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From: Colin Perkins <csp@csperkins.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Cc: alan@telchemy.com, avt@ietf.org
Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
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Folks,

Would Magnus' proposal satisfy all involved in this discussion?
Colin


--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Hi,
> 
> Looking at this discussion it seems that one can make two resolutions.
> 
> 1. Do not mandate calculation of all fields in report block. I would 
> think that a SHOULD calculate all field is a good level. Text motivating 
> that all fields should be calculated is good.
> 
> 2. Keep the structure as is, define for each field a "NOT VALID" value.
> 
> I think this is reasonable compromise. It keeps the VoIP report block 
> simple to process and at the same time a implementor that is unable to 
> provide a measurement does not need to.
> 
> Best Regards
> 
> Magnus
> 
> Alan Clark wrote:
> > Dave,
> > 
> > RTCP XR can be extremely useful in supporting problem detection, call
> > admission control etc. as it provides a fast method for a probe,
> > analyzer, session controller... to get call quality related information
> > from the endpoint.  In general the data available in RTCP XR should not
> > represent a security concern - unlike SDES - and hence (I think) there
> > is no advantage to applying encryption and considerable disadvantage.
> > 
> > In an Enterprise network using VPN for inter-site connections you could
> > consider using SRTP for RTP payloads and rely on the VPN tunnel to
> > protect RTCP SDES information, which is a good compromise between
> > management and security.  In some other applications, notably IP
> > Centrex, the service provider is already handling signaling - RTCP XR
> > gives them feedback on call quality from the customer desktop and
> > alerts them to customer LAN problems that affect call quality.
> > 
> > Regards
> > 
> > Alan
> > 
> > 
> > 
> > 
> > -----Original Message-----
> > From: David R. Oran [mailto:oran@cisco.com]
> > Sent: Sunday, May 25, 2003 5:46 PM
> > To: Alan Clark; Paul E. Jones; 'Siddiqui, Anwar A (Anwar)'; 'Colin
> > Perkins'; 'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> > Cc: avt@ietf.org
> > Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> > 
> > 
> > --On Sunday, May 25, 2003 8:11 AM -0400 Alan Clark <alan@telchemy.com>
> > wrote:
> > 
> > 
> >>Paul
> >>
> >>If you bear in mind that RTCP XR is likely to be used to exchange
> >>information between endpoints through the media path then there are
> >>several implementation considerations.
> >>
> >>(i) It is quite likely that intermediate systems such as Ethernet
> >>switches, edge routers and shapers may scan RTCP XR payloads.  A number
> >>of such systems already do this with RTCP SR/RR reports, either for QoS
> >>monitoring or for active control.  With switch speeds getting to 10
> >>gigabit then it is important to keep parsing complexity very low.
> >>
> > 
> > Really? Boy, I'm glad I'm using SRTP and encrypting my RTCP packets.
> > Put another way, I don't find snooping to be a valid argument in favor
> > of fast parsing.
> > That said, I do prefer to have an encoding which uses "value not valid"
> > semantics rather than "value not present", because, like is done with
> > IEEE floating point, it has much better robustness in the face of
> > cascaded computations that may use inappropriate arithmetic. In fact, I
> > think that's where the real savings is - in not having to scatter your
> > data reduction for stats with a zillion if tests and multiple code
> > patsh based on what dat ayou have. You just do the computation and
> > error propagation will take care of ensuring that invalid stuff is
> > marked as such.
> > 
> > 
> >>(ii) High density gateways are pushing the density of CODECs per DSP
> >but>also pushing RTP packet processing into the DSP.  This places severe
> >>constraints on the available program and data memory space - again a
> >>strong incentive to keep things as simple as possible.
> >>
> > 
> > Simplicity does resonate as a good argument here.
> > 
> > Dave.
> > 
> > 
> >>Regards
> >>
> >>Alan
> >>
> >>-----Original Message-----
> >>From: Paul E. Jones [mailto:paulej@packetizer.com]
> >>Sent: Saturday, May 24, 2003 10:40 AM
> >>To: alan@telchemy.com; 'Siddiqui, Anwar A (Anwar)'; 'Colin Perkins';
> >>'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> >>Cc: avt@ietf.org
> >>Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> >>
> >>
> >>Alan,
> >>
> >>I still think we should allow every field to be optional.  I recognize
> >the>desire to encourage reporting a complete set of stats, but I would
> >tend to>believe that there will be environments wherein not all stats
> >are>necessary.
> >>
> >>Now, not to sound that I'm in disagreement with Anwar, but I think it
> >>might be useful to allow one to indicate that a particular parameter is
> >>not measured by populating the field with some default value.  This
> >>default value would not indicate a "valid", but be an indicator of the
> >>fact that the field is not present.  This would address your concern
> >that>the fields and reporting blocks are fixed in size.
> >>
> >>I also realize that some time will be wasted to determine that a
> >>measurement is not present, but the same amount of time will be
> >consumed>by a system to extract a valid value today.  So, the system
> >performance>is the same, albeit with potentially less useful data.
> >>
> >>Paul
> >>
> >>----- Original Message -----
> >>From: "Alan Clark" <alan@telchemy.com>
> >>To: "'Siddiqui, Anwar A (Anwar)'" <anwars@avaya.com>; "'Colin Perkins'"
> >><csp@csperkins.org>; "'Timur Friedman'" <timur.friedman@lip6.fr>;
> >>"'Stephen Casner'" <casner@acm.org>; "'Ramon Caceres'"
> >><ramon@shieldip.com> Cc: <avt@ietf.org>
> >>Sent: Thursday, May 22, 2003 10:14 AM
> >>Subject: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> >>
> >>
> >>
> >>>Anwar
> >>>
> >>>This was brought up several weeks ago on the mailing list.  This
> >approach>>was taken deliberately for several reasons:-
> >>>
> >>>(i) To encourage implementors to move towards implementing the
> >complete>
> >>set
> >>
> >>>of metrics.  I've socialized this data set with a variety of service
> >>>providers, enterprise network managers, equipment manufacturers and
> >there>>seems to be general agreement that this metrics set provides a
> >good>
> >>"minimum
> >>
> >>>set" for diagnosing VoIP performance problems.
> >>>
> >>>(ii) There is overhead associated with identifying whether fields are
> >or>
> >>are
> >>
> >>>not present (see Colin Perkins comment on this issue several weeks
> >ago).>>
> >>>I am aware of several major equipment manufacturers and DSP software
> >>
> >>vendors
> >>
> >>>that intend to fully implement the RTCP XR VoIP metrics, and several
> >tier>>one carriers that are currently specifying that this metric set be
> >used>
> >>for
> >>
> >>>VoIP service management. There is also considerable interest from
> >other>>standards groups in this work.
> >>>
> >>>Regards
> >>>
> >>>Alan Clark
> >>>Telchemy
> >>>
> >>>-----Original Message-----
> >>>From: Siddiqui, Anwar A (Anwar) [mailto:anwars@avaya.com]
> >>>Sent: Thursday, May 22, 2003 9:23 AM
> >>>To: Colin Perkins; Timur Friedman; Stephen Casner; alan@telchemy.com;
> >>>Ramon Caceres
> >>>Cc: avt@ietf.org
> >>>Subject: Re: draft-ietf-avt-rtcp-report-extns-06.txt
> >>>
> >>>
> >>>Timur, Alan
> >>>
> >>>I did mention this during last IETF session. And I believe few other
> >>
> >>people
> >>
> >>>also have similar concerns. So here is the issue I am requesting this
> >be>>reconsidered and if possible be avoided.
> >>>
> >>>------------------------------
> >>>Is there anyway that we can avoid hard coding VoIP metrics like it is
> >>>identified in this draft. Its not the usefulness of the metrics that I
> >am>>having difficulties with. I actually believe they are useful
> >parameters>>to be included in RTCP. But its just that its hardcoded
> >right now>>....and if
> >>
> >>an
> >>
> >>>RTP source does not have parameters to report it will keep sending
> >these>>default hard coded values! It is this kind of hardcoding in SR/RR
> >reports>>that makes us do this XR reports now. So I am requesting if it
> >possible>
> >>such
> >>
> >>>hard coding of VoIP metrics be avoided.
> >>>-------------------------------
> >>>
> >>>Thanks in advance
> >>>
> >>>Anwar
> >>>
> >>>
> >>>
> >>>----- Original Message -----
> >>>From: "Colin Perkins" <csp@csperkins.org>
> >>>To: "Timur Friedman" <timur.friedman@lip6.fr>
> >>>Cc: <avt@ietf.org>; "Stephen Casner" <casner@acm.org>; "Alan Clark"
> >>><alan@telchemy.com>; "Ramon Caceres" <ramon@shieldip.com>; "Kevin C.
> >>>Almeroth" <almeroth@cs.ucsb.edu>; "Robert Cole" <rgcole@att.com>;
> >"Nick>>Duffield" <duffield@research.att.com>; "Kaynam Hedayat"
> >>><khedayat@brixnet.com>; "Kamil Sarac" <ksarac@utdallas.edu>; "Magnus
> >>>Westerlund" <magnus.westerlund@era.ericsson.se>
> >>>Sent: Monday, May 19, 2003 8:05 PM
> >>>Subject: [AVT] Re: draft-ietf-avt-rtcp-report-extns-06.txt
> >>>
> >>>
> >>>
> >>>>Thanks. Unless there are objections, we intend to request that the
> >IESG>>>consider this version for publication as a proposed standard.
> >>>>
> >>>>Colin
> >>>>
> >>>>
> >>>>--> Timur Friedman writes:
> >>>>
> >>>>>Hello,
> >>>>>
> >>>>>Please find attached the latest version of the RTP Control Protocol
> >>>>
> >>>Extended
> >>>
> >>>>>Reports (RTCP XR) draft, draft-ietf-avt-rtcp-report-extns-06.txt.
> >>>>>
> >>>>>This draft was produced in response to AVT working group last call
> >>>>
> >>>comments.
> >>>
> >>>>>In addition to the changes discussed online on the AVT list, some
> >>>>
> >>>additional
> >>>
> >>>>>detail has been added to the SDP signaling section, to allow
> >>>>>individual
> >>>>
> >>>flags
> >>>
> >>>>>in the Statistics Summary report blocks to be signaled.  Also,
> >>>>>default
> >>>>
> >>>values
> >>>
> >>>>>have been specified for the VoIP metrics section in places where
> >>>>>there
> >>>>
> >>>might
> >>>
> >>>>>have been some confusion about what to do if no packets had been
> >>>>
> >>>received.
> >>>
> >>>>>And the time frame over which some of the VoIP fields are calculated
> >>>>
> >>has
> >>
> >>>>>also been specified.
> >>>>>
> >>>>>A PDF version of this draft is available at the following URL:
> >>>>>
> >>>>
> >>>http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-r
> >epo>>rt- extns-06.pdf
> >>>
> >>>>>Regards,
> >>>>>
> >>>>>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
> > 
> > 
> > 
> > 
> > ------------------------
> > David R. Oran
> > Cisco Systems
> > 7 Ladyslipper Lane
> > Acton, MA 01720
> > Office: +1 978 264 2048
> > VoIP: +1 408 571 4576
> > Email: oran@cisco.com
> > 
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> > 
> 
> 
> -- 
> 
> 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@ericsson.com
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
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>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Cc: <avt@ietf.org>
Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
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Colin

I agree with the approach.  I think that we have tried to achieve this with
"undefined" values for most fields - the exceptions were those that related
to packet loss .. why would you want to send a report that didn't at least
contain loss metrics??

Regards

Alan


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Sunday, June 01, 2003 11:03 AM
To: Magnus Westerlund
Cc: alan@telchemy.com; avt@ietf.org
Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt


