
this area as to how many voice messages should be sent per RTP messages? =
 Or
even some "testing" strageties so I can look into this myself (as the =
resuts
will most likely depend upon the available bandwidth).
=20
Thanks!
mark
=20
Mark McFarland=20
Electronics Engineer=20
United States Department of Commerce=20
National Telecommunications & Information Administration=20
The Institute for Telecommunication Sciences=20
Boulder, CO=20

mailing address:=20
ITS-P=20
325 Broadway=20
Boulder, CO 80305=20

phone: 303/497-4132=20

=20

------=_NextPart_000_0061_01C30A4E.56815830
Content-Type: text/html;
	charset="US-ASCII"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1141" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D833562016-24042003><FONT face=3D"Comic Sans =
MS">Guidance is=20
requested for the TR-8 Project 25 ISSI Messages TSB (and other interface =
TSBs=20
yet to be drafted) about a preferred use of 3 voice messages per RTP=20
message:</FONT></SPAN></DIV>
<DIV><SPAN class=3D833562016-24042003><FONT=20
face=3D"Comic Sans MS"></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D833562016-24042003><FONT face=3D"Comic Sans MS">The =
draft states=20
now that 3 voice messages will be used&nbsp;per RTP message, but this =
has been=20
chosen rather arbitrarly.&nbsp;Does anybody have any experience in this =
area as=20
to how many voice messages should be sent per RTP messages?&nbsp; Or =
even some=20
"testing" strageties so I can look into this myself (as the resuts will =
most=20
likely depend upon the available bandwidth).</FONT></SPAN></DIV>
<DIV><SPAN class=3D833562016-24042003><FONT=20
face=3D"Comic Sans MS"></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D833562016-24042003><FONT=20
face=3D"Comic Sans MS">Thanks!</FONT></SPAN></DIV>
<DIV><SPAN class=3D833562016-24042003><FONT=20
face=3D"Comic Sans MS">mark</FONT></SPAN></DIV>
<DIV><SPAN class=3D833562016-24042003><FONT=20
face=3D"Comic Sans MS"></FONT></SPAN>&nbsp;</DIV>
<DIV align=3Dleft>Mark McFarland <BR>Electronics Engineer <BR>United =
States=20
Department of Commerce <BR>National Telecommunications &amp; Information =

Administration <BR>The Institute for Telecommunication Sciences =
<BR>Boulder, CO=20
<BR><BR>mailing address: <BR>ITS-P <BR>325 Broadway <BR>Boulder, CO =
80305=20
<BR><BR>phone: 303/497-4132 <BR></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_0061_01C30A4E.56815830--

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Thu Apr 24 14:20:42 2003
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Date: Thu, 24 Apr 2003 13:21:40 -0500
From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: [AVT] parameter registration with IANA
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Hi,

I wonder whether all the media parameters defined in an AVT RFC are
registered with IANA or just some of them. 

For instance, in RFC3267 (AMR and AMR-WB formats), there are quite a few
parameters defined in section 8 (e.g., octet-align, mode-set, ...), but
I only see one of them (maxptime) defined at the IANA page
<http://www.iana.org/assignments/sdp-parameters>. Am I looking at the
wrong page? I'd appreciate it if some one could explain how this IANA
thing works here to me.

thanks,
-Qiaobing
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Thu Apr 24 16:05:24 2003
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To: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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Subject: Re: [AVT] parameter registration with IANA 
In-Reply-To: Your message of "Thu, 24 Apr 2003 13:21:40 CDT."
             <3EA82B34.6BECF014@motorola.com> 
Date: Thu, 24 Apr 2003 16:03:21 -0400
From: Colin Perkins <csp@csperkins.org>
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--> Qiaobing Xie writes:
>Hi,
>
>I wonder whether all the media parameters defined in an AVT RFC are
>registered with IANA or just some of them. 
>
>For instance, in RFC3267 (AMR and AMR-WB formats), there are quite a few
>parameters defined in section 8 (e.g., octet-align, mode-set, ...), but
>I only see one of them (maxptime) defined at the IANA page
><http://www.iana.org/assignments/sdp-parameters>. Am I looking at the
>wrong page? I'd appreciate it if some one could explain how this IANA
>thing works here to me.