Folks,

Would Magnus' proposal satisfy all involved in this discussion?
Colin


--> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> Hi,
>
> Looking at this discussion it seems that one can make two resolutions.
>
> 1. Do not mandate calculation of all fields in report block. I would
> think that a SHOULD calculate all field is a good level. Text motivating
> that all fields should be calculated is good.
>
> 2. Keep the structure as is, define for each field a "NOT VALID" value.
>
> I think this is reasonable compromise. It keeps the VoIP report block
> simple to process and at the same time a implementor that is unable to
> provide a measurement does not need to.
>
> Best Regards
>
> Magnus
>
> Alan Clark wrote:
> > Dave,
> >
> > RTCP XR can be extremely useful in supporting problem detection, call
> > admission control etc. as it provides a fast method for a probe,
> > analyzer, session controller... to get call quality related information
> > from the endpoint.  In general the data available in RTCP XR should not
> > represent a security concern - unlike SDES - and hence (I think) there
> > is no advantage to applying encryption and considerable disadvantage.
> >
> > In an Enterprise network using VPN for inter-site connections you could
> > consider using SRTP for RTP payloads and rely on the VPN tunnel to
> > protect RTCP SDES information, which is a good compromise between
> > management and security.  In some other applications, notably IP
> > Centrex, the service provider is already handling signaling - RTCP XR
> > gives them feedback on call quality from the customer desktop and
> > alerts them to customer LAN problems that affect call quality.
> >
> > Regards
> >
> > Alan
> >
> >
> >
> >
> > -----Original Message-----
> > From: David R. Oran [mailto:oran@cisco.com]
> > Sent: Sunday, May 25, 2003 5:46 PM
> > To: Alan Clark; Paul E. Jones; 'Siddiqui, Anwar A (Anwar)'; 'Colin
> > Perkins'; 'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> > Cc: avt@ietf.org
> > Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> >
> >
> > --On Sunday, May 25, 2003 8:11 AM -0400 Alan Clark <alan@telchemy.com>
> > wrote:
> >
> >
> >>Paul
> >>
> >>If you bear in mind that RTCP XR is likely to be used to exchange
> >>information between endpoints through the media path then there are
> >>several implementation considerations.
> >>
> >>(i) It is quite likely that intermediate systems such as Ethernet
> >>switches, edge routers and shapers may scan RTCP XR payloads.  A number
> >>of such systems already do this with RTCP SR/RR reports, either for QoS
> >>monitoring or for active control.  With switch speeds getting to 10
> >>gigabit then it is important to keep parsing complexity very low.
> >>
> >
> > Really? Boy, I'm glad I'm using SRTP and encrypting my RTCP packets.
> > Put another way, I don't find snooping to be a valid argument in favor
> > of fast parsing.
> > That said, I do prefer to have an encoding which uses "value not valid"
> > semantics rather than "value not present", because, like is done with
> > IEEE floating point, it has much better robustness in the face of
> > cascaded computations that may use inappropriate arithmetic. In fact, I
> > think that's where the real savings is - in not having to scatter your
> > data reduction for stats with a zillion if tests and multiple code
> > patsh based on what dat ayou have. You just do the computation and
> > error propagation will take care of ensuring that invalid stuff is
> > marked as such.
> >
> >
> >>(ii) High density gateways are pushing the density of CODECs per DSP
> >but>also pushing RTP packet processing into the DSP.  This places severe
> >>constraints on the available program and data memory space - again a
> >>strong incentive to keep things as simple as possible.
> >>
> >
> > Simplicity does resonate as a good argument here.
> >
> > Dave.
> >
> >
> >>Regards
> >>
> >>Alan
> >>
> >>-----Original Message-----
> >>From: Paul E. Jones [mailto:paulej@packetizer.com]
> >>Sent: Saturday, May 24, 2003 10:40 AM
> >>To: alan@telchemy.com; 'Siddiqui, Anwar A (Anwar)'; 'Colin Perkins';
> >>'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> >>Cc: avt@ietf.org
> >>Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> >>
> >>
> >>Alan,
> >>
> >>I still think we should allow every field to be optional.  I recognize
> >the>desire to encourage reporting a complete set of stats, but I would
> >tend to>believe that there will be environments wherein not all stats
> >are>necessary.
> >>
> >>Now, not to sound that I'm in disagreement with Anwar, but I think it
> >>might be useful to allow one to indicate that a particular parameter is
> >>not measured by populating the field with some default value.  This
> >>default value would not indicate a "valid", but be an indicator of the
> >>fact that the field is not present.  This would address your concern
> >that>the fields and reporting blocks are fixed in size.
> >>
> >>I also realize that some time will be wasted to determine that a
> >>measurement is not present, but the same amount of time will be
> >consumed>by a system to extract a valid value today.  So, the system
> >performance>is the same, albeit with potentially less useful data.
> >>
> >>Paul
> >>
> >>----- Original Message -----
> >>From: "Alan Clark" <alan@telchemy.com>
> >>To: "'Siddiqui, Anwar A (Anwar)'" <anwars@avaya.com>; "'Colin Perkins'"
> >><csp@csperkins.org>; "'Timur Friedman'" <timur.friedman@lip6.fr>;
> >>"'Stephen Casner'" <casner@acm.org>; "'Ramon Caceres'"
> >><ramon@shieldip.com> Cc: <avt@ietf.org>
> >>Sent: Thursday, May 22, 2003 10:14 AM
> >>Subject: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> >>
> >>
> >>
> >>>Anwar
> >>>
> >>>This was brought up several weeks ago on the mailing list.  This
> >approach>>was taken deliberately for several reasons:-
> >>>
> >>>(i) To encourage implementors to move towards implementing the
> >complete>
> >>set
> >>
> >>>of metrics.  I've socialized this data set with a variety of service
> >>>providers, enterprise network managers, equipment manufacturers and
> >there>>seems to be general agreement that this metrics set provides a
> >good>
> >>"minimum
> >>
> >>>set" for diagnosing VoIP performance problems.
> >>>
> >>>(ii) There is overhead associated with identifying whether fields are
> >or>
> >>are
> >>
> >>>not present (see Colin Perkins comment on this issue several weeks
> >ago).>>
> >>>I am aware of several major equipment manufacturers and DSP software
> >>
> >>vendors
> >>
> >>>that intend to fully implement the RTCP XR VoIP metrics, and several
> >tier>>one carriers that are currently specifying that this metric set be
> >used>
> >>for
> >>
> >>>VoIP service management. There is also considerable interest from
> >other>>standards groups in this work.
> >>>
> >>>Regards
> >>>
> >>>Alan Clark
> >>>Telchemy
> >>>
> >>>-----Original Message-----
> >>>From: Siddiqui, Anwar A (Anwar) [mailto:anwars@avaya.com]
> >>>Sent: Thursday, May 22, 2003 9:23 AM
> >>>To: Colin Perkins; Timur Friedman; Stephen Casner; alan@telchemy.com;
> >>>Ramon Caceres
> >>>Cc: avt@ietf.org
> >>>Subject: Re: draft-ietf-avt-rtcp-report-extns-06.txt
> >>>
> >>>
> >>>Timur, Alan
> >>>
> >>>I did mention this during last IETF session. And I believe few other
> >>
> >>people
> >>
> >>>also have similar concerns. So here is the issue I am requesting this
> >be>>reconsidered and if possible be avoided.
> >>>
> >>>------------------------------
> >>>Is there anyway that we can avoid hard coding VoIP metrics like it is
> >>>identified in this draft. Its not the usefulness of the metrics that I
> >am>>having difficulties with. I actually believe they are useful
> >parameters>>to be included in RTCP. But its just that its hardcoded
> >right now>>....and if
> >>
> >>an
> >>
> >>>RTP source does not have parameters to report it will keep sending
> >these>>default hard coded values! It is this kind of hardcoding in SR/RR
> >reports>>that makes us do this XR reports now. So I am requesting if it
> >possible>
> >>such
> >>
> >>>hard coding of VoIP metrics be avoided.
> >>>-------------------------------
> >>>
> >>>Thanks in advance
> >>>
> >>>Anwar
> >>>
> >>>
> >>>
> >>>----- Original Message -----
> >>>From: "Colin Perkins" <csp@csperkins.org>
> >>>To: "Timur Friedman" <timur.friedman@lip6.fr>
> >>>Cc: <avt@ietf.org>; "Stephen Casner" <casner@acm.org>; "Alan Clark"
> >>><alan@telchemy.com>; "Ramon Caceres" <ramon@shieldip.com>; "Kevin C.
> >>>Almeroth" <almeroth@cs.ucsb.edu>; "Robert Cole" <rgcole@att.com>;
> >"Nick>>Duffield" <duffield@research.att.com>; "Kaynam Hedayat"
> >>><khedayat@brixnet.com>; "Kamil Sarac" <ksarac@utdallas.edu>; "Magnus
> >>>Westerlund" <magnus.westerlund@era.ericsson.se>
> >>>Sent: Monday, May 19, 2003 8:05 PM
> >>>Subject: [AVT] Re: draft-ietf-avt-rtcp-report-extns-06.txt
> >>>
> >>>
> >>>
> >>>>Thanks. Unless there are objections, we intend to request that the
> >IESG>>>consider this version for publication as a proposed standard.
> >>>>
> >>>>Colin
> >>>>
> >>>>
> >>>>--> Timur Friedman writes:
> >>>>
> >>>>>Hello,
> >>>>>
> >>>>>Please find attached the latest version of the RTP Control Protocol
> >>>>
> >>>Extended
> >>>
> >>>>>Reports (RTCP XR) draft, draft-ietf-avt-rtcp-report-extns-06.txt.
> >>>>>
> >>>>>This draft was produced in response to AVT working group last call
> >>>>
> >>>comments.
> >>>
> >>>>>In addition to the changes discussed online on the AVT list, some
> >>>>
> >>>additional
> >>>
> >>>>>detail has been added to the SDP signaling section, to allow
> >>>>>individual
> >>>>
> >>>flags
> >>>
> >>>>>in the Statistics Summary report blocks to be signaled.  Also,
> >>>>>default
> >>>>
> >>>values
> >>>
> >>>>>have been specified for the VoIP metrics section in places where
> >>>>>there
> >>>>
> >>>might
> >>>
> >>>>>have been some confusion about what to do if no packets had been
> >>>>
> >>>received.
> >>>
> >>>>>And the time frame over which some of the VoIP fields are calculated
> >>>>
> >>has
> >>
> >>>>>also been specified.
> >>>>>
> >>>>>A PDF version of this draft is available at the following URL:
> >>>>>
> >>>>
> >>>http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-r
> >epo>>rt- extns-06.pdf
> >>>
> >>>>>Regards,
> >>>>>
> >>>>>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
> >
> >
> >
> >
> > ------------------------
> > David R. Oran
> > Cisco Systems
> > 7 Ladyslipper Lane
> > Acton, MA 01720
> > Office: +1 978 264 2048
> > VoIP: +1 408 571 4576
> > Email: oran@cisco.com
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> >
>
>
> --
>
> 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@ericsson.com
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


--
Colin Perkins
csp@csperkins.org

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



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Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
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To: "Alan Clark" <alan@telchemy.com>, "Colin Perkins" <csp@csperkins.org>,
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Colin,

I agree with Alan.  We ought to consider having a minimum
level of  compliance, else don't implement.  Beyond this minimum,
we allow for vendors choosing which of the remaining they wish to
implement.

Bob Cole

> -----Original Message-----
> From:	Alan Clark [SMTP:alan@telchemy.com]
> Sent:	Sunday, June 01, 2003 4:02 PM
> To:	Colin Perkins; Magnus Westerlund
> Cc:	avt@ietf.org
> Subject:	RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> 
> Colin
> 
> I agree with the approach.  I think that we have tried to achieve this with
> "undefined" values for most fields - the exceptions were those that related
> to packet loss .. why would you want to send a report that didn't at least
> contain loss metrics??
> 
> Regards
> 
> Alan
> 
> 
> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Sunday, June 01, 2003 11:03 AM
> To: Magnus Westerlund
> Cc: alan@telchemy.com; avt@ietf.org
> Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> 
> 
> Folks,
> 
> Would Magnus' proposal satisfy all involved in this discussion?
> Colin
> 
> 
> --> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> > Hi,
> >
> > Looking at this discussion it seems that one can make two resolutions.
> >
> > 1. Do not mandate calculation of all fields in report block. I would
> > think that a SHOULD calculate all field is a good level. Text motivating
> > that all fields should be calculated is good.
> >
> > 2. Keep the structure as is, define for each field a "NOT VALID" value.
> >
> > I think this is reasonable compromise. It keeps the VoIP report block
> > simple to process and at the same time a implementor that is unable to
> > provide a measurement does not need to.
> >
> > Best Regards
> >
> > Magnus
> >
> > Alan Clark wrote:
> > > Dave,
> > >
> > > RTCP XR can be extremely useful in supporting problem detection, call
> > > admission control etc. as it provides a fast method for a probe,
> > > analyzer, session controller... to get call quality related information
> > > from the endpoint.  In general the data available in RTCP XR should not
> > > represent a security concern - unlike SDES - and hence (I think) there
> > > is no advantage to applying encryption and considerable disadvantage.
> > >
> > > In an Enterprise network using VPN for inter-site connections you could
> > > consider using SRTP for RTP payloads and rely on the VPN tunnel to
> > > protect RTCP SDES information, which is a good compromise between
> > > management and security.  In some other applications, notably IP
> > > Centrex, the service provider is already handling signaling - RTCP XR
> > > gives them feedback on call quality from the customer desktop and
> > > alerts them to customer LAN problems that affect call quality.
> > >
> > > Regards
> > >
> > > Alan
> > >
> > >
> > >
> > >
> > > -----Original Message-----
> > > From: David R. Oran [mailto:oran@cisco.com]
> > > Sent: Sunday, May 25, 2003 5:46 PM
> > > To: Alan Clark; Paul E. Jones; 'Siddiqui, Anwar A (Anwar)'; 'Colin
> > > Perkins'; 'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> > > Cc: avt@ietf.org
> > > Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> > >
> > >
> > > --On Sunday, May 25, 2003 8:11 AM -0400 Alan Clark <alan@telchemy.com>
> > > wrote:
> > >
> > >
> > >>Paul
> > >>
> > >>If you bear in mind that RTCP XR is likely to be used to exchange
> > >>information between endpoints through the media path then there are
> > >>several implementation considerations.
> > >>
> > >>(i) It is quite likely that intermediate systems such as Ethernet
> > >>switches, edge routers and shapers may scan RTCP XR payloads.  A number
> > >>of such systems already do this with RTCP SR/RR reports, either for QoS
> > >>monitoring or for active control.  With switch speeds getting to 10
> > >>gigabit then it is important to keep parsing complexity very low.
> > >>
> > >
> > > Really? Boy, I'm glad I'm using SRTP and encrypting my RTCP packets.> 
> > > Put another way, I don't find snooping to be a valid argument in favor
> > > of fast parsing.
> > > That said, I do prefer to have an encoding which uses "value not valid"
> > > semantics rather than "value not present", because, like is done with
> > > IEEE floating point, it has much better robustness in the face of
> > > cascaded computations that may use inappropriate arithmetic. In fact, I
> > > think that's where the real savings is - in not having to scatter your
> > > data reduction for stats with a zillion if tests and multiple code
> > > patsh based on what dat ayou have. You just do the computation and
> > > error propagation will take care of ensuring that invalid stuff is
> > > marked as such.
> > >
> > >
> > >>(ii) High density gateways are pushing the density of CODECs per DSP
> > >but>also pushing RTP packet processing into the DSP.  This places severe
> > >>constraints on the available program and data memory space - again a
> > >>strong incentive to keep things as simple as possible.
> > >>
> > >
> > > Simplicity does resonate as a good argument here.
> > >
> > > Dave.
> > >
> > >
> > >>Regards
> > >>
> > >>Alan
> > >>
> > >>-----Original Message-----
> > >>From: Paul E. Jones [mailto:paulej@packetizer.com]
> > >>Sent: Saturday, May 24, 2003 10:40 AM
> > >>To: alan@telchemy.com; 'Siddiqui, Anwar A (Anwar)'; 'Colin Perkins';
> > >>'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
> > >>Cc: avt@ietf.org
> > >>Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> > >>
> > >>
> > >>Alan,
> > >>
> > >>I still think we should allow every field to be optional.  I recognize
> > >the>desire to encourage reporting a complete set of stats, but I would
> > >tend to>believe that there will be environments wherein not all stats
> > >are>necessary.
> > >>
> > >>Now, not to sound that I'm in disagreement with Anwar, but I think it
> > >>might be useful to allow one to indicate that a particular parameter is
> > >>not measured by populating the field with some default value.  This
> > >>default value would not indicate a "valid", but be an indicator of the
> > >>fact that the field is not present.  This would address your concern
> > >that>the fields and reporting blocks are fixed in size.
> > >>
> > >>I also realize that some time will be wasted to determine that a
> > >>measurement is not present, but the same amount of time will be
> > >consumed>by a system to extract a valid value today.  So, the system
> > >performance>is the same, albeit with potentially less useful data.
> > >>
> > >>Paul
> > >>
> > >>----- Original Message -----
> > >>From: "Alan Clark" <alan@telchemy.com>
> > >>To: "'Siddiqui, Anwar A (Anwar)'" <anwars@avaya.com>; "'Colin Perkins'"
> > >><csp@csperkins.org>; "'Timur Friedman'" <timur.friedman@lip6.fr>;
> > >>"'Stephen Casner'" <casner@acm.org>; "'Ramon Caceres'"
> > >><ramon@shieldip.com> Cc: <avt@ietf.org>
> > >>Sent: Thursday, May 22, 2003 10:14 AM
> > >>Subject: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
> > >>
> > >>
> > >>
> > >>>Anwar
> > >>>
> > >>>This was brought up several weeks ago on the mailing list.  This
> > >approach>>was taken deliberately for several reasons:-
> > >>>
> > >>>(i) To encourage implementors to move towards implementing the
> > >complete>
> > >>set
> > >>
> > >>>of metrics.  I've socialized this data set with a variety of service
> > >>>providers, enterprise network managers, equipment manufacturers and
> > >there>>seems to be general agreement that this metrics set provides a
> > >good>
> > >>"minimum
> > >>
> > >>>set" for diagnosing VoIP performance problems.
> > >>>
> > >>>(ii) There is overhead associated with identifying whether fields are
> > >or>
> > >>are
> > >>
> > >>>not present (see Colin Perkins comment on this issue several weeks
> > >ago).>>
> > >>>I am aware of several major equipment manufacturers and DSP software
> > >>
> > >>vendors
> > >>
> > >>>that intend to fully implement the RTCP XR VoIP metrics, and several> 
> > >tier>>one carriers that are currently specifying that this metric set be
> > >used>
> > >>for
> > >>
> > >>>VoIP service management. There is also considerable interest from
> > >other>>standards groups in this work.
> > >>>
> > >>>Regards
> > >>>
> > >>>Alan Clark
> > >>>Telchemy
> > >>>
> > >>>-----Original Message-----
> > >>>From: Siddiqui, Anwar A (Anwar) [mailto:anwars@avaya.com]
> > >>>Sent: Thursday, May 22, 2003 9:23 AM
> > >>>To: Colin Perkins; Timur Friedman; Stephen Casner; alan@telchemy.com;
> > >>>Ramon Caceres
> > >>>Cc: avt@ietf.org
> > >>>Subject: Re: draft-ietf-avt-rtcp-report-extns-06.txt
> > >>>
> > >>>
> > >>>Timur, Alan
> > >>>
> > >>>I did mention this during last IETF session. And I believe few other
> > >>
> > >>people
> > >>
> > >>>also have similar concerns. So here is the issue I am requesting this
> > >be>>reconsidered and if possible be avoided.
> > >>>
> > >>>------------------------------
> > >>>Is there anyway that we can avoid hard coding VoIP metrics like it is
> > >>>identified in this draft. Its not the usefulness of the metrics that I
> > >am>>having difficulties with. I actually believe they are useful
> > >parameters>>to be included in RTCP. But its just that its hardcoded
> > >right now>>....and if
> > >>
> > >>an
> > >>
> > >>>RTP source does not have parameters to report it will keep sending
> > >these>>default hard coded values! It is this kind of hardcoding in SR/RR
> > >reports>>that makes us do this XR reports now. So I am requesting if it
> > >possible>
> > >>such
> > >>
> > >>>hard coding of VoIP metrics be avoided.
> > >>>-------------------------------
> > >>>
> > >>>Thanks in advance
> > >>>
> > >>>Anwar
> > >>>
> > >>>
> > >>>
> > >>>----- Original Message -----
> > >>>From: "Colin Perkins" <csp@csperkins.org>
> > >>>To: "Timur Friedman" <timur.friedman@lip6.fr>
> > >>>Cc: <avt@ietf.org>; "Stephen Casner" <casner@acm.org>; "Alan Clark"
> > >>><alan@telchemy.com>; "Ramon Caceres" <ramon@shieldip.com>; "Kevin C.
> > >>>Almeroth" <almeroth@cs.ucsb.edu>; "Robert Cole" <rgcole@att.com>;
> > >"Nick>>Duffield" <duffield@research.att.com>; "Kaynam Hedayat"
> > >>><khedayat@brixnet.com>; "Kamil Sarac" <ksarac@utdallas.edu>; "Magnus
> > >>>Westerlund" <magnus.westerlund@era.ericsson.se>
> > >>>Sent: Monday, May 19, 2003 8:05 PM
> > >>>Subject: [AVT] Re: draft-ietf-avt-rtcp-report-extns-06.txt
> > >>>
> > >>>
> > >>>
> > >>>>Thanks. Unless there are objections, we intend to request that the
> > >IESG>>>consider this version for publication as a proposed standard.
> > >>>>
> > >>>>Colin
> > >>>>
> > >>>>
> > >>>>--> Timur Friedman writes:
> > >>>>
> > >>>>>Hello,
> > >>>>>
> > >>>>>Please find attached the latest version of the RTP Control Protocol
> > >>>>
> > >>>Extended
> > >>>
> > >>>>>Reports (RTCP XR) draft, draft-ietf-avt-rtcp-report-extns-06.txt.
> > >>>>>
> > >>>>>This draft was produced in response to AVT working group last call
> > >>>>
> > >>>comments.
> > >>>
> > >>>>>In addition to the changes discussed online on the AVT list, some
> > >>>>
> > >>>additional
> > >>>
> > >>>>>detail has been added to the SDP signaling section, to allow
> > >>>>>individual
> > >>>>
> > >>>flags
> > >>>
> > >>>>>in the Statistics Summary report blocks to be signaled.  Also,
> > >>>>>default
> > >>>>
> > >>>values
> > >>>
> > >>>>>have been specified for the VoIP metrics section in places where
> > >>>>>there
> > >>>>
> > >>>might
> > >>>
> > >>>>>have been some confusion about what to do if no packets had been
> > >>>>
> > >>>received.
> > >>>
> > >>>>>And the time frame over which some of the VoIP fields are calculated
> > >>>>
> > >>has
> > >>
> > >>>>>also been specified.
> > >>>>>
> > >>>>>A PDF version of this draft is available at the following URL:
> > >>>>>
> > >>>>
> > >>>http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-r
> > >epo>>rt- extns-06.pdf
> > >>>
> > >>>>>Regards,
> > >>>>>
> > >>>>>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
> > >
> > >
> > >
> > >
> > > ------------------------
> > > David R. Oran
> > > Cisco Systems
> > > 7 Ladyslipper Lane
> > > Acton, MA 01720
> > > Office: +1 978 264 2048
> > > VoIP: +1 408 571 4576
> > > Email: oran@cisco.com
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> > >
> >
> >
> > --
> >
> > 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@ericsson.com
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
> 
> 
> --
> Colin Perkins
> csp@csperkins.org
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
_______________________________________________
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Subject: [AVT] updated RTP MIDI implementation guide I-D ...
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Hi everyone,