It's a little confusing, certainly. The parameters from RFC 3267 that are
on the IANA page are SDP attributes (e.g. a=maxptime:). The parameters that
go under a=fmtp: are not listed (they should be included with the MIME type
registration for "audio/amr", which is currently just references the RFC).

Colin
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Qiaobing Xie <Qiaobing.Xie@motorola.com>
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> ... The parameters that
> go under a=fmtp: are not listed (they should be included with the MIME type
> registration for "audio/amr", which is currently just references the RFC).

Colin,

So, this means that IANA will not collectively manage any format
specific parameters.

Thanks and regards,
-Qiaobing

> 
> Colin
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--> Qiaobing Xie writes:
>> ... The parameters that
>> go under a=fmtp: are not listed (they should be included with the MIME type
>> registration for "audio/amr", which is currently just references the RFC).
>
>So, this means that IANA will not collectively manage any format
>specific parameters.

They're specific to the payload format, and are registered as part of the
MIME registration for the payload format. There is no need for collective
management.

Colin
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Hi Steve

I think I have located two errors in the draft. They are located in 
section 5, the example:

1. Isn't the values for RR and RS wrong. I believe they should change 
place. If one reads the text below the values in the audio media block 
should be equal to the default values. That would mean 1.25 % of session 
to RS, which is 800. While RR should be 2400 or equal to 3.75%.

2. This does also concern the text below the example. In my 
interpretation there is difference between including both RR and RS and 
leaving them out while you have the bit-rates equal to the default 3.75 
and 1.25 %. When you have the RR and RS in place the 1/4 active sender 
rule is not in place. While when you don't have them it will be in 
place. Therefore there will be a significant difference in how you 
assign the bit-rate. Also I think that one could clarify this difference.

I hope there is possibility to fix these errors before the publication.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se


_______________________________________________
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https://www1.ietf.org/mailman/listinfo/avt



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> At 2:04 PM -0400 4/22/03, Colin Perkins wrote:

> >  Are there minutes of this meeting available? I'd like to understand the
> >  rationale behind the choice (especially the choice of changing the MIME
> >  type, since "audio/qcp" seems appropriate).

> Regarding the MIME type, "audio/qcp" was the initial suggested MIME
> type, but there was pushback based on the principle that MIME types
> should describe the content, not the format.  In general, neglect of
> this principle reduces interoperability, since it then means that an
> implementation that supports a certain MIME type may not be able to
> handle content received in that MIME type.

Yes, although there's sometimes a fine line between content differences and the
use of different content options. The standard example is PostScript, which has
numerous optional facilities, so many that describing them all in  either
parameters or in subtype names is clearly impractical.

> So, in accordance with
> the principle, it makes sense to use "audio/qcelp" for QCELP 13K data
> in this format, "audio/evrc-qcp" for EVRC data stored in the QCP
> format, and "audio/smv-qcp" for SMV data in the QCP format.

> I suppose "audio/qcelp" plus "audio/qcp" with a mandatory parameter
> for the codec included, which could only be SMV or EVRC, not QCELP,
> could also work.  I've added Ned Freed to the distribution list to
> get his views, as the IANA-designated MIME expert.

(FYI, I'm on the AVT list already...)

I tend to lean towards separate types myself, but either way will work.

				Ned
_______________________________________________
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Magnus

> I think I have located two errors in the draft. They are located in
> section 5, the example:
>
> 1. Isn't the values for RR and RS wrong. I believe they should change
> place. If one reads the text below the values in the audio media block
> should be equal to the default values. That would mean 1.25 % of session
> to RS, which is 800. While RR should be 2400 or equal to 3.75%.

You are right, these are swapped.  Thanks for finding this error.