	Just sent in an updated (and renamed) "An Implementation Guide
for RTP MIDI" I-D off to internet-drafts@ietf.org.  Comments are
welcome as always.  You can download a copy now from:

  http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-guide.txt

	Th document reflects side conversations at the San Francisco
IETF: the 50% of the previous I-D draft that offered generic guidance
about RTP has been deleted.

	What remains is MIDI-specific advice about how to packetize a
source into RTP, how to unpacketize RTP MIDI back into a high-quality
MIDI command stream, and how to send and receive recovery journals.

	I'd like to request WG Item status for the document, since I
believe that coders that don't read it might end up wasting a lot of
time re-discovering or reverse-engineering core algorithms.

	I would think the re-badged document would be ready for Last
Call in parallel with the normative RTP MIDI payload format document.
I believe the document satisfies http://www.ietf.org/ID-nits.html.

-------------------------------------------------------------------------
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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--NextPart

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

	Title		: The Secure Real-time Transport Protocol
	Author(s)	: M. Baugher et al.
	Filename	: draft-ietf-avt-srtp-07.txt
	Pages		: 54
	Date		: 2003-5-30
	
This document describes the Secure Real-time Transport Protocol   
(SRTP), a profile of the Real-time Transport Protocol (RTP), which 
can provide confidentiality, message authentication, and replay 
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Date: Mon, 02 Jun 2003 11:08:19 -0400
From: "David R. Oran" <oran@cisco.com>
To: Colin Perkins <csp@csperkins.org>,
        Magnus Westerlund <magnus.westerlund@ericsson.com>
cc: alan@telchemy.com, avt@ietf.org
Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
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References: <NLEHICNGLFCIFLBLOAKLGENNCCAA.alan@telchemy.com>
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Works for me.
Dave.

--On Sunday, June 01, 2003 11:02 AM -0400 Colin Perkins <csp@csperkins.org> 
wrote:

> Folks,
>
> Would Magnus' proposal satisfy all involved in this discussion?
> Colin
>
>
> --> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>> Hi,
>>
>> Looking at this discussion it seems that one can make two resolutions.
>>
>> 1. Do not mandate calculation of all fields in report block. I would
>> think that a SHOULD calculate all field is a good level. Text motivating
>> that all fields should be calculated is good.
>>
>> 2. Keep the structure as is, define for each field a "NOT VALID" value.
>>
>> I think this is reasonable compromise. It keeps the VoIP report block
>> simple to process and at the same time a implementor that is unable to
>> provide a measurement does not need to.
>>
>> Best Regards
>>
>> Magnus
>>
>> Alan Clark wrote:
>> > Dave,
>> >
>> > RTCP XR can be extremely useful in supporting problem detection, call
>> > admission control etc. as it provides a fast method for a probe,
>> > analyzer, session controller... to get call quality related information
>> > from the endpoint.  In general the data available in RTCP XR should not
>> > represent a security concern - unlike SDES - and hence (I think) there
>> > is no advantage to applying encryption and considerable disadvantage.
>> >
>> > In an Enterprise network using VPN for inter-site connections you could
>> > consider using SRTP for RTP payloads and rely on the VPN tunnel to
>> > protect RTCP SDES information, which is a good compromise between
>> > management and security.  In some other applications, notably IP
>> > Centrex, the service provider is already handling signaling - RTCP XR
>> > gives them feedback on call quality from the customer desktop and
>> > alerts them to customer LAN problems that affect call quality.
>> >
>> > Regards
>> >
>> > Alan
>> >
>> >
>> >
>> >
>> > -----Original Message-----
>> > From: David R. Oran [mailto:oran@cisco.com]
>> > Sent: Sunday, May 25, 2003 5:46 PM
>> > To: Alan Clark; Paul E. Jones; 'Siddiqui, Anwar A (Anwar)'; 'Colin
>> > Perkins'; 'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
>> > Cc: avt@ietf.org
>> > Subject: RE: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
>> >
>> >
>> > --On Sunday, May 25, 2003 8:11 AM -0400 Alan Clark <alan@telchemy.com>
>> > wrote:
>> >
>> >
>> >> Paul
>> >>
>> >> If you bear in mind that RTCP XR is likely to be used to exchange
>> >> information between endpoints through the media path then there are
>> >> several implementation considerations.
>> >>
>> >> (i) It is quite likely that intermediate systems such as Ethernet
>> >> switches, edge routers and shapers may scan RTCP XR payloads.  A
>> >> number of such systems already do this with RTCP SR/RR reports,
>> >> either for QoS monitoring or for active control.  With switch speeds
>> >> getting to 10 gigabit then it is important to keep parsing complexity
>> >> very low.
>> >>
>> >
>> > Really? Boy, I'm glad I'm using SRTP and encrypting my RTCP packets.
>> > Put another way, I don't find snooping to be a valid argument in favor
>> > of fast parsing.
>> > That said, I do prefer to have an encoding which uses "value not valid"
>> > semantics rather than "value not present", because, like is done with
>> > IEEE floating point, it has much better robustness in the face of
>> > cascaded computations that may use inappropriate arithmetic. In fact, I
>> > think that's where the real savings is - in not having to scatter your
>> > data reduction for stats with a zillion if tests and multiple code
>> > patsh based on what dat ayou have. You just do the computation and
>> > error propagation will take care of ensuring that invalid stuff is
>> > marked as such.
>> >
>> >
>> >> (ii) High density gateways are pushing the density of CODECs per DSP
>> > but>also pushing RTP packet processing into the DSP.  This places
>> > severe
>> >> constraints on the available program and data memory space - again a
>> >> strong incentive to keep things as simple as possible.
>> >>
>> >
>> > Simplicity does resonate as a good argument here.
>> >
>> > Dave.
>> >
>> >
>> >> Regards
>> >>
>> >> Alan
>> >>
>> >> -----Original Message-----
>> >> From: Paul E. Jones [mailto:paulej@packetizer.com]
>> >> Sent: Saturday, May 24, 2003 10:40 AM
>> >> To: alan@telchemy.com; 'Siddiqui, Anwar A (Anwar)'; 'Colin Perkins';
>> >> 'Timur Friedman'; 'Stephen Casner'; 'Ramon Caceres'
>> >> Cc: avt@ietf.org
>> >> Subject: Re: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
>> >>
>> >>
>> >> Alan,
>> >>
>> >> I still think we should allow every field to be optional.  I recognize
>> > the>desire to encourage reporting a complete set of stats, but I would
>> > tend to>believe that there will be environments wherein not all stats
>> > are>necessary.
>> >>
>> >> Now, not to sound that I'm in disagreement with Anwar, but I think it
>> >> might be useful to allow one to indicate that a particular parameter
>> >> is not measured by populating the field with some default value.  This
>> >> default value would not indicate a "valid", but be an indicator of the
>> >> fact that the field is not present.  This would address your concern
>> > that>the fields and reporting blocks are fixed in size.
>> >>
>> >> I also realize that some time will be wasted to determine that a
>> >> measurement is not present, but the same amount of time will be
>> > consumed>by a system to extract a valid value today.  So, the system
>> > performance>is the same, albeit with potentially less useful data.
>> >>
>> >> Paul
>> >>
>> >> ----- Original Message -----
>> >> From: "Alan Clark" <alan@telchemy.com>
>> >> To: "'Siddiqui, Anwar A (Anwar)'" <anwars@avaya.com>; "'Colin
>> >> Perkins'" <csp@csperkins.org>; "'Timur Friedman'"
>> >> <timur.friedman@lip6.fr>; "'Stephen Casner'" <casner@acm.org>;
>> >> "'Ramon Caceres'"
>> >> <ramon@shieldip.com> Cc: <avt@ietf.org>
>> >> Sent: Thursday, May 22, 2003 10:14 AM
>> >> Subject: [AVT] RE: draft-ietf-avt-rtcp-report-extns-06.txt
>> >>
>> >>
>> >>
>> >>> Anwar
>> >>>
>> >>> This was brought up several weeks ago on the mailing list.  This
>> > approach>>was taken deliberately for several reasons:-
>> >>>
>> >>> (i) To encourage implementors to move towards implementing the
>> > complete>
>> >> set
>> >>
>> >>> of metrics.  I've socialized this data set with a variety of service
>> >>> providers, enterprise network managers, equipment manufacturers and
>> > there>>seems to be general agreement that this metrics set provides a
>> > good>
>> >> "minimum
>> >>
>> >>> set" for diagnosing VoIP performance problems.
>> >>>
>> >>> (ii) There is overhead associated with identifying whether fields are
>> > or>
>> >> are
>> >>
>> >>> not present (see Colin Perkins comment on this issue several weeks
>> > ago).>>
>> >>> I am aware of several major equipment manufacturers and DSP software
>> >>
>> >> vendors
>> >>
>> >>> that intend to fully implement the RTCP XR VoIP metrics, and several
>> > tier>>one carriers that are currently specifying that this metric set
>> > be used>
>> >> for
>> >>
>> >>> VoIP service management. There is also considerable interest from
>> > other>>standards groups in this work.
>> >>>
>> >>> Regards
>> >>>
>> >>> Alan Clark
>> >>> Telchemy
>> >>>
>> >>> -----Original Message-----
>> >>> From: Siddiqui, Anwar A (Anwar) [mailto:anwars@avaya.com]
>> >>> Sent: Thursday, May 22, 2003 9:23 AM
>> >>> To: Colin Perkins; Timur Friedman; Stephen Casner; alan@telchemy.com;
>> >>> Ramon Caceres
>> >>> Cc: avt@ietf.org
>> >>> Subject: Re: draft-ietf-avt-rtcp-report-extns-06.txt
>> >>>
>> >>>
>> >>> Timur, Alan
>> >>>
>> >>> I did mention this during last IETF session. And I believe few other
>> >>
>> >> people
>> >>
>> >>> also have similar concerns. So here is the issue I am requesting this
>> > be>>reconsidered and if possible be avoided.
>> >>>
>> >>> ------------------------------
>> >>> Is there anyway that we can avoid hard coding VoIP metrics like it is
>> >>> identified in this draft. Its not the usefulness of the metrics that
>> >>> I
>> > am>>having difficulties with. I actually believe they are useful
>> > parameters>>to be included in RTCP. But its just that its hardcoded
>> > right now>>....and if
>> >>
>> >> an
>> >>
>> >>> RTP source does not have parameters to report it will keep sending
>> > these>>default hard coded values! It is this kind of hardcoding in
>> > SR/RR reports>>that makes us do this XR reports now. So I am
>> > requesting if it possible>
>> >> such
>> >>
>> >>> hard coding of VoIP metrics be avoided.
>> >>> -------------------------------
>> >>>
>> >>> Thanks in advance
>> >>>
>> >>> Anwar
>> >>>
>> >>>
>> >>>
>> >>> ----- Original Message -----
>> >>> From: "Colin Perkins" <csp@csperkins.org>
>> >>> To: "Timur Friedman" <timur.friedman@lip6.fr>
>> >>> Cc: <avt@ietf.org>; "Stephen Casner" <casner@acm.org>; "Alan Clark"
>> >>> <alan@telchemy.com>; "Ramon Caceres" <ramon@shieldip.com>; "Kevin C.
>> >>> Almeroth" <almeroth@cs.ucsb.edu>; "Robert Cole" <rgcole@att.com>;
>> > "Nick>>Duffield" <duffield@research.att.com>; "Kaynam Hedayat"
>> >>> <khedayat@brixnet.com>; "Kamil Sarac" <ksarac@utdallas.edu>; "Magnus
>> >>> Westerlund" <magnus.westerlund@era.ericsson.se>
>> >>> Sent: Monday, May 19, 2003 8:05 PM
>> >>> Subject: [AVT] Re: draft-ietf-avt-rtcp-report-extns-06.txt
>> >>>
>> >>>
>> >>>
>> >>>> Thanks. Unless there are objections, we intend to request that the
>> > IESG>>>consider this version for publication as a proposed standard.
>> >>>>
>> >>>> Colin
>> >>>>
>> >>>>
>> >>>> --> Timur Friedman writes:
>> >>>>
>> >>>>> Hello,
>> >>>>>
>> >>>>> Please find attached the latest version of the RTP Control Protocol
>> >>>>
>> >>> Extended
>> >>>
>> >>>>> Reports (RTCP XR) draft, draft-ietf-avt-rtcp-report-extns-06.txt.
>> >>>>>
>> >>>>> This draft was produced in response to AVT working group last call
>> >>>>
>> >>> comments.
>> >>>
>> >>>>> In addition to the changes discussed online on the AVT list, some
>> >>>>
>> >>> additional
>> >>>
>> >>>>> detail has been added to the SDP signaling section, to allow
>> >>>>> individual
>> >>>>
>> >>> flags
>> >>>
>> >>>>> in the Statistics Summary report blocks to be signaled.  Also,
>> >>>>> default
>> >>>>
>> >>> values
>> >>>
>> >>>>> have been specified for the VoIP metrics section in places where
>> >>>>> there
>> >>>>
>> >>> might
>> >>>
>> >>>>> have been some confusion about what to do if no packets had been
>> >>>>
>> >>> received.
>> >>>
>> >>>>> And the time frame over which some of the VoIP fields are
>> >>>>> calculated
>> >>>>
>> >> has
>> >>
>> >>>>> also been specified.
>> >>>>>
>> >>>>> A PDF version of this draft is available at the following URL:
>> >>>>>
>> >>>>
>> >>> http://www-rp.lip6.fr/site_rp/_publications/378-draft-ietf-avt-rtcp-r
>> > epo>>rt- extns-06.pdf
>> >>>
>> >>>>> Regards,
>> >>>>>
>> >>>>> 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
>> >
>> >
>> >
>> >
>> > ------------------------
>> > David R. Oran
>> > Cisco Systems
>> > 7 Ladyslipper Lane
>> > Acton, MA 01720
>> > Office: +1 978 264 2048
>> > VoIP: +1 408 571 4576
>> > Email: oran@cisco.com
>> >
>> > _______________________________________________
>> > Audio/Video Transport Working Group
>> > avt@ietf.org
>> > https://www1.ietf.org/mailman/listinfo/avt
>> >
>>
>>
>> --
>>
>> 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@ericsson.com
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
>
> --
> Colin Perkins
> csp@csperkins.org
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt



------------------------
David R. Oran
Cisco Systems
7 Ladyslipper Lane
Acton, MA 01720
Office: +1 978 264 2048
VoIP: +1 408 571 4576
Email: oran@cisco.com
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Mon, 02 Jun 2003 13:49:33 -0400
Subject: [AVT] Last Call: RTP Payload Format for Transport of MPEG-4
 Elementary Streams to Proposed Standard
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The IESG has received a request from the Audio/Video Transport 
Working Group to consider RTP Payload Format for Transport of 
MPEG-4 Elementary Streams <draft-ietf-avt-mpeg4-simple-07.txt> 
as a Proposed Standard.  