> 2. This does also concern the text below the example. In my
> interpretation there is difference between including both RR and RS and
> leaving them out while you have the bit-rates equal to the default 3.75
> and 1.25 %. When you have the RR and RS in place the 1/4 active sender
> rule is not in place. While when you don't have them it will be in
> place. Therefore there will be a significant difference in how you
> assign the bit-rate. Also I think that one could clarify this difference.

I'm not sure what point you are making.  Is it that the 1/4 active
sender rule says that the active senders get _at least_ 1/4 of the
RTCP bandwidth but that if RS is specified the active senders would
never get more than that?

It was my intention that the behavior be the same when RS and RR are
specified, or even if just one is specified.  These bw specifiers just
set the two input values to the algorithm.  However, I see that the
document does not make this clear.

Do you think there is a reason that the behavior I inteded will not
work?

> I hope there is possibility to fix these errors before the publication.

I believe so.

                                                        -- Steve

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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I've been reading through the RFC 1889 and draft-ietf-avt-rtp-new-12.txt
and I still can't quite figure out a few things about the SSRC collision
resolution algorithm.

I will first start with the simple case where there is one sender in the
RTP session. In this case there is only one sending SSRC and the "send a
BYE for the old SSRC and pick a new SSRC" is pretty clear and easy for the
client to figure out what is happening. Does the sender keep the same
sequence number and timestamp offset across this switch? My current
assumption is that it does, but I could not find that explicitly stated
anywhere. Is that assumption correct? I hope so since it might cause all
sorts of problems for certain payload types like interleaved AMR.

The second case that I am confused about is when there are more than one
sender in an RTP session. For this example I'll use the case where we are
doing SSRC multiplexing of a stream and a retransmission stream. In this
case both SSRCs are associated with the same CNAME. In the highly unlikely
chance that both SSRCs conflict with existing SSRCs in the session, how
will the client resolve the proper transition of both of these SSRCs to
new SSRCs? The only way that I can think of to resolve this would be to
use the payload type. Is that how the resolution should happen?

My third case is similar to my second case. I have seen examples in the
archives where people talk about using SSRC multiplexing for a lecture
type environment where there is a SSRC assigned to a lecturer, and a
SSRC assigned to a question microphone. In this scenario, the two SSRCs
likely have the same payload type so the resolution mechanism from case 2
would not work. How is the client supposed to resolve the transitions in
this case? Do all of these sources have the same CNAME? If not, then the
resolution can be done based on the CNAME. I was under the impression
that the CNAME would be the same for all the SSRCs in this scenario
though.

One final question about SSRC multiplexing in general. Is it assumed that
all SSRCs associated with the same CNAME should be mixed together on the
client side?

Any clarification people can provide on these questions would be great.

Thanks,

Aaron
_______________________________________________
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Subject: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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Hi Steve,

Stephen Casner wrote:

>>2. This does also concern the text below the example. In my
>>interpretation there is difference between including both RR and RS and
>>leaving them out while you have the bit-rates equal to the default 3.75
>>and 1.25 %. When you have the RR and RS in place the 1/4 active sender
>>rule is not in place. While when you don't have them it will be in
>>place. Therefore there will be a significant difference in how you
>>assign the bit-rate. Also I think that one could clarify this difference.
>>    
>>
>
>I'm not sure what point you are making.  Is it that the 1/4 active
>sender rule says that the active senders get _at least_ 1/4 of the
>RTCP bandwidth but that if RS is specified the active senders would
>never get more than that?
>
>It was my intention that the behavior be the same when RS and RR are
>specified, or even if just one is specified.  These bw specifiers just
>set the two input values to the algorithm.  However, I see that the
>document does not make this clear.
>
>Do you think there is a reason that the behavior I inteded will not
>work?
>  
>

The interpretation of your draft that I and other in 3GPP SA4 have made, 
is that when you use both RR and RS you don't use the 1/4 active sender 
rule. If both the values are specified one simple plugs them in as the 
values for each group. I think there are a good reason for doing this 
interpretation.