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the 
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-6-16.

Files can be obtained via 
http://www.ietf.org/internet-drafts/draft-ietf-avt-mpeg4-simple-07.txt



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



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Date: Mon, 02 Jun 2003 14:05:38 -0700
To: avt@ietf.org
From: Mark Baugher <mbaugher@cisco.com>
Subject: Fwd: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
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Hello,
   This I-D contains errors that have been fixed in 
http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt, which I 
just submitted to internet-drafts.  Please refer to revision 08.

   This revision of the draft was made for IESG consumption though it has 
some clarifications that we should point out to you.

  - We clarified the positioning of SRTP as a profile of RTP which 
incorporates all the features of RTP/AVP except where explicitly noted

  - The 1st paragraph after Figure 1 has been re-written to clarify that 
SRTP does not use RTP padding bytes and that any future block-mode cipher 
needs to freshly describe how it does padding in its RFC

  - Corrected the 4th paragraph of section 3.2.3 to refer to SRTP packets only

  - Section 3.3.1 now completely describes how s_l and ROC are 
conditionally updated (in the two paragraphs that follow "i = 2^16 * v + 
SEQ") and we don't use the word "estimated" to describe packet-index 
determination

  - Caption of Figure 2: "with a Report" --> "with a Sender Report"

  - We made it clear that ROC || SEQ must be distinct in 4.1.1 and not just SEQ

  - We now reference rtp-new rather than RFC 1889

Again, sorry for the confusion and please refer to 
http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt until 08 
is properly posted

Mark
>To: IETF-Announce: ;
>Cc: avt@ietf.org
>From: Internet-Drafts@ietf.org
>Reply-to: Internet-Drafts@ietf.org
>Date: Mon, 02 Jun 2003 07:28:34 -0400
>Subject: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
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>
>A New Internet-Draft is available from the on-line Internet-Drafts 
>directories.
>This draft is a work item of the Audio/Video Transport Working Group of 
>the IETF.
>
>         Title           : The Secure Real-time Transport Protocol
>         Author(s)       : M. Baugher et al.
>         Filename        : draft-ietf-avt-srtp-07.txt
>         Pages           : 54
>         Date            : 2003-5-30
>
>This document describes the Secure Real-time Transport Protocol
>(SRTP), a profile of the Real-time Transport Protocol (RTP), which
>can provide confidentiality, message authentication, and replay
>protection to the RTP/RTCP traffic.
>
>A URL for this Internet-Draft is:
>http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt
>
>To remove yourself from the IETF Announcement list, send a message to
>ietf-announce-request with the word unsubscribe in the body of the message.
>
>Internet-Drafts are also available by anonymous FTP. Login with the username
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>type "cd internet-drafts" and then
>         "get draft-ietf-avt-srtp-07.txt".
>
>A list of Internet-Drafts directories can be found in
>http://www.ietf.org/shadow.html
>or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
>
>
>Internet-Drafts can also be obtained by e-mail.
>
>Send a message to:
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>In the body type:
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>
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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	(envelope-from csp@csperkins.org)
Date: Mon, 2 Jun 2003 17:38:53 -0400
From: Colin Perkins <csp@csperkins.org>
To: Mark Baugher <mbaugher@cisco.com>
Cc: avt@ietf.org, mankin@psg.com
Subject: Re: Fwd: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
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Mark,

Apart from the filename, your -08 and the version of -07 posted appear to
be identical. Looks like the internet-drafts adminstrator may have posted
your update as -07?

Colin


--> Mark Baugher <mbaugher@cisco.com> writes:
> Hello,
>    This I-D contains errors that have been fixed in 
> http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt, which
> I just submitted to internet-drafts.  Please refer to revision 08.
> 
>    This revision of the draft was made for IESG consumption though it has
>    
> some clarifications that we should point out to you.
> 
>   - We clarified the positioning of SRTP as a profile of RTP which 
> incorporates all the features of RTP/AVP except where explicitly noted
> 
>   - The 1st paragraph after Figure 1 has been re-written to clarify that 
> SRTP does not use RTP padding bytes and that any future block-mode cipher
> needs to freshly describe how it does padding in its RFC
> 
>   - Corrected the 4th paragraph of section 3.2.3 to refer to SRTP packets
>   only
> 
>   - Section 3.3.1 now completely describes how s_l and ROC are 
> conditionally updated (in the two paragraphs that follow "i = 2^16 * v + 
> SEQ") and we don't use the word "estimated" to describe packet-index 
> determination
> 
>   - Caption of Figure 2: "with a Report" --> "with a Sender Report"
> 
>   - We made it clear that ROC || SEQ must be distinct in 4.1.1 and not
>   just SEQ
> 
>   - We now reference rtp-new rather than RFC 1889
> 
> Again, sorry for the confusion and please refer to 
> http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt until
> 08 is properly posted
> 
> Mark
> >To: IETF-Announce: ;
> >Cc: avt@ietf.org
> >From: Internet-Drafts@ietf.org
> >Reply-to: Internet-Drafts@ietf.org
> >Date: Mon, 02 Jun 2003 07:28:34 -0400
> >Subject: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
> >Sender: avt-admin@ietf.org
> >X-BeenThere: avt@ietf.org
> >X-Mailman-Version: 2.0.12
> >List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> >         <mailto:avt-request@ietf.org?subject=unsubscribe>
> >List-Id: Audio/Video Transport Working Group <avt.ietf.org>
> >List-Post: <mailto:avt@ietf.org>
> >List-Help: <mailto:avt-request@ietf.org?subject=help>
> >List-Subscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> >         <mailto:avt-request@ietf.org?subject=subscribe>
> >
> >A New Internet-Draft is available from the on-line Internet-Drafts 
> >directories.
> >This draft is a work item of the Audio/Video Transport Working Group of 
> >the IETF.
> >
> >         Title           : The Secure Real-time Transport Protocol
> >         Author(s)       : M. Baugher et al.
> >         Filename        : draft-ietf-avt-srtp-07.txt
> >         Pages           : 54
> >         Date            : 2003-5-30
> >
> >This document describes the Secure Real-time Transport Protocol
> >(SRTP), a profile of the Real-time Transport Protocol (RTP), which
> >can provide confidentiality, message authentication, and replay
> >protection to the RTP/RTCP traffic.
> >
> >A URL for this Internet-Draft is:
> >http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt
> >
> >To remove yourself from the IETF Announcement list, send a message to
> >ietf-announce-request with the word unsubscribe in the body of the
> >message.
> >
> >Internet-Drafts are also available by anonymous FTP. Login with the
> >username"anonymous" and a password of your e-mail address. After logging
> >in, type "cd internet-drafts" and then
> >         "get draft-ietf-avt-srtp-07.txt".
> >
> >A list of Internet-Drafts directories can be found in
> >http://www.ietf.org/shadow.html
> >or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> >
> >
> >Internet-Drafts can also be obtained by e-mail.
> >
> >Send a message to:
> >         mailserv@ietf.org.
> >In the body type:
> >         "FILE /internet-drafts/draft-ietf-avt-srtp-07.txt".
> >
> >NOTE:   The mail server at ietf.org can return the document in
> >         MIME-encoded form by using the "mpack" utility.  To use this
> >         feature, insert the command "ENCODING mime" before the "FILE"
> >         command.  To decode the response(s), you will need "munpack" or
> >         a MIME-compliant mail reader.  Different MIME-compliant mail
> >         readers exhibit different behavior, especially when dealing
> >         with"multipart" MIME messages (i.e. documents which have been
> >         split up into multiple messages), so check your local
> >         documentation on how to manipulate these messages.
> >
> >
> >Below is the data which will enable a MIME compliant mail reader
> >implementation to automatically retrieve the ASCII version of the
> >Internet-Draft.
> >Content-Type: text/plain
> >Content-ID:     <2003-5-30152004.I-D@ietf.org>
> >
> >ENCODING mime
> >FILE /internet-drafts/draft-ietf-avt-srtp-07.txt
> >
> ><ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt>
> 
> 
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


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



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Date: Mon, 02 Jun 2003 15:11:32 -0700
To: Colin Perkins <csp@csperkins.org>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: Fwd: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
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Yes, you're right.  I don't know what I was looking at in 
internet-drafts.  Did I at least get the synopsis correct below?

thanks, Mark
At 05:38 PM 6/2/2003 -0400, Colin Perkins wrote:
>Mark,
>
>Apart from the filename, your -08 and the version of -07 posted appear to
>be identical. Looks like the internet-drafts adminstrator may have posted
>your update as -07?
>
>Colin
>
>
>--> Mark Baugher <mbaugher@cisco.com> writes:
> > Hello,
> >    This I-D contains errors that have been fixed in
> > http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt, which
> > I just submitted to internet-drafts.  Please refer to revision 08.
> >
> >    This revision of the draft was made for IESG consumption though it has
> >
> > some clarifications that we should point out to you.
> >
> >   - We clarified the positioning of SRTP as a profile of RTP which
> > incorporates all the features of RTP/AVP except where explicitly noted
> >
> >   - The 1st paragraph after Figure 1 has been re-written to clarify that
> > SRTP does not use RTP padding bytes and that any future block-mode cipher
> > needs to freshly describe how it does padding in its RFC
> >
> >   - Corrected the 4th paragraph of section 3.2.3 to refer to SRTP packets
> >   only
> >
> >   - Section 3.3.1 now completely describes how s_l and ROC are
> > conditionally updated (in the two paragraphs that follow "i = 2^16 * v +
> > SEQ") and we don't use the word "estimated" to describe packet-index
> > determination
> >
> >   - Caption of Figure 2: "with a Report" --> "with a Sender Report"
> >
> >   - We made it clear that ROC || SEQ must be distinct in 4.1.1 and not
> >   just SEQ
> >
> >   - We now reference rtp-new rather than RFC 1889
> >
> > Again, sorry for the confusion and please refer to
> > http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt until
> > 08 is properly posted
> >
> > Mark
> > >To: IETF-Announce: ;
> > >Cc: avt@ietf.org
> > >From: Internet-Drafts@ietf.org
> > >Reply-to: Internet-Drafts@ietf.org
> > >Date: Mon, 02 Jun 2003 07:28:34 -0400
> > >Subject: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
> > >Sender: avt-admin@ietf.org
> > >X-BeenThere: avt@ietf.org
> > >X-Mailman-Version: 2.0.12
> > >List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> > >         <mailto:avt-request@ietf.org?subject=unsubscribe>
> > >List-Id: Audio/Video Transport Working Group <avt.ietf.org>
> > >List-Post: <mailto:avt@ietf.org>
> > >List-Help: <mailto:avt-request@ietf.org?subject=help>
> > >List-Subscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> > >         <mailto:avt-request@ietf.org?subject=subscribe>
> > >
> > >A New Internet-Draft is available from the on-line Internet-Drafts
> > >directories.
> > >This draft is a work item of the Audio/Video Transport Working Group of
> > >the IETF.
> > >
> > >         Title           : The Secure Real-time Transport Protocol
> > >         Author(s)       : M. Baugher et al.
> > >         Filename        : draft-ietf-avt-srtp-07.txt
> > >         Pages           : 54
> > >         Date            : 2003-5-30
> > >
> > >This document describes the Secure Real-time Transport Protocol
> > >(SRTP), a profile of the Real-time Transport Protocol (RTP), which
> > >can provide confidentiality, message authentication, and replay
> > >protection to the RTP/RTCP traffic.
> > >
> > >A URL for this Internet-Draft is:
> > >http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt
> > >
> > >To remove yourself from the IETF Announcement list, send a message to
> > >ietf-announce-request with the word unsubscribe in the body of the
> > >message.
> > >
> > >Internet-Drafts are also available by anonymous FTP. Login with the
> > >username"anonymous" and a password of your e-mail address. After logging
> > >in, type "cd internet-drafts" and then
> > >         "get draft-ietf-avt-srtp-07.txt".
> > >
> > >A list of Internet-Drafts directories can be found in
> > >http://www.ietf.org/shadow.html
> > >or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > >
> > >
> > >Internet-Drafts can also be obtained by e-mail.
> > >
> > >Send a message to:
> > >         mailserv@ietf.org.
> > >In the body type:
> > >         "FILE /internet-drafts/draft-ietf-avt-srtp-07.txt".
> > >
> > >NOTE:   The mail server at ietf.org can return the document in
> > >         MIME-encoded form by using the "mpack" utility.  To use this
> > >         feature, insert the command "ENCODING mime" before the "FILE"
> > >         command.  To decode the response(s), you will need "munpack" or
> > >         a MIME-compliant mail reader.  Different MIME-compliant mail
> > >         readers exhibit different behavior, especially when dealing
> > >         with"multipart" MIME messages (i.e. documents which have been
> > >         split up into multiple messages), so check your local
> > >         documentation on how to manipulate these messages.
> > >
> > >
> > >Below is the data which will enable a MIME compliant mail reader
> > >implementation to automatically retrieve the ASCII version of the
> > >Internet-Draft.
> > >Content-Type: text/plain
> > >Content-ID:     <2003-5-30152004.I-D@ietf.org>
> > >
> > >ENCODING mime
> > >FILE /internet-drafts/draft-ietf-avt-srtp-07.txt
> > >
> > ><ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt>
> >
> >
> > _______________________________________________
> > Audio/Video Transport Working Group
> > avt@ietf.org
> > https://www1.ietf.org/mailman/listinfo/avt
>
>
>--
>Colin Perkins
>csp@csperkins.org


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



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Date: Mon, 2 Jun 2003 18:24:18 -0400
From: Colin Perkins <csp@csperkins.org>
To: Mark Baugher <mbaugher@cisco.com>
Cc: avt@ietf.org, mankin@psg.com
Subject: Re: Fwd: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
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Yes, I think so.
Colin