1. Unless one disables the 1/4 rule it is impossible to turn off RTCP 
receiver reports in cases where there are more then one 1/4 senders, 
which include the so important 1 to 1 case.
2. It becomes impossible to set unsymmetric RTCP bit-rates in 1-1 cases. 
This would be a problem in for example streaming cases where one might 
desire to control the bit-rate in each direction. This can be especially 
important in use cases over wireless links that need rather good control 
over bit-rate.

So based on these arguments I would suggest that you actually clarify 
the draft in this direction.

Best Regards


Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se



_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: Colin Perkins <csp@csperkins.org>
Subject: [AVT] Comments on draft-ietf-avt-rtcp-report-extns-04.txt
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Please consider the following as my last call comments on the RTCP XR
draft.  The working group last call is due to end today - if you have
comments on this draft, please send them shortly.

Colin




 - The draft needs an explicit statement that the standard RTCP timing
   rules apply, and that use of the extended report blocks will likely
   increase the reporting interval.

 - The second paragraph of section 1.1 talks about "broadcast" sessions.
   I assume that one-to-many streaming is intended, but the text should
   be clarified since broadcast IP is presumably not used.

 - First paragraph of section 2: "thus backwards compatibility with" ->
   "this backwards compatible with"

 - The packet format in section 2 has an "SSRC/CSRC" field; this should
   presumably be "SSRC" only? (diagram and text following)

 - In explanation of the DLRR field in section 4.5 mentions LSR and DLSR
   in a couple of places. Should be talking about LRR and DLRR?

 - Second paragraph of section 4.6 "The fields corresponding to unreported 
   parameters MUST be set to zero" may be clearer as "...MUST be present,
   but are set to zero"?

 - In section 4.6, the avg_jitter and avt_ttl_or_hl fields are listed as 
   being the "average" of a particular statistic. Which average is used?
   I assume the mean, but should clarify.

 - Section 4.7, second paragraph: "between RTCP end points" -> "between
   RTP end points"

 - I don't understand the first paragraph of section 4.7.2; please reword
   for clarity.

 - The example of page 28 reference loss density and discard density, but 
   these do not appear to be defined? (they're briefly mentioned earlier 
   in the section, but are not reported)

 - In section 4.7.3, the round trip delay is calculated in milliseconds,
   but the discussion talk about calculating it using RTCP LSR/DLSR. Can
   you confirm that milliseconds are intended? And, if so, note in the
   explanation that a units conversion is needed.

 - In section 4.7.3, the end system delay estimate makes the assumption 
   that the system can send and receive, and that it has a local analog
   interface to loop-back audio. Neither of these are necessarily valid.
   Accordingly, the I'd like to see the "MUST be provided" turn into a
   SHOULD, and some explanation of what to do if the data cannot be
   provided.

 - In section 4.7.4, the signal and noise levels are defined as a ratio in
   terms of a "reference digital milliwatt" and "rms talkspurt power (mW)".
   Can you clarify the definition of these terms? I wouldn't know how to
   calculate them, in the PC-based implementation I'm familiar with.

 - In section 4.7.4, the definition of the residual echo return loss is not
   clear, and seems to assume particular implmentations of gateway devices.
   Recommend separating the normative definition of the field out from the
   non-normative examples of how it might be measured in particular cases.

 - In section 4.7.5, the fields are mostly 8 bits, with only a subset of 
   values being legal. Would it be appropriate to use smaller fields in
   some cases? What should receivers do if they receive a report where a 
   field has an illegal value?

 - In section 4.7.5, the chosen values for the PLC are not contiguous: I
   can indicate a PLC that can conceal less than 3% (standard PLC), or 
   more than 10% (enhanced PLC), but nothing in the middle. Is there
   intended to be a "better-than-standard but not quite enhanced" range?
   How does this relate to the "unspecified" category?

 - In section 4.7.6, the jitter buffer rate assumes that the jitter buffer
   can cope with a step change in jitter from 30ms to 100ms. This seems to
   be somewhat aggressive?

 - In section 4.7.7, the units for the "JB max" and "JB abs max" fields are
   not specified. Also, what is reported if the "JB abs max" is larger than
   the maximum expressible value? 