--> Mark Baugher <mbaugher@cisco.com> writes:
> Yes, you're right.  I don't know what I was looking at in 
> internet-drafts.  Did I at least get the synopsis correct below?
> 
> thanks, Mark
> At 05:38 PM 6/2/2003 -0400, Colin Perkins wrote:
> >Mark,
> >
> >Apart from the filename, your -08 and the version of -07 posted appear
> >to be identical. Looks like the internet-drafts adminstrator may have
> >posted your update as -07?
> >
> >Colin
> >
> >
> >--> Mark Baugher <mbaugher@cisco.com> writes:
> > > Hello,
> > >    This I-D contains errors that have been fixed in
> > > http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt,
> > > which I just submitted to internet-drafts.  Please refer to revision
> > > 08.
> > >
> > >    This revision of the draft was made for IESG consumption though it
> > >    has
> > >
> > > some clarifications that we should point out to you.
> > >
> > >   - We clarified the positioning of SRTP as a profile of RTP which
> > > incorporates all the features of RTP/AVP except where explicitly
> > > noted
> > >
> > >   - The 1st paragraph after Figure 1 has been re-written to clarify
> > >   that
> > > SRTP does not use RTP padding bytes and that any future block-mode
> > > cipher needs to freshly describe how it does padding in its RFC
> > >
> > >   - Corrected the 4th paragraph of section 3.2.3 to refer to SRTP
> > >   packets only
> > >
> > >   - Section 3.3.1 now completely describes how s_l and ROC are
> > > conditionally updated (in the two paragraphs that follow "i = 2^16 *
> > > v + SEQ") and we don't use the word "estimated" to describe
> > > packet-index determination
> > >
> > >   - Caption of Figure 2: "with a Report" --> "with a Sender Report"
> > >
> > >   - We made it clear that ROC || SEQ must be distinct in 4.1.1 and
> > >   not just SEQ
> > >
> > >   - We now reference rtp-new rather than RFC 1889
> > >
> > > Again, sorry for the confusion and please refer to
> > > http://www.rdrop.com/users/mbaugher/I-D/draft-ietf-avt-srtp-08.txt
> > > until 08 is properly posted
> > >
> > > Mark
> > > >To: IETF-Announce: ;
> > > >Cc: avt@ietf.org
> > > >From: Internet-Drafts@ietf.org
> > > >Reply-to: Internet-Drafts@ietf.org
> > > >Date: Mon, 02 Jun 2003 07:28:34 -0400
> > > >Subject: [AVT] I-D ACTION:draft-ietf-avt-srtp-07.txt
> > > >Sender: avt-admin@ietf.org
> > > >X-BeenThere: avt@ietf.org
> > > >X-Mailman-Version: 2.0.12
> > > >List-Unsubscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> > > >         <mailto:avt-request@ietf.org?subject=unsubscribe>
> > > >List-Id: Audio/Video Transport Working Group <avt.ietf.org>
> > > >List-Post: <mailto:avt@ietf.org>
> > > >List-Help: <mailto:avt-request@ietf.org?subject=help>
> > > >List-Subscribe: <https://www1.ietf.org/mailman/listinfo/avt>,
> > > >         <mailto:avt-request@ietf.org?subject=subscribe>
> > > >
> > > >A New Internet-Draft is available from the on-line Internet-Drafts
> > > >directories.
> > > >This draft is a work item of the Audio/Video Transport Working Group
> > > >of the IETF.
> > > >
> > > >         Title           : The Secure Real-time Transport Protocol
> > > >         Author(s)       : M. Baugher et al.
> > > >         Filename        : draft-ietf-avt-srtp-07.txt
> > > >         Pages           : 54
> > > >         Date            : 2003-5-30
> > > >
> > > >This document describes the Secure Real-time Transport Protocol
> > > >(SRTP), a profile of the Real-time Transport Protocol (RTP), which
> > > >can provide confidentiality, message authentication, and replay
> > > >protection to the RTP/RTCP traffic.
> > > >
> > > >A URL for this Internet-Draft is:
> > > >http://www.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt
> > > >
> > > >To remove yourself from the IETF Announcement list, send a message
> > > >to ietf-announce-request with the word unsubscribe in the body of
> > > >the message.
> > > >
> > > >Internet-Drafts are also available by anonymous FTP. Login with the
> > > >username"anonymous" and a password of your e-mail address. After
> > > >logging in, type "cd internet-drafts" and then
> > > >         "get draft-ietf-avt-srtp-07.txt".
> > > >
> > > >A list of Internet-Drafts directories can be found in
> > > >http://www.ietf.org/shadow.html
> > > >or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
> > > >
> > > >
> > > >Internet-Drafts can also be obtained by e-mail.
> > > >
> > > >Send a message to:
> > > >         mailserv@ietf.org.
> > > >In the body type:
> > > >         "FILE /internet-drafts/draft-ietf-avt-srtp-07.txt".
> > > >
> > > >NOTE:   The mail server at ietf.org can return the document in
> > > >         MIME-encoded form by using the "mpack" utility.  To use
> > > >         this feature, insert the command "ENCODING mime" before the
> > > >         "FILE" command.  To decode the response(s), you will need
> > > >         "munpack" or a MIME-compliant mail reader.  Different
> > > >         MIME-compliant mail readers exhibit different behavior,
> > > >         especially when dealing with"multipart" MIME messages (i.e.
> > > >         documents which have been split up into multiple messages),
> > > >         so check your local documentation on how to manipulate
> > > >         these messages.
> > > >
> > > >
> > > >Below is the data which will enable a MIME compliant mail reader
> > > >implementation to automatically retrieve the ASCII version of the
> > > >Internet-Draft.
> > > >Content-Type: text/plain
> > > >Content-ID:     <2003-5-30152004.I-D@ietf.org>
> > > >
> > > >ENCODING mime
> > > >FILE /internet-drafts/draft-ietf-avt-srtp-07.txt
> > > >
> > > ><ftp://ftp.ietf.org/internet-drafts/draft-ietf-avt-srtp-07.txt>
> > >
> > >
> > > _______________________________________________
> > > Audio/Video Transport Working Group
> > > avt@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/avt
> >
> >
> >--
> >Colin Perkins
> >csp@csperkins.org
> 
> 


-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
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.


	Title		: An Implementation Guide for RTP MIDI
	Author(s)	: J. Lazzaro, J. Wawrzynek
	Filename	: draft-lazzaro-avt-mwpp-coding-guidelines-03.txt
	Pages		: 35
	Date		: 2003-6-2
	
This memo offers non-normative implementation guidance for the RTP
MIDI payload format.  The memo presents its advice in the context
of a network musical performance application.  In this application
two musicians, located in different physical locations, interact
over a network to perform as they would if located in the same
room.  Underlying the performances are RTP MIDI sessions over
unicast UDP.  Algorithms for sending and receiving recovery
journals (the resiliency structure for the payload format) are
described in detail.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-lazzaro-avt-mwpp-coding-guidelines-03.txt

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ietf-announce-request with the word unsubscribe in the body of the message.

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"anonymous" and a password of your e-mail address. After logging in,
type "cd internet-drafts" and then
	"get draft-lazzaro-avt-mwpp-coding-guidelines-03.txt".

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	exhibit different behavior, especially when dealing with
	"multipart" MIME messages (i.e. documents which have been split
	up into multiple messages), so check your local documentation on
	how to manipulate these messages.
		
		
Below is the data which will enable a MIME compliant mail reader
implementation to automatically retrieve the ASCII version of the
Internet-Draft.

--NextPart
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--OtherAccess
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Content-ID:	<2003-6-2145511.I-D@ietf.org>

ENCODING mime
FILE /internet-drafts/draft-lazzaro-avt-mwpp-coding-guidelines-03.txt

--OtherAccess
Content-Type: Message/External-body;
	name="draft-lazzaro-avt-mwpp-coding-guidelines-03.txt";
	site="ftp.ietf.org";
	access-type="anon-ftp";
	directory="internet-drafts"

Content-Type: text/plain
Content-ID:	<2003-6-2145511.I-D@ietf.org>

--OtherAccess--

--NextPart--


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



From mailnull@www1.ietf.org  Tue Jun  3 10:46:58 2003
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Date: Tue, 3 Jun 2003 10:44:08 -0400
From: Colin Perkins <csp@csperkins.org>
To: John Lazzaro <lazzaro@CS.Berkeley.EDU>
Cc: avt@ietf.org
Subject: Re: [AVT] updated RTP MIDI implementation guide I-D ...
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--> John Lazzaro <lazzaro@CS.Berkeley.EDU> writes:
...
> 	Just sent in an updated (and renamed) "An Implementation Guide
> for RTP MIDI" I-D off to internet-drafts@ietf.org.  Comments are
> welcome as always.  You can download a copy now from:
> 
>   http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-guide.txt
> 
> 	Th document reflects side conversations at the San Francisco
> IETF: the 50% of the previous I-D draft that offered generic guidance
> about RTP has been deleted.
> 
> 	What remains is MIDI-specific advice about how to packetize a
> source into RTP, how to unpacketize RTP MIDI back into a high-quality
> MIDI command stream, and how to send and receive recovery journals.
> 
> 	I'd like to request WG Item status for the document, since I
> believe that coders that don't read it might end up wasting a lot of
> time re-discovering or reverse-engineering core algorithms.

Making this a working group draft, which can be submitted for informational
along with the payload format, seems like a reasonable request. Are there
any objections from the group?

Cheers,
-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Tue Jun  3 19:38:21 2003
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Subject: [AVT] Revised version of RFC 2250: Where is it?
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In the London IETF (August 2001), there was a discussion of a proposed new 
I-D (->RFC) that would revise RFC 2250 (the RTP payload format for MPEG-1,2 
audio,video).  Is this still an AVT work item?  If so, where is it?

	Ross.

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



From mailnull@www1.ietf.org  Wed Jun  4 00:13:47 2003
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Subject: [AVT] Interleaving out of band DTMF and Encoded Audio  packets
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This is a multi-part message in MIME format.

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----------------------------------------- (on the network)----------------

email-body was scanned and no virus found
email-body was scanned and no virus found
-----------------------------------------D-Link R & D E-Mail Traffic-----------

------=_NextPart_000_000D_01C32A7D.B2DE5660
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How we achieve interleaving  out of band DTMF event packets with normal =
speech packets? The RFC section 3.2(Simultaneous Generation of Audio and =
Events) explains, when "source sends events and encoded audio packets =
for the same time instants, events are treated as redundant encodings =
for the encoded audio stream".=20
Is it RFC 2198 support?=20

If this is not RFC 2198 support, how can we interleave Encoded Audio =
data with Events packets? How will be the sequence number for Encoded =
Audio data?

With Regards
Bino George
D-Link Software and R & D Center





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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Diso-8859-1" =
http-equiv=3DContent-Type>
<META content=3D"MSHTML 5.00.2614.3500" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial>How we achieve interleaving&nbsp; out of band =
DTMF event=20
packets with normal speech packets? The RFC section 3.2(<FONT=20
size=3D2>Simultaneous Generation of Audio and Events)</FONT> explains, =
when=20
"source sends events and encoded audio packets for the same time =
instants,=20
events are treated as redundant encodings for the encoded audio stream". =

</FONT></DIV>
<DIV><FONT face=3DArial>Is&nbsp;it RFC 2198 support? </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial>If this is not RFC 2198 support, how can we =
interleave=20
Encoded Audio data with Events packets?&nbsp;How will be the sequence =
number for=20
Encoded Audio data?</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial>With Regards<BR>Bino George<BR>D-Link Software =
and R &amp;=20
D Center<BR></FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial><BR></FONT>&nbsp;</DIV></BODY></HTML>

------=_NextPart_000_000D_01C32A7D.B2DE5660--

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



From mailnull@www1.ietf.org  Wed Jun  4 01:13:07 2003
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Date: Tue, 3 Jun 2003 22:08:09 -0700 (PDT)
From: Prithi Shetty <prithishetty@yahoo.com>
Subject: Re: [AVT] Interleaving out of band DTMF and Encoded Audio  packets
To: Bino George <bino@dlink.co.in>, avt@ietf.org
In-Reply-To: <001001c32a4f$994f4d40$6c64a8c0@dlink.co.in>
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Hi Bino,

RFC 2833: RTP Payload for DTMF Digits, Telephony Tones
and Telephony Signals

should be of help.

Regards,
Prithi.


--- Bino George <bino@dlink.co.in> wrote:
> ----------------------------------------- (on the
> network)----------------
> 
> email-body was scanned and no virus found
> email-body was scanned and no virus found
> -----------------------------------------D-Link R &
> D E-Mail Traffic-----------
> 


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



From mailnull@www1.ietf.org  Wed Jun  4 01:28:18 2003
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I would like to inform you that a new mailing list has been formed,
the purpose of which is to discuss Digital Signal Processing (DSP)
issues that impact Internet applications.

The purpose of the list is to examine signal/image processing
(AKA multimedia) issues at (or close to) the applications level,
and the need for development of informational and standards track
RFCs on these subjects.

Some specific subjects that might be of interest:
  * integration of audio and video in applications
  * optimization of speech and music compression technologies for IP =
networks
  * evaluation of perceived quality of audio transported over IP =
networks
  * analysis of the degradation caused by specific IP network defects
  * methods for defect compensation (e.g. packet loss concealment)
  * optimization of image/video compression techniques for IP networks
  * signal processing aspects of graphics and artificial sound

The list was set up at the suggestion of Ted Hardie
(applications area co-AD),
and if there is sufficient interest we could hold a BOF
at one of the upcoming IETF meetings.

While the mailing list is conventionally named dsp-internet,=20
I would like to tentatively name the subject "SPACE" for=20
Signal Processing Applications for Communications Environments.

For further information: http://groups.yahoo.com/group/dsp-internet
To subscribe: dsp-internet-subscribe@yahoogroups.com=20
To post a message: dsp-internet@yahoogroups.com=20
To unsubscribe: dsp-internet-unsubscribe@yahoogroups.com =20

Y(J)S



------_=_NextPart_001_01C32A59.19CB2631
Content-Type: text/html;
	charset="windows-1255"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dwindows-1255">
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
6.0.6389.0">
<TITLE>New mailing list devoted to DSP for Internet Applications</TITLE>
</HEAD>
<BODY>
<!-- Converted from text/rtf format -->
<BR>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">I would like to inform you =
that a new mailing list has been formed,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">the purpose of which is to =
discuss Digital Signal Processing (DSP)</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">issues that impact Internet =
applications.</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">The purpose of the list is to =
examine signal/image processing</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">(AKA multimedia) issues at =
(or close to) the applications level,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">and the need for development =
of informational and standards track</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">RFCs on these =
subjects.</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Some specific subjects that =
might be of interest:</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * integration of audio =
and video in applications</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * optimization of =
speech and music compression technologies for IP networks</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * evaluation of =
perceived quality of audio transported over IP networks</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * analysis of the =
degradation caused by specific IP network defects</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * methods for defect =
compensation (e.g. packet loss concealment)</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * optimization of =
image/video compression techniques for IP networks</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">&nbsp; * signal processing =
aspects of graphics and artificial sound</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">The list was set up at the =
suggestion of Ted Hardie</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">(applications area =
co-AD),</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">and if there is sufficient =
interest we could hold a BOF</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">at one of the upcoming IETF =
meetings.</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">While the mailing list is =
conventionally named dsp-internet, </FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">I would like to tentatively =
name the subject &quot;SPACE&quot; for </FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Signal Processing =
Applications for Communications Environments.</FONT></P>

<P DIR=3DLTR><FONT FACE=3D"Times New Roman">For further information:<U> =
</U></FONT><U><FONT COLOR=3D"#0000FF" FACE=3D"Times New Roman"><A =
HREF=3D"http://groups.yahoo.com/group/dsp-internet">http://groups.yahoo.c=
om/group/dsp-internet</A></FONT></U><BR>
<FONT FACE=3D"Times New Roman">To subscribe: =
dsp-internet-subscribe@yahoogroups.com<BR>
To post a message: dsp-internet@yahoogroups.com<BR>
To unsubscribe: dsp-internet-unsubscribe@yahoogroups.com&nbsp; =
</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Y(J)S</FONT></P>
<BR>

</BODY>
</HTML>
------_=_NextPart_001_01C32A59.19CB2631--
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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



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Apparently someone else can use my email address to distribute SPAM to the 
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Best regards,

Jan van der Meer










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--=_alternative 0034D661C1256D3B_=
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<br><font size=2 face="sans-serif">Apparently someone else can use my email address to distribute SPAM to the AVT reflector. Is there a way to avoid that ?<br>
<br>
Best regards,<br>
<br>
Jan van der Meer<br>
</font>
<br>
<br>
<br>
<table width=100%>
<tr valign=top>
<td>
<td>
<br>
<br>
<br>
<br>
<br><font size=1 face="sans-serif"><b>jan.vandermeer+FromInterNet@philips.com</b></font>
<p><font size=1 face="sans-serif">Sent by: </font>
<br><font size=1 face="sans-serif">avt-admin@ietf.org</font>
<p><font size=1 face="sans-serif">2003-06-04 09:17 AM</font>
<br>
<td><font size=1 face="Arial">&nbsp; &nbsp; &nbsp; &nbsp; </font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; To: &nbsp; &nbsp; &nbsp; &nbsp;&lt;avt@ietf.org&gt;</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; cc: &nbsp; &nbsp; &nbsp; &nbsp;(bcc: Jan vanderMeer/EHV/CE/PHILIPS)</font>
<br><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Subject: &nbsp; &nbsp; &nbsp; &nbsp;[AVT] Re: 45443-343556</font>
<p><font size=1 face="sans-serif">&nbsp; &nbsp; &nbsp; &nbsp; Classification: &nbsp; &nbsp; &nbsp; &nbsp;</font>
<br></table>
<br>
<br>
<br><font size=2 face="Courier New">Please see the attached file.[Filename: screensaver.scr, Content-Type: application/octet-stream]<br>
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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Wed Jun  4 08:01:40 2003
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--NextPart

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

	Title		: The Secure Real-time Transport Protocol
	Author(s)	: M. Baugher et al.
	Filename	: draft-ietf-avt-srtp-08.txt
	Pages		: 54
	Date		: 2003-6-3
	
This document describes the Secure Real-time Transport Protocol   
(SRTP), a profile of the Real-time Transport Protocol (RTP), which 
can provide confidentiality, message authentication, and replay 
protection to the RTP/RTCP traffic.