 - In section 5.1, need a reference to the definitions of DIGIT and CRLF.

 - Section 6.2 should define what information needs to be included in any
   new registrations of attribute block types. There is, presumably, some
   minimum degree of specification that is required for interoperability?

 - In section 6.2, if the BT value of 255 is reserved for future extension,
   it becomes necessary to indicate how it might be extended. I expect that
   this space is large enough, that we don't have to worry about reserving
   the block.

 - End of section 6.3, talks about an ABNF definition of new parameters, in
   accordance3 with the "format-ext" definition. It needs to reference the
   location of the "format-ext" definition.

 - Section 7 needs to reference the standard RTCP reporting mechanism, when
   it says that the security considerations therein apply.

 - In section 7, I'm not sure it's appropriate to mandate that XR packets
   SHOULD NOT be encrypted or filtered. 

 - Appendix A.2: If differences are found, or if the ETSI spec is updated,
   what happens?

 - References: RFC 1700 is listed as Historic in my copy of the RFC index.

_______________________________________________
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avt@ietf.org
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                      NOSSDAV 2003: CALL FOR PARTICIPATION

                       The 13th International Workshop on
     Network and Operating Systems Support for Digital Audio and Video
                               (NOSSDAV 2003)
                        Held in conjuction with IWQOS

                           Sponsored by ACM SIGMM
                              June 1-3, 2003
                            Monterey, California

                        http://www.nossdav.org/2003

We are pleased to invite you to participate to NOSSDAV 2003, held in
conjuction with IWQOS.  Registration and local information are now
available via the NOSSDAV WWW page.  Please note that early
registration and hotel cutoff date is MAY 10, 2003.

PROGRAM SCHEDULE

Sunday June 1:
  5:00 PM Registration

Monday June 2:
  08:00	AM	 Registration
  08:00-09:00 AM Breakfast
  09:00-10:00 AM Session 1 (Keynote)
  10:00-10:30 AM Break
  10:30-Noon	 Session 2 (Video Adaptation)
  Noon-1:30 PM	 Lunch
  1:30-3:30 PM	 Session 3 (Content Management)
  3:30-4:00 PM	 Break
  4:00-5:00 PM	 Session 4 (Conferencing)

Tuesday June 3:
  08:00	AM	 Registration
  08:00-09:00 AM Breakfast
  09:00-10:30 AM Session 5 (QoS and Congestion Control)
  10:30-11:00 AM Break
  11:00-Noon	 Session 6 (User Perception in Audio)
  Noon-1:30 PM   Lunch
  1:30-3:00 PM   Session 7 (Panel)
  3:00-3:30 PM	 Break
  3:30-5:30 PM   Session 8 (Peer to Peer Systems)
________________________________________________________________________

ACCEPTED PAPERS

Session 1:  Keynote
-------------------
Why Johnny Can't Multicast:  Lessons about the Evolution of the Internet
- Mostafa H. Ammar

Session 2:  Video Adaptation
----------------------------

User-Perceived Quality-Aware Adaptive Delivery of MPEG-4 Content. 
- Liam Murphy, Nicola Cranley, Philip Perry

Receiver Initiated Just-in-Time Tree Adaptation for Rich Media Distribution. 
- Zhichen Xu, Chunqiang Tang, Sujata Banerjee, Sung-Ju Lee

Rate-Distortion Framework for Scalable Internet Video Streaming. 
- Min Dai, Dmitri Loguinov


Session 3:  Content Management
------------------------------
Design and Implementation of a Distributed Content Management System. 
- Chuck Cranor, Rick Ethington, Amit Sehgal, David Shur, Cormac Sreenan, Kobus van der Merwe

MediSyn: A Synthetic Streaming Media Service Workload Generator. 
- Wenting Tang, Yun Fu, Lucy Cherkasova, Amin Vahdat

Adaptive and Lazy Segmentation Based Proxy Caching for Streaming Media Delivery. 
- Songqing Chen, Bo Shen, Susie Wee, Xiaodong Zhang