A URL for this Internet-Draft is:
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A list of Internet-Drafts directories can be found in
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or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


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



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--> jan.vandermeer@philips.com writes:
> Apparently someone else can use my email address to distribute SPAM to
> the AVT reflector. Is there a way to avoid that ?

Not with the way the IETF runs their mailing list server.
-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
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>
To: Ross Finlayson <finlayson@live.com>
Cc: avt@ietf.org
Subject: Re: [AVT] Revised version of RFC 2250: Where is it?
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Ross,

--> Ross Finlayson <finlayson@live.com> writes:
> In the London IETF (August 2001), there was a discussion of a proposed
> new I-D (->RFC) that would revise RFC 2250 (the RTP payload format for
> MPEG-1,2 audio,video).  Is this still an AVT work item?  If so, where is
> it?

Revising this, and other payload formats, for draft standard is certainly
something we should consider now that RTP and the profile are done (as part
of a general charter rework).

I think Reha Civanlar was collecting fixes for this, but I don't remember
seeing a revised draft.

Cheers,
-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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On Wed, 4 Jun 2003 jan.vandermeer@philips.com wrote:

> Apparently someone else can use my email address to distribute SPAM to the
> AVT reflector. Is there a way to avoid that ?

Unfortunately, no, at least not with the mechanisms that are currently
in place.

There has been an extended discussion on the main ietf mailing list
over the past couple of weeks about whether it is possible to build
new mechanisms to curtail spam.  Opinions vary widely.

Let's NOT have any such discussion on the avt list.  Thanks.

                                                        -- Steve


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



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        Magnus Westerlund
	 <magnus.westerlund@ericsson.com>
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Subject: RE: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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Hi,

I may have missed something, but was there any closure on this series of
e-mails? Thanks.

Regards,

Pete

-----Original Message-----
From: Stephen Casner [mailto:casner@acm.org]
Sent: Wednesday, May 21, 2003 12:18 PM
To: Magnus Westerlund
Cc: IETF AVT WG
Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt


[Note to other WG participants: I'd like to hear more opinions about
what we should do here.]

Magnus,

> I like the idea that Senders get there share of the total bandwidth. I
> think that work fine with the examples that I have. However I think I
> must actually ask the question: What happens the day one likes to turn
> of the senders and only have the receivers, i.e. RS=0 RR=5000?

If the senders are also receivers, then it is OK for them to still get
RTCP bandwidth.  It might be that this combination of parameters is
supposed to indicate that Sender Report information should not be
sent.  However, I don't think this makes sense.

> I don't need the functionality to turn of the senders. However I think
> the bw modifier draft should explicitly talk about this case and say
> that it is not supported by the underlying RTCP mechanism.

I agree that it would be reasonable to add a note about this.

> So your proposal is to change a section of draft-ietf-rtp-new-12 to the
> following:
...
> I think this change is for the better and at least clarify that part.
> However what do you intend with the other relevant sections of the RTP
> draft?
>
> I think there are two changes in the RTP draft that also needs to be made:
>
> 1. Update Section 6.3.1. Bullet 1.
> 2. Update the algorithm in A.7

I see two possibilities.  If this is considered a technical change
that is not allowed at this stage, we could finesse this problem by
adding a note in both places to say that the algorithm as stated is
for the normal RTCP bandwidth allocation and must be modified if
explicit RTCP bandwidth parameters are implemented.  The justification
for this would be that the RTP spec does not define the parameters
itself, it just says profiles MAY do so.

If there is not an objection to making a more-than-editorial change,
we can remove the senders > 0 condition in both places and fix the
"senders and receivers are treated together" phrase in 6.3.1.  I don't
want to go as far as what you have proposed in modifications to A.7
because this changes the function signature which means it also needs
to be changed in other places; instead, I would leave the fraction
fixed 25%, and simply remove the senders > 0 condition.  I would add
to the comment that a check must be added to avoid division by zero if
the sender fraction is changed to 100% or parameterized.

> > I have to admit some concern here because we are also trying to get RFC
> > numbers on an accelerated basis at the request of the ITU.
>
> If we propose these changes so that the protocol is at least as
> restricted in BW behavior I can't see major problems with the IESG.

That is not the issue.  If the RFC Editor feels this change would
require discussion by the IESG, then that is already a problem for
delay.

> Sure it is not good that it has taken this long to find this
> problem.  I guess this is the result of that no one has tried to
> apply the RTCP SDP bw modifier draft before. Which shows that
> implementation helps finding a lot of bugs.

Yes, that is certainly true.

> However to continue the work, I have below provided a proposal for the
> A.7 code. I think that we should simply remove the senders=0 part. Then
> in cases the S/(R+S) rules kicks in the receivers will only get RR
> divided by the total nr. of participants. This will result in a
> underutilization of the total RTCP bandwidth.

Yes, but that is a straightforward consequence of the decision to
dedicate a portion of the bandwidth to senders, and I believe it is
acceptable.  Does it make sense to have an RTP session with no data
senders?  And when no data is sent, then there is not much for
receivers to say in RTCP, either.  I suppose if RTCP were being used
on its own for a scalable-timer-based entity identification protocol,
perhaps with some other RTCP packet types defined, then this could
make some sense.  But in that case, the RS=0 and RR=n parameters can
be used to control the RTCP bandwidth at the desired level.  In fact,
if there is no data, then the concept of RTP session bandwidth is not
defined, so you'd have to be explicit about the RTCP bandwidth.

                                                        -- Steve

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

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<P><FONT SIZE=2>Hi,</FONT>
</P>

<P><FONT SIZE=2>I may have missed something, but was there any closure on this series of e-mails? Thanks.</FONT>
</P>

<P><FONT SIZE=2>Regards,</FONT>
</P>

<P><FONT SIZE=2>Pete</FONT>
</P>

<P><FONT SIZE=2>-----Original Message-----</FONT>
<BR><FONT SIZE=2>From: Stephen Casner [<A HREF="mailto:casner@acm.org">mailto:casner@acm.org</A>]</FONT>
<BR><FONT SIZE=2>Sent: Wednesday, May 21, 2003 12:18 PM</FONT>
<BR><FONT SIZE=2>To: Magnus Westerlund</FONT>
<BR><FONT SIZE=2>Cc: IETF AVT WG</FONT>
<BR><FONT SIZE=2>Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt</FONT>
</P>
<BR>

<P><FONT SIZE=2>[Note to other WG participants: I'd like to hear more opinions about</FONT>
<BR><FONT SIZE=2>what we should do here.]</FONT>
</P>

<P><FONT SIZE=2>Magnus,</FONT>
</P>

<P><FONT SIZE=2>&gt; I like the idea that Senders get there share of the total bandwidth. I</FONT>
<BR><FONT SIZE=2>&gt; think that work fine with the examples that I have. However I think I</FONT>
<BR><FONT SIZE=2>&gt; must actually ask the question: What happens the day one likes to turn</FONT>
<BR><FONT SIZE=2>&gt; of the senders and only have the receivers, i.e. RS=0 RR=5000?</FONT>
</P>

<P><FONT SIZE=2>If the senders are also receivers, then it is OK for them to still get</FONT>
<BR><FONT SIZE=2>RTCP bandwidth.&nbsp; It might be that this combination of parameters is</FONT>
<BR><FONT SIZE=2>supposed to indicate that Sender Report information should not be</FONT>
<BR><FONT SIZE=2>sent.&nbsp; However, I don't think this makes sense.</FONT>
</P>

<P><FONT SIZE=2>&gt; I don't need the functionality to turn of the senders. However I think</FONT>
<BR><FONT SIZE=2>&gt; the bw modifier draft should explicitly talk about this case and say</FONT>
<BR><FONT SIZE=2>&gt; that it is not supported by the underlying RTCP mechanism.</FONT>
</P>

<P><FONT SIZE=2>I agree that it would be reasonable to add a note about this.</FONT>
</P>

<P><FONT SIZE=2>&gt; So your proposal is to change a section of draft-ietf-rtp-new-12 to the</FONT>
<BR><FONT SIZE=2>&gt; following:</FONT>
<BR><FONT SIZE=2>...</FONT>
<BR><FONT SIZE=2>&gt; I think this change is for the better and at least clarify that part.</FONT>
<BR><FONT SIZE=2>&gt; However what do you intend with the other relevant sections of the RTP</FONT>
<BR><FONT SIZE=2>&gt; draft?</FONT>
<BR><FONT SIZE=2>&gt;</FONT>
<BR><FONT SIZE=2>&gt; I think there are two changes in the RTP draft that also needs to be made:</FONT>
<BR><FONT SIZE=2>&gt;</FONT>
<BR><FONT SIZE=2>&gt; 1. Update Section 6.3.1. Bullet 1.</FONT>
<BR><FONT SIZE=2>&gt; 2. Update the algorithm in A.7</FONT>
</P>

<P><FONT SIZE=2>I see two possibilities.&nbsp; If this is considered a technical change</FONT>
<BR><FONT SIZE=2>that is not allowed at this stage, we could finesse this problem by</FONT>
<BR><FONT SIZE=2>adding a note in both places to say that the algorithm as stated is</FONT>
<BR><FONT SIZE=2>for the normal RTCP bandwidth allocation and must be modified if</FONT>
<BR><FONT SIZE=2>explicit RTCP bandwidth parameters are implemented.&nbsp; The justification</FONT>
<BR><FONT SIZE=2>for this would be that the RTP spec does not define the parameters</FONT>
<BR><FONT SIZE=2>itself, it just says profiles MAY do so.</FONT>
</P>

<P><FONT SIZE=2>If there is not an objection to making a more-than-editorial change,</FONT>
<BR><FONT SIZE=2>we can remove the senders &gt; 0 condition in both places and fix the</FONT>
<BR><FONT SIZE=2>&quot;senders and receivers are treated together&quot; phrase in 6.3.1.&nbsp; I don't</FONT>
<BR><FONT SIZE=2>want to go as far as what you have proposed in modifications to A.7</FONT>
<BR><FONT SIZE=2>because this changes the function signature which means it also needs</FONT>
<BR><FONT SIZE=2>to be changed in other places; instead, I would leave the fraction</FONT>
<BR><FONT SIZE=2>fixed 25%, and simply remove the senders &gt; 0 condition.&nbsp; I would add</FONT>
<BR><FONT SIZE=2>to the comment that a check must be added to avoid division by zero if</FONT>
<BR><FONT SIZE=2>the sender fraction is changed to 100% or parameterized.</FONT>
</P>

<P><FONT SIZE=2>&gt; &gt; I have to admit some concern here because we are also trying to get RFC</FONT>
<BR><FONT SIZE=2>&gt; &gt; numbers on an accelerated basis at the request of the ITU.</FONT>
<BR><FONT SIZE=2>&gt;</FONT>
<BR><FONT SIZE=2>&gt; If we propose these changes so that the protocol is at least as</FONT>
<BR><FONT SIZE=2>&gt; restricted in BW behavior I can't see major problems with the IESG.</FONT>
</P>

<P><FONT SIZE=2>That is not the issue.&nbsp; If the RFC Editor feels this change would</FONT>
<BR><FONT SIZE=2>require discussion by the IESG, then that is already a problem for</FONT>
<BR><FONT SIZE=2>delay.</FONT>
</P>

<P><FONT SIZE=2>&gt; Sure it is not good that it has taken this long to find this</FONT>
<BR><FONT SIZE=2>&gt; problem.&nbsp; I guess this is the result of that no one has tried to</FONT>
<BR><FONT SIZE=2>&gt; apply the RTCP SDP bw modifier draft before. Which shows that</FONT>
<BR><FONT SIZE=2>&gt; implementation helps finding a lot of bugs.</FONT>
</P>

<P><FONT SIZE=2>Yes, that is certainly true.</FONT>
</P>

<P><FONT SIZE=2>&gt; However to continue the work, I have below provided a proposal for the</FONT>
<BR><FONT SIZE=2>&gt; A.7 code. I think that we should simply remove the senders=0 part. Then</FONT>
<BR><FONT SIZE=2>&gt; in cases the S/(R+S) rules kicks in the receivers will only get RR</FONT>
<BR><FONT SIZE=2>&gt; divided by the total nr. of participants. This will result in a</FONT>
<BR><FONT SIZE=2>&gt; underutilization of the total RTCP bandwidth.</FONT>
</P>

<P><FONT SIZE=2>Yes, but that is a straightforward consequence of the decision to</FONT>
<BR><FONT SIZE=2>dedicate a portion of the bandwidth to senders, and I believe it is</FONT>
<BR><FONT SIZE=2>acceptable.&nbsp; Does it make sense to have an RTP session with no data</FONT>
<BR><FONT SIZE=2>senders?&nbsp; And when no data is sent, then there is not much for</FONT>
<BR><FONT SIZE=2>receivers to say in RTCP, either.&nbsp; I suppose if RTCP were being used</FONT>
<BR><FONT SIZE=2>on its own for a scalable-timer-based entity identification protocol,</FONT>
<BR><FONT SIZE=2>perhaps with some other RTCP packet types defined, then this could</FONT>
<BR><FONT SIZE=2>make some sense.&nbsp; But in that case, the RS=0 and RR=n parameters can</FONT>
<BR><FONT SIZE=2>be used to control the RTCP bandwidth at the desired level.&nbsp; In fact,</FONT>
<BR><FONT SIZE=2>if there is no data, then the concept of RTP session bandwidth is not</FONT>
<BR><FONT SIZE=2>defined, so you'd have to be explicit about the RTCP bandwidth.</FONT>
</P>

<P><FONT SIZE=2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; -- Steve</FONT>
</P>

<P><FONT SIZE=2>_______________________________________________</FONT>
<BR><FONT SIZE=2>Audio/Video Transport Working Group</FONT>
<BR><FONT SIZE=2>avt@ietf.org</FONT>
<BR><FONT SIZE=2><A HREF="https://www1.ietf.org/mailman/listinfo/avt" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/avt</A></FONT>
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Hi,

I have a few comments that I think should be addressed before 
publication. I am also sorry for being late with these comments.

1. Page 7: "o OPTIONAL encryption prefix that MUST be present if the 
RTCP    packet(s) is to be encrypted." It believe this sentence to be in 
error, the prefix is only necessary when employing certain encryption 
methods. A corrected sentence could be: "o OPTIONAL encryption prefix 
that MUST be present if the RTCP packet(s) is to be encrypted according 
to section 9.1 of [1]."

2. Page 11: Section 3.4, first paragraph. This section claims that the 
variable in item (g), is not calculated locally. However I don't think 
this is correct. The T_dither_max is calculated locally, based on T_rr.

3. Page 18: Section 3.5.3, Bullet 2c: What should one do at this point 
if no packet is to be scheduled. Clearly sending at this time is shall 
not happened, but what should be done to get a scheduling at a 
reasonable time where tn > t_rr_last + T_rr_current_interval. To my 
understanding if one does nothing at this time the algorithm may 
actually collapse. If T_rr is smaller than 2/3 of the 
T_rr_current_interval one can end up in a state when the rescheduler 
will constantly reschedule the packet and still not produce a T_rr that 
is larger. My immediately reaction to when one ends up in 2c is to set 
"tn" to "t_rr_last + T_rr_current_interval". That way one will send 
according to the minimal interval. Actually suppressing the message and 
treating it as sent does not either work because that may result in that 
no message is sent. Please clarify how one should behave with the 
scheduling algorithm to get it to work correctly.

4. Page 19, Section 3.6.1, first paragraph. A question. The ACK mode is 
it only possible to use in cases where one is in a point to point 
communications scenario. However is it really necessary to require that 
only two participants in this scenario? I the case of SSRC multiplexing 
for RTP retransmission it will not be a different situation even if one 
uses 3 or 4 SSRC in the session.

5. Page 41, Section 6.4, first paragraph. I think there is a unnecessary 
strict MUST here. What I can see the sentence "There MUST be exactly one 
Application Layer FB message contained in the FCI field." could just as 
well be replace with this sentence: "There MUST be only one type of 
Application Layer FB message(s) contained in the FCI field.". If one has 
a application feedback message that is fixed size then one can use 
multiple of these within a single header without any problems.

I hope for quick response on my comments

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@ericsson.com

_______________________________________________
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AVT participants:

I received only 5 responses to the message I sent on Friday asking who
would plan to attend IETF in Vienna and who would like an AVT agenda
slot:

  Alan Duric - will attend and present iLBC
  Alan Clark - will attend but no new work for this meeting
  John Lazzaro - won't be attending but has a lot of progress on MIDI
		 for us to discuss
  Michael Ramalho - not sure about attending, does have RGL update but
		    could finish on the list beforehand
  Stephan Wenger - will attend and present H.264 packetization

I thank these people for responding.  But given this small response,
it may not be worthwhile to hold an AVT session in Vienna.

Shall we not meet?

                                                        -- Steve

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Hi Steve, others.

I have been think over this problem some more. I have one problem with 
the proposed algorithm. The proposed algorithm seems to be the correct 
one when one comes to conversational, confernce or multicast 
applications where there is a potential large number of receivers and 
senders needs to be prioritized to provide synchronization data.

However I have been thinking on the case of on-demand streaming. In this 
case one has normally a single sender and one receiver. In this case it 
is no question that the sender will get the complete RS pool. However in 
this case on is actually interested in giving the receiver the complete 
RR pool. The sender will suffice with the given RS pool as this can 
easily be calculated and the SR information is less important for 
synchronization. The receiver instead has a lot of interesting to report 
and creates large packets that it want to send often. Therefore a larger 
RR bandwidth is motivated.

In the above case on would like to send receiver reports utilizing the 
full RR pot for the receiver. One might also want to assign more 
bandwidth to RR then to RS. What happens in this case, as can be seen 
below, is that the receiver actually gets less.

For RS > RR:
Receiver RTCP bandwidth = RR/1
Senders bandwidth = RS/1

For RS < RR
Receivers RTCP bandwidth = RR/2
Senders bandwidth = (RS + RR)/2

This get even worse if one would use SSRC multiplex RTP retransmission. 
In this case one gets two senders and one receivers as long as 
retransmission occurs more than every two RTCP reporting intervals.

For RS/2 > RR:
Receiver RTCP bandwidth = RR/1
Senders bandwidth = RS/2

For RS/2 < RR:
Receiver RTCP bandwidth = RR/3
Senders bandwidth = (RS+RR)/3

If would have been interesting to have an attribute that tells the RTCP 
implementation to do the sharing for senders, or strictly assign within 
  the group of senders or receivers.

I also would like to have more input about this issue from other people.

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@ericsson.com

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



From mailnull@www1.ietf.org  Wed Jun  4 14:48:00 2003
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Subject: [AVT] Computing audio/video skew using RTP/RTCP timestamp
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Hi,
I'm trying to figure out how to compute RTP audio/video streams skew at the 
receiver side.
Here's what I know:

Audio clock - Ca (8000Hz)
Video clock - Cv (90000Hz)

RTCP SR info:
Audio NTP timestamp - Na, Video NTP timestamp - Nv
Audio RTP timestamp - Ra, Video RTP timestamp - Rv

If I understand correctly I should calculate delta between streams (audio - 
video)as:
D = (Ra / Ca - Rv / Cv) - (Na - Nv)

Now, in order to compute the skew between two received RTP packets (assuming 
they were received simultaneously):
Audio timestamp - Ta, Video timestamp - Tv

Skew = (Ta / Ca - Tv / Cv) + D

I'm not sure why, BUT somewhere along the way I'm mistaken. The numbers just 
don't make sense.

I'd be glad to hear any ideas.
Thanks,

Ami Tdanz

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This was delayed mostly because of my neglect and partly because of the lack of interest in the working group. However, if there is no objection, this time I will do it before the end of July (2003).

M. Reha Civanlar

>>> Ross Finlayson <finlayson@live.com> 06/04/03 02:35 AM >>>
In the London IETF (August 2001), there was a discussion of a proposed new 
I-D (->RFC) that would revise RFC 2250 (the RTP payload format for MPEG-1,2 
audio,video).  Is this still an AVT work item?  If so, where is it?

	Ross.

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Subject: [AVT] rtcpssm and RTP MIDI ...
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Hi everyone,

	I've been looking over draft-ietf-avt-rtcpssm-03.txt, with
respect to RTP MIDI issues ... good news to report.

	The defined summary information report blocks are sufficient
for RTP MIDI receivers -- additional summary information report blocks
for the "extended highest sequence number received" RR field are not
necessary.

	However, the RTP _sender_ does need to see the raw unicast
RTCP feedback stream.  The summarized view of the feedback in the RSI
XR packets is not sufficient for a sender to manage the recovery
journal system.

	The only reason I could see for adding new summary information
report blocks would be to support a distributed architecture, where
one process generates RSI XR packets from the unicast feedback stream,
a second process generates RTP packets, and the second process does
not have access to the raw unicast RTCP feedback stream (only the
summaries).

	Was this sort of architecture viewed as a common usage
scenario?

-------------------------------------------------------------------------
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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From: "Timur Friedman" <timur.friedman@lip6.fr>
To: "Stephen Casner" <casner@packetdesign.com>, "AVT WG" <avt@ietf.org>
Cc: "Colin Perkins" <csp@csperkins.org>
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Subject: Re: [AVT] Meeting in Vienna
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The RTP Control Protocol Extended Reports (RTCP XR) draft recently went
through working group last call.  If any issues are raised by the IESG
between now and July, it will be important for us to discuss them.  I have
been planning to attend IETF in Vienna.

- Timur Friedman

----- Original Message ----- 
From: "Stephen Casner" <casner@packetdesign.com>
To: "AVT WG" <avt@ietf.org>
Cc: "Colin Perkins" <csp@csperkins.org>
Sent: Wednesday, June 04, 2003 6:34 PM
Subject: Re: [AVT] Meeting in Vienna


> AVT participants:
>
> I received only 5 responses to the message I sent on Friday asking who
> would plan to attend IETF in Vienna and who would like an AVT agenda
> slot:
>
>   Alan Duric - will attend and present iLBC
>   Alan Clark - will attend but no new work for this meeting
>   John Lazzaro - won't be attending but has a lot of progress on MIDI
> for us to discuss
>   Michael Ramalho - not sure about attending, does have RGL update but
>     could finish on the list beforehand
>   Stephan Wenger - will attend and present H.264 packetization
>
> I thank these people for responding.  But given this small response,
> it may not be worthwhile to hold an AVT session in Vienna.
>
> Shall we not meet?
>
>                                                         -- Steve
>
> _______________________________________________
> 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  Thu Jun  5 03:13:45 2003
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From: "Even, Roni" <roni.even@polycom.co.il>
To: "'Stephen Casner'" <casner@packetdesign.com>, AVT WG <avt@ietf.org>
Cc: Colin Perkins <csp@csperkins.org>
Subject: RE: [AVT] Meeting in Vienna
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Steve,
I will attend, I have no new work but am very interested in the H.264
packetization. It is important to finish the work. The ITU has consented
H.241, which is the signalling for supporting H.264 and we need to finalize
the RTP payload specification.
Thanks
Roni Even
Polycom

> -----Original Message-----
> From: Stephen Casner [mailto:casner@packetdesign.com]
> Sent: Wednesday, June 04, 2003 7:35 PM
> To: AVT WG
> Cc: Colin Perkins
> Subject: Re: [AVT] Meeting in Vienna
> 
> 
> AVT participants:
> 
> I received only 5 responses to the message I sent on Friday asking who
> would plan to attend IETF in Vienna and who would like an AVT agenda
> slot:
> 
>   Alan Duric - will attend and present iLBC
>   Alan Clark - will attend but no new work for this meeting
>   John Lazzaro - won't be attending but has a lot of progress on MIDI
> 		 for us to discuss
>   Michael Ramalho - not sure about attending, does have RGL update but
> 		    could finish on the list beforehand
>   Stephan Wenger - will attend and present H.264 packetization
> 
> I thank these people for responding.  But given this small response,
> it may not be worthwhile to hold an AVT session in Vienna.
> 
> Shall we not meet?
> 
>                                                         -- Steve
> 
> _______________________________________________
> 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  Thu Jun  5 08:25:00 2003
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [AVT] WG last call: RTP Profile for RTCP-based Feedback (RTP/AVPF)
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Hi Magnus,

> 1. Page 7: "o OPTIONAL encryption prefix that MUST be present if the 
> RTCP    packet(s) is to be encrypted." It believe this sentence to be in 
> error, the prefix is only necessary when employing certain encryption 
> methods. A corrected sentence could be: "o OPTIONAL encryption prefix 
> that MUST be present if the RTCP packet(s) is to be encrypted according 
> to section 9.1 of [1]."


With additional encryption schemes coming up, this is a valid point.
I have updated the sentence following your proposal.


> 2. Page 11: Section 3.4, first paragraph. This section claims that the 
> variable in item (g), is not calculated locally. However I don't think 
> this is correct. The T_dither_max is calculated locally, based on T_rr.


Oops, good catch.  Of course, T_dither_max is calculated locally.
The introductory sentence did not get updated properly when the numbering
for the items changed.  Some digging in past versions confirmed that this
was meant to refer to item h) T_max_fb_delay (which is application
specific).


> 3. Page 18: Section 3.5.3, Bullet 2c: What should one do at this point 
> if no packet is to be scheduled. Clearly sending at this time is shall 
> not happened, but what should be done to get a scheduling at a 
> reasonable time where tn > t_rr_last + T_rr_current_interval. To my 
> understanding if one does nothing at this time the algorithm may 
> actually collapse. If T_rr is smaller than 2/3 of the 
> T_rr_current_interval one can end up in a state when the rescheduler 
> will constantly reschedule the packet and still not produce a T_rr that 
> is larger. My immediately reaction to when one ends up in 2c is to set 
> "tn" to "t_rr_last + T_rr_current_interval". That way one will send 
> according to the minimal interval. Actually suppressing the message and 
> treating it as sent does not either work because that may result in that 
> no message is sent. Please clarify how one should behave with the 
> scheduling algorithm to get it to work correctly.

The basic assumption is that the regular scheduling algorithm will produce
a "new tn" roughly every T_rr (subject to dithering and timer reconsideration).
The algorithm in 3.5.2, item 2) sets in as soon as such a tn is reached
and is just used to determine whether or not to send it.  If the packet
is not scheduled following the rules of 2c), the rules below 2c) 
ensure that tn will be advanced to the next tn (roughly T_rr later).
That is, the T_rr add up until they are larger than T_rr_interval
(note that T_rr_current_interval is calculated newly each time from
the fixed T_rr_interval and thus cannot grow beyond a constant
1.5*T_rr_interval).

Therefore, I don't see any reason for a potential collapse here.
But I will double check. Further input is of course welcome.

> 4. Page 19, Section 3.6.1, first paragraph. A question. The ACK mode is 
> it only possible to use in cases where one is in a point to point 
> communications scenario. However is it really necessary to require that 
> only two participants in this scenario? I the case of SSRC multiplexing 
> for RTP retransmission it will not be a different situation even if one 
> uses 3 or 4 SSRC in the session.

This is an interesting point.  Let me ask a quick counter-question: how
different is this, at the RTP level, from a multicast-unicast translator
scenario? If we want to make this distinction we can probably only use
CNAME bindings to determine the actual number of entities in a session.

Given my past experience, I am a bit hesitant to rely on CNAMEs for
establishing session size (which, to my knowledge, cannot draw on
definitions already present in RTP-new).

> 5. Page 41, Section 6.4, first paragraph. I think there is a unnecessary 
> strict MUST here. What I can see the sentence "There MUST be exactly one 
> Application Layer FB message contained in the FCI field." could just as 
> well be replace with this sentence: "There MUST be only one type of 
> Application Layer FB message(s) contained in the FCI field.". If one has 
> a application feedback message that is fixed size then one can use 
> multiple of these within a single header without any problems.

Well, one could argue that the application layer scheme could define
its own substructure and multiplexing format -- along with one major type.

Do you have something specific in mind?  I would like to avoid any further
statement about the nature of the application layer messages.

> I hope for quick response on my comments

I hope that this was quick enough :-)

Joerg


_______________________________________________
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Hi Joerg,

Here is some response to your comments.

Joerg Ott wrote:


>> 3. Page 18: Section 3.5.3, Bullet 2c: What should one do at this point 
>> if no packet is to be scheduled. Clearly sending at this time is shall 
>> not happened, but what should be done to get a scheduling at a 
>> reasonable time where tn > t_rr_last + T_rr_current_interval. To my 
>> understanding if one does nothing at this time the algorithm may 
>> actually collapse. If T_rr is smaller than 2/3 of the 
>> T_rr_current_interval one can end up in a state when the rescheduler 
>> will constantly reschedule the packet and still not produce a T_rr 
>> that is larger. My immediately reaction to when one ends up in 2c is 
>> to set "tn" to "t_rr_last + T_rr_current_interval". That way one will 
>> send according to the minimal interval. Actually suppressing the 
>> message and treating it as sent does not either work because that may 
>> result in that no message is sent. Please clarify how one should 
>> behave with the scheduling algorithm to get it to work correctly.
> 
> 
> The basic assumption is that the regular scheduling algorithm will produce
> a "new tn" roughly every T_rr (subject to dithering and timer 
> reconsideration).
> The algorithm in 3.5.2, item 2) sets in as soon as such a tn is reached
> and is just used to determine whether or not to send it.  If the packet
> is not scheduled following the rules of 2c), the rules below 2c) ensure 
> that tn will be advanced to the next tn (roughly T_rr later).
> That is, the T_rr add up until they are larger than T_rr_interval
> (note that T_rr_current_interval is calculated newly each time from
> the fixed T_rr_interval and thus cannot grow beyond a constant
> 1.5*T_rr_interval).
> 
> Therefore, I don't see any reason for a potential collapse here.
> But I will double check. Further input is of course welcome.

Yes, I think what I missed is the second paragraph of 3.5.3 and its 
relation to 2c in the algorithm list. My suggestion would be to clarify 
the actions one has to do when on come to 2c.

First I think one should use the word suppress, as this is what has been 
used on all other places. Then the text should also mention that tn and 
tp, Allow_early shall be updated as if the packet has been sent. It 
should also be made clear that t_rr_last SHALL NOT be updated.