Dynamic Program Insertion in High Quality Video over IP. 
- Taehyun Kim, Jack Brassil


Session 4:  Conferencing
------------------------
A Protocol for Reliable Decentralized Conferencing. 
- Jonathan Lennox, Henning Schulzrinne

Ubiquitous Computing Using SIP. 
- Stylianos Sidiroglou, Xiaotao Wu, Stefan Berger, Henning Schulzrinne


Session 5:  QoS and Congestion Control
--------------------------------------
Quality-Adaptive Media Streaming by Priority Drop. 
- Charles Krasic, Jonathan Walpole, Wu-chi Feng

Adjusting Forward Error Correction for TCP-Friendly Streaming MPEG. 
- Huahui Wu

Improving TCP Smoothness by Synchronized and Measurement-based Congestion Avoidance. 
- Chi Zhang


Session 6:  User Perception in Audio
------------------------------------
A Source and Channel Rate Adaptation Algorithm for AMR in VoIP Using the Emodel. 
- Johnny Matta, Christine Pepin, Khosrow Lashkari, Ravi Jain

Loss Concealment for Multi-Channel Streaming Audio. 
- Rishi Sinha, Christos Papadopoulos, Chris Kyriakakis

Session 7:  Panel
-----------------
- TBA

Session 8:  Peer to Peer Systems
--------------------------------
A Reputation System for Peer-to-Peer Networks. 
- Minaxi Gupta, Paul Judge, Mostafa Ammar

PALS: Peer to Peer Adaptive Layered Streaming. 
- Reza Rejaie, Antonio Ortega

Layered Peer-to-Peer Streaming. 
- Yi Cui, Klara Nahrstedt

Borg: a Hybrid Protocol for Scalable Application-level Multicast in Peer-to-Peer Networks. 
- Rongmei Zhang, Y. Charlie Hu
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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From: "Alan Clark" <alan@telchemy.com>
To: "'Colin Perkins'" <csp@csperkins.org>,
        "Timur Friedman \(E-mail\)" <timur.friedman@lip6.fr>
Cc: <avt@ietf.org>
Subject: RE: [AVT] Comments on draft-ietf-avt-rtcp-report-extns-04.txt
Date: Mon, 28 Apr 2003 19:28:05 -0400
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Colin

Thanks for the comments - I've responded on those related to VoIP

Regards

Alan Clark


 - The draft needs an explicit statement that the standard RTCP timing
   rules apply, and that use of the extended report blocks will likely
   increase the reporting interval.

[Alan] Suggest adding to Section 1.1 paragraph 1
"The XR report blocks defined in this document MUST be sent within the
defined guidelines for RTCP reporting intervals.  In certain applications
the use of RTCP XR payloads may mean that less frequent reports are needed
in order to provide a sufficient degree of feedback on performance."

 - The second paragraph of section 1.1 talks about "broadcast" sessions.
   I assume that one-to-many streaming is intended, but the text should
   be clarified since broadcast IP is presumably not used.

[Alan] Suggest either
(i) replacing "broadcast" with "one-to-many streaming"
(ii) delete broadcast as multicast will cover most cases

 - First paragraph of section 2: "thus backwards compatibility with" ->
   "this backwards compatible with"

 - The packet format in section 2 has an "SSRC/CSRC" field; this should
   presumably be "SSRC" only? (diagram and text following)

 - In explanation of the DLRR field in section 4.5 mentions LSR and DLSR
   in a couple of places. Should be talking about LRR and DLRR?

 - Second paragraph of section 4.6 "The fields corresponding to unreported
   parameters MUST be set to zero" may be clearer as "...MUST be present,
   but are set to zero"?

 - In section 4.6, the avg_jitter and avt_ttl_or_hl fields are listed as
   being the "average" of a particular statistic. Which average is used?
   I assume the mean, but should clarify.

 - Section 4.7, second paragraph: "between RTCP end points" -> "between
   RTP end points"

[Alan] Agreed

 - I don't understand the first paragraph of section 4.7.2; please reword
   for clarity.

[Alan] "A burst is a period during which a high proportion of packets are
either lost in transit or discarded due to late arrival.  A burst is defined
as a longest sequence that (a) starts with a lost or discarded packet, (b)
does not contain any occurrences of Gmin or more consecutive received (and
not discarded) packets, and (c) ends with a lost or discarded packet."

 - The example of page 28 reference loss density and discard density, but
   these do not appear to be defined? (they're briefly mentioned earlier
   in the section, but are not reported)

[Alan] these should be "loss rate" and "discard rate", both of which are
defined in section 4.7.1

 - In section 4.7.3, the round trip delay is calculated in milliseconds,
   but the discussion talk about calculating it using RTCP LSR/DLSR. Can
   you confirm that milliseconds are intended? And, if so, note in the
   explanation that a units conversion is needed.

[Alan] The use of milliseconds was intentional.
Suggest changing second sentence of "round trip delay" definition to:
"This value MAY be measured using RTCP or the DLRR method defined in Section
4.5 of this document, in which case it is necessary to convert the units of
measurement from NTP to milliseconds, or other approaches."

 - In section 4.7.3, the end system delay estimate makes the assumption
   that the system can send and receive, and that it has a local analog
   interface to loop-back audio. Neither of these are necessarily valid.
   Accordingly, the I'd like to see the "MUST be provided" turn into a
   SHOULD, and some explanation of what to do if the data cannot be
   provided.

[Alan] This was defined as an estimated rather than measured delay, and can
typically be determined by adding known framing and encode/decode delays -
it was not intended that it be measured literally. We can change the wording
to clarify.
"The most recently estimated end system delay, expressed in milliseconds.
End system delay is defined as the sum of the total sample accumulation and
encoding delay associated with the sending direction and the jitter buffer,
decoding and playout buffer delay associated with the receiving direction;
this delay MAY be estimated or measured.  This value MUST be provided in all
VoIP metrics reports."

 - In section 4.7.4, the signal and noise levels are defined as a ratio in
   terms of a "reference digital milliwatt" and "rms talkspurt power (mW)".
   Can you clarify the definition of these terms? I wouldn't know how to
   calculate them, in the PC-based implementation I'm familiar with.

[Alan] The term "digital milliwatt" is widely used in both PCM and VoIP
systems to denote the digital representation of a 0 dBm0 signal.  I've not
yet found the official definition of it, although it may originate in an
IEEE standard - we can substitute a 0dBm0 reference signal.
Suggest:-

for "signal level" and "noise level" change "digital milliwatt" to "0 dBm0
reference [18]"

add reference [18] TIA/EIA-810-A Transmission Requirements for Narrowband
Voice over IP and Voice over PCM Digital Wireline Telephones, December 2000


 - In section 4.7.4, the definition of the residual echo return loss is not
   clear, and seems to assume particular implmentations of gateway devices.
   Recommend separating the normative definition of the field out from the
   non-normative examples of how it might be measured in particular cases.

[Alan] This should be typical of VoIP gateways with line echo cancellers and
isn't specific to any implementations.  I've checked with a number of echo
canceller software developers and the parameters are typically available as
ERL and ERLE in dB or in some format that can easily be converted.
Suggest clarifying the wording to:

The residual echo return loss is defined as the sum of the measured echo
return loss (ERL) and the echo return loss enhancement (ERLE) of the echo
canceller (if present) expressed in dB as a signed integer in 2's complement
form.

The residual echo return loss relates to the effective echo level that a
user would hear.  The amount of echo that can occur in analog loops can be
significant, and therefore echo cancellers are typically used to try and
eliminate or reduce the audible echo. The amount of echo is typically
described as a "echo return loss" (i.e. ratio between the transmitted signal
level and the received echo level) expressed in dB.  The effectiveness of an
echo canceller is normally described in terms of "echo return loss
enhancement", i.e. an echo canceller increases the effective echo return
loss and makes the audible echo smaller.  Hence the residual echo return
loss (RERL) is the sum of the echo return loss (ERL) and the echo return
loss enhancement (ERLE) expressed in dB.


 - In section 4.7.5, the fields are mostly 8 bits, with only a subset of
   values being legal. Would it be appropriate to use smaller fields in
   some cases? What should receivers do if they receive a report where a
   field has an illegal value?

[Alan] I suggest keeping these as 8 bit, otherwise we would need to handle 6
bit values.  We could change the "A value of" sentences to read

"A value of 127 indicates that this parameter is unavailable.  Values other
than 127 and the valid range defined above MUST not be sent and MUST be
ignored by the receiving system."

 - In section 4.7.5, the chosen values for the PLC are not contiguous: I
   can indicate a PLC that can conceal less than 3% (standard PLC), or
   more than 10% (enhanced PLC), but nothing in the middle. Is there
   intended to be a "better-than-standard but not quite enhanced" range?
   How does this relate to the "unspecified" category?

[Alan] There is a continuum of possibilities in terms of PLC effectiveness.
"Standard" in this case refers to algorithms that use some form of last
packet replay with attenuation and filtering/smoothing.  "Enhanced" refers
to algorithms that use some form of error correction(sometimes introducing
additional delay or bandwidth overhead). "Unspecified" simply means that the
CODEC details are outside the domain of knowledge of the RTCP XR reporting
instance - they are typically internal to a DSP.
I think the standard and enhanced definitions cover what is used in the
industry today, and generally implementors could figure out which to use.

 - In section 4.7.6, the jitter buffer rate assumes that the jitter buffer
   can cope with a step change in jitter from 30ms to 100ms. This seems to
   be somewhat aggressive?

[Alan] Let's hope people aren't using 10BaseT and hubs for VoIP!! :-)  Many
adaptive jitter buffers max out at 200-400mS although they would typically
operate in the 20-50mS range.  As the description suggests, this is intended
to be an easily measured parameter that provides a guide to the JB ramp
time, in reality jitter/PDV would not behave quite this way.

 - In section 4.7.7, the units for the "JB max" and "JB abs max" fields are
   not specified. Also, what is reported if the "JB abs max" is larger than
   the maximum expressible value?

[Alan] Suggested changes

jitter buffer maximum delay...
This is the current maximum jitter buffer delay in milliseconds which
corresponds to the earliest arriving packet that would not be discarded.  In
simple queue.......

jitter buffer absolute maximum delay...
This is the absolute maximum delay in milliseconds that the adaptive jitter
buffer can reach under worst case conditions.  If this value exceeds 65535
milliseconds then it shall be limited to 65535. This parameter MUST....

 - In section 5.1, need a reference to the definitions of DIGIT and CRLF.

 - Section 6.2 should define what information needs to be included in any
   new registrations of attribute block types. There is, presumably, some
   minimum degree of specification that is required for interoperability?

 - In section 6.2, if the BT value of 255 is reserved for future extension,
   it becomes necessary to indicate how it might be extended. I expect that
   this space is large enough, that we don't have to worry about reserving
   the block.

 - End of section 6.3, talks about an ABNF definition of new parameters, in
   accordance3 with the "format-ext" definition. It needs to reference the
   location of the "format-ext" definition.

 - Section 7 needs to reference the standard RTCP reporting mechanism, when
   it says that the security considerations therein apply.

 - In section 7, I'm not sure it's appropriate to mandate that XR packets
   SHOULD NOT be encrypted or filtered.

[Alan] Isn't SHOULD an advisory qualifier, unlike MUST?  The goal is to
encourage implementors to turn encryption of RTCP off when sending RTCP XR
reports as this is desirable from a network management perspective.

 - Appendix A.2: If differences are found, or if the ETSI spec is updated,
   what happens?

[Alan] We are refering to a specific dated release of the ETSI spec.  The
VoIP metrics payload is actually substantially more comprehensive than the
ETSI spec, which only relates to methods for calculating R factors.
We should make the reference at the end of the second paragraph in 4.7 "(as
described by the E Model defined in [6,3]) and.."


 - References: RFC 1700 is listed as Historic in my copy of the RFC index.

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt

_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