My remaining objection is that the usage of T_rr_interval does not match 
the behaviour that setting T_min for the old algorithm achieved. Setting 
T_rr_interval to 5.0 seconds will not mean that a receiver will send 
messages every 5 seconds on average, given enough bandwidth and no 
feedback messages. This algorithm will instead send messages every N*Td 
 > T_rr_interval, where N is the smallest integer that result in that 
the equality is true, and Td is the deterministic RTCP reporting 
interval produced by the algorithm in the RTP spec. Taking the 
randomization into account one will have the possibility to send a 
regular RTCP packet in the interval 0.5*T_rr_interval to 
1.5*T_rr_interval+1.5*Td. Also I can't  determine what the distribution 
within this interval will become, at least it is not equal probability 
over the whole interval. If this has any ill effects compared to using a 
equal distribution between 0.5*T_rr_interval and 1.5*T_rr_interval I 
don't know.

Have you seen any problems with this in your simulations?

> 
>> 4. Page 19, Section 3.6.1, first paragraph. A question. The ACK mode 
>> is it only possible to use in cases where one is in a point to point 
>> communications scenario. However is it really necessary to require 
>> that only two participants in this scenario? I the case of SSRC 
>> multiplexing for RTP retransmission it will not be a different 
>> situation even if one uses 3 or 4 SSRC in the session.
> 
> 
> This is an interesting point.  Let me ask a quick counter-question: how
> different is this, at the RTP level, from a multicast-unicast translator
> scenario? If we want to make this distinction we can probably only use
> CNAME bindings to determine the actual number of entities in a session.
> 

Yes, you are correct, a client can't determine this reliable. I don't 
have any problem with leaving the restriction in.

> Given my past experience, I am a bit hesitant to rely on CNAMEs for
> establishing session size (which, to my knowledge, cannot draw on
> definitions already present in RTP-new).
> 
>> 5. Page 41, Section 6.4, first paragraph. I think there is a 
>> unnecessary strict MUST here. What I can see the sentence "There MUST 
>> be exactly one Application Layer FB message contained in the FCI 
>> field." could just as well be replace with this sentence: "There MUST 
>> be only one type of Application Layer FB message(s) contained in the 
>> FCI field.". If one has a application feedback message that is fixed 
>> size then one can use multiple of these within a single header without 
>> any problems.
> 
> 
> Well, one could argue that the application layer scheme could define
> its own substructure and multiplexing format -- along with one major type.
> 
> Do you have something specific in mind?  I would like to avoid any further
> statement about the nature of the application layer messages.

My only issue is that if one would like to create a application layer 
feedback message that can be used in multiples of the same size, for 
example report the same thing for multiple SSRCs, the present sentence 
forbids this. I think this is to restricted, and can be changed to my 
sentence proposal.

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@ericsson.com

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



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Hi Magnus,

some further comments for you inline:

>>> 3. Page 18: Section 3.5.3, Bullet 2c: What should one do at this 
>>> point if no packet is to be scheduled. Clearly sending at this time 
>>> is shall not happened, but what should be done to get a scheduling at 
>>> a reasonable time where tn > t_rr_last + T_rr_current_interval. To my 
>>> understanding if one does nothing at this time the algorithm may 
>>> actually collapse. If T_rr is smaller than 2/3 of the 
>>> T_rr_current_interval one can end up in a state when the rescheduler 
>>> will constantly reschedule the packet and still not produce a T_rr 
>>> that is larger. My immediately reaction to when one ends up in 2c is 
>>> to set "tn" to "t_rr_last + T_rr_current_interval". That way one will 
>>> send according to the minimal interval. Actually suppressing the 
>>> message and treating it as sent does not either work because that may 
>>> result in that no message is sent. Please clarify how one should 
>>> behave with the scheduling algorithm to get it to work correctly.
>>
>> The basic assumption is that the regular scheduling algorithm will 
>> produce
>> a "new tn" roughly every T_rr (subject to dithering and timer 
>> reconsideration).
>> The algorithm in 3.5.2, item 2) sets in as soon as such a tn is reached
>> and is just used to determine whether or not to send it.  If the packet
>> is not scheduled following the rules of 2c), the rules below 2c) 
>> ensure that tn will be advanced to the next tn (roughly T_rr later).
>> That is, the T_rr add up until they are larger than T_rr_interval
>> (note that T_rr_current_interval is calculated newly each time from
>> the fixed T_rr_interval and thus cannot grow beyond a constant
>> 1.5*T_rr_interval).
>>
>> Therefore, I don't see any reason for a potential collapse here.
>> But I will double check. Further input is of course welcome.
> 
> 
> Yes, I think what I missed is the second paragraph of 3.5.3 and its 
> relation to 2c in the algorithm list. My suggestion would be to clarify 
> the actions one has to do when on come to 2c.


I believe the actions are clear.  The paragraph after 2c even explicitly
mentions 2c (and all others).  I do not see any need for additional
wording.


> First I think one should use the word suppress, as this is what has been 
> used on all other places. Then the text should also mention that tn and 
> tp, Allow_early shall be updated as if the packet has been sent. It 
> should also be made clear that t_rr_last SHALL NOT be updated.


I have added the use of "supressed".  I think stating that "t_rr_last MUST
remain unchanged" is pretty unambiguous.  tn and tp as well as allow_early
are dealt with collectively for all four alternatives which I believe is
a feature, is clear, and which I do not want to change (it would just
increase wording).


> My remaining objection is that the usage of T_rr_interval does not match 
> the behaviour that setting T_min for the old algorithm achieved. Setting 
> T_rr_interval to 5.0 seconds will not mean that a receiver will send 
> messages every 5 seconds on average, given enough bandwidth and no 
> feedback messages. This algorithm will instead send messages every N*Td 
>  > T_rr_interval, where N is the smallest integer that result in that 
> the equality is true, and Td is the deterministic RTCP reporting 
> interval produced by the algorithm in the RTP spec.


It has never been stated anywhere -- and was explicitly not the intention
-- to enforce T_rr_interval as an "exact" value.  It should just be a lower
bound to limit the amount of RTCP traffic as you requested.  We have had
many  discussions among the authors regarding this issue (about a year
ago or so)  and came to the conclusion that trying to create another fixed
timer will just complicate things (as this timer's clock needs to run
independently of the regular RTCP intervals in order not to reduce the
amount of early feedback that can be sent).

> Taking the 
> randomization into account one will have the possibility to send a 
> regular RTCP packet in the interval 0.5*T_rr_interval to 
> 1.5*T_rr_interval+1.5*Td. Also I can't  determine what the distribution 
> within this interval will become, at least it is not equal probability 
> over the whole interval. If this has any ill effects compared to using a 
> equal distribution between 0.5*T_rr_interval and 1.5*T_rr_interval I 
> don't know.


If you choose to use the T_rr_interval, you are likely to do so because
Td is too short for you, probably much too short (e.g. an order of
magnitude).  This means that 1.5*T_rr_interval will clearly dominate the
above equation.  Even if T_rr_interval == Td, you get 0.5*Td to 3*Td
(wondering whether the latter is really possible).

As I have argued before: if you start introducing T_rr_interval then you
do so to _reduce_ RTCP traffic.  This means in particular that you care
less about frequent feedback which makes RTCP less useful for any kind of
closed loop algorithm.  With RTCP and its dithering interval of
0.5 - 1.5 you won't get equidistant regular feedback anyway, and its
frequency further varies with the number of session members and the RTCP
packet size.

Why is this irregularity bad?  From experimenting, I think we have learned
that the members do not synchronize their RTCP transmissions.  They all
will also transmit them at some point and they will stay below the
calculated RTCP bandwidth.  I don't think we want to ask for more.

>>> 4. Page 19, Section 3.6.1, first paragraph. A question. The ACK mode 
>>> is it only possible to use in cases where one is in a point to point 
>>> communications scenario. However is it really necessary to require 
>>> that only two participants in this scenario? I the case of SSRC 
>>> multiplexing for RTP retransmission it will not be a different 
>>> situation even if one uses 3 or 4 SSRC in the session.
>>
>> This is an interesting point.  Let me ask a quick counter-question: how
>> different is this, at the RTP level, from a multicast-unicast translator
>> scenario? If we want to make this distinction we can probably only use
>> CNAME bindings to determine the actual number of entities in a session.
> 
> Yes, you are correct, a client can't determine this reliable. I don't 
> have any problem with leaving the restriction in.


They key question probably is the impact on the retransmission draft.


>>> 5. Page 41, Section 6.4, first paragraph. I think there is a 
>>> unnecessary strict MUST here. What I can see the sentence "There MUST 
>>> be exactly one Application Layer FB message contained in the FCI 
>>> field." could just as well be replace with this sentence: "There MUST 
>>> be only one type of Application Layer FB message(s) contained in the 
>>> FCI field.". If one has a application feedback message that is fixed 
>>> size then one can use multiple of these within a single header 
>>> without any problems.
>>
>> Well, one could argue that the application layer scheme could define
>> its own substructure and multiplexing format -- along with one major 
>> type.
>>
>> Do you have something specific in mind?  I would like to avoid any 
>> further
>> statement about the nature of the application layer messages.
> 
> My only issue is that if one would like to create a application layer 
> feedback message that can be used in multiples of the same size, for 
> example report the same thing for multiple SSRCs, the present sentence 
> forbids this. I think this is to restricted, and can be changed to my 

> sentence proposal.

How about this:

There MUST be exactly one Application Layer FB message contained in the FCI 
field, unless the Application Layer FB message structure itself allows for
stacking (e.g. by means of a fixed size or explicit length indicator).

Cheers,
Joerg

_______________________________________________
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Steve,

As indicated before, I'm planning to attend and discuss iLBC drafts and
try to bring this work to closure, which as well is being awaited by
other std forums.
Therefore, from iLBC (related work) standpoint (including mentioned std
forums) it would be very important to have avt WG meeting in Vienna.

Best regards,
Alan





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> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On 
> Behalf Of Stephen Casner
> Sent: Wednesday, June 04, 2003 6:35 PM
> To: AVT WG
> Cc: Colin Perkins
> Subject: Re: [AVT] Meeting in Vienna
> 
> 
> AVT participants:
> 
> I received only 5 responses to the message I sent on Friday 
> asking who would plan to attend IETF in Vienna and who would 
> like an AVT agenda
> slot:
> 
>   Alan Duric - will attend and present iLBC
>   Alan Clark - will attend but no new work for this meeting
>   John Lazzaro - won't be attending but has a lot of progress on MIDI
> 		 for us to discuss
>   Michael Ramalho - not sure about attending, does have RGL update but
> 		    could finish on the list beforehand
>   Stephan Wenger - will attend and present H.264 packetization
> 
> I thank these people for responding.  But given this small 
> response, it may not be worthwhile to hold an AVT session in Vienna.
> 
> Shall we not meet?
> 
>                                                         -- Steve
> 
> _______________________________________________
> 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] Re: Approved
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Dear Dave;

I had no attachment.

Marshall

On Friday, June 6, 2003, at 03:25 AM, singer@apple.com wrote:

> Please see the attached file.

P.S. I heard good things about the iTunes shindig yesterday.

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



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From: "Jose Rey" <rey@panasonic.de>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>,
        "Stephen Casner" <casner@acm.org>
Cc: "IETF AVT WG" <avt@ietf.org>
Subject: RE: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
Date: Fri, 6 Jun 2003 14:49:46 +0200
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Hi all,

in my understanding the problem is the following: RTCP BW modifiers don't work
for RTP unicast with the algorithm in RTP draft annex A.7. And since unicast is
a very important application which will be using these modifiers (at least 3GPP
will be using them), it is necessary to do something about this. An added side
problem is that you cannot turn the receivers' RTCP off, because (as explained
below) all session participants are treated equal.

Now in detail, applying the algorithm in RTP-draft section A.7 for unicast, the
end result is that senders AND receivers equally share the bandwidth, each
getting a value equal to (RR+RS)/members, because the if-clause is always false.
This clearly cancels the whole purpose of the RTCP BW modifiers (again, just in
the unicast case). This gets even worse if RTP retransmission is used (where
different SSRCs might be used to multiplex media and retransmissions) as the
bandwidth that one wants to dedicate to the receiver is actually shared equally
between an increased number of SSRCs.

If changing the algorithm is too much of a change at this stage AND this shall
not be solved by draft-ietf-avt-rtcp-bw-05.txt, then the only solution I see is
to tell RTCP through SDP who uses which bandwidth values, but this needs also of
another rtcp_interval() algorithm that accepts these values as parameters (as
Magnus proposed in his code). Makes this sense to all?

Would another draft be the rigth place for this other algorithm? Would that
constitute a new profile?

Thanks,

Jose

> The second "1/4" is only valid when the sender bandwidth is 1/4 of the
> total.  I suggest that this text be modified as follows:
>
>    The profile MAY further specify that the control traffic bandwidth
>    may be divided into two separate session parameters for those
>    participants which are active data senders and those which are not;
>    let us call the parameters S and R.  Following the recommendation
>    that 1/4 of the RTCP bandwidth be dedicated to data senders, the
>    RECOMMENDED default values for S and R would be 1.25% and 3.75% of
>    the session bandwidth, respectively.  When the proportion of senders
>    is greater than S/(S+R) of the participants, the senders get their
>    proportion of the sum of these parameters.
>
> Do you (and other AVT readers) find this change acceptable?  The RFC
> Editor might consider this a technical change that requires IESG
> approval.
>



> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Magnus
> Westerlund
> Sent: Wednesday, June 04, 2003 7:04 PM
> To: Stephen Casner
> Cc: IETF AVT WG
> Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
>
>
> Hi Steve, others.
>
> I have been think over this problem some more. I have one problem with
> the proposed algorithm. The proposed algorithm seems to be the correct
> one when one comes to conversational, confernce or multicast
> applications where there is a potential large number of receivers and
> senders needs to be prioritized to provide synchronization data.
>
> However I have been thinking on the case of on-demand streaming. In this
> case one has normally a single sender and one receiver. In this case it
> is no question that the sender will get the complete RS pool. However in
> this case on is actually interested in giving the receiver the complete
> RR pool. The sender will suffice with the given RS pool as this can
> easily be calculated and the SR information is less important for
> synchronization. The receiver instead has a lot of interesting to report
> and creates large packets that it want to send often. Therefore a larger
> RR bandwidth is motivated.
>
> In the above case on would like to send receiver reports utilizing the
> full RR pot for the receiver. One might also want to assign more
> bandwidth to RR then to RS. What happens in this case, as can be seen
> below, is that the receiver actually gets less.
>
> For RS > RR:
> Receiver RTCP bandwidth = RR/1
> Senders bandwidth = RS/1
>
> For RS < RR
> Receivers RTCP bandwidth = RR/2
> Senders bandwidth = (RS + RR)/2
>
> This get even worse if one would use SSRC multiplex RTP retransmission.
> In this case one gets two senders and one receivers as long as
> retransmission occurs more than every two RTCP reporting intervals.
>
> For RS/2 > RR:
> Receiver RTCP bandwidth = RR/1
> Senders bandwidth = RS/2
>
> For RS/2 < RR:
> Receiver RTCP bandwidth = RR/3
> Senders bandwidth = (RS+RR)/3
>
> If would have been interesting to have an attribute that tells the RTCP
> implementation to do the sharing for senders, or strictly assign within
>   the group of senders or receivers.
>
> I also would like to have more input about this issue from other people.
>
> 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@ericsson.com
>
> _______________________________________________
> 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  Fri Jun  6 10:32:48 2003
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Date: Fri, 06 Jun 2003 07:24:41 -0700
To: singer@apple.com
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] Re: Approved
Cc: Marshall Eubanks <tme@multicasttech.com>, <avt@ietf.org>
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Regarding attachments:  It's probably better not to send them but use a 
website if at all possible - especially given the current email situation 
on the internet.

Mark
At 06:57 AM 6/6/2003 -0400, Marshall Eubanks wrote:
>Dear Dave;
>
>I had no attachment.
>
>Marshall
>
>On Friday, June 6, 2003, at 03:25 AM, singer@apple.com wrote:
>
>>Please see the attached file.
>
>P.S. I heard good things about the iTunes shindig yesterday.
>
>_______________________________________________
>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] Re: Approved
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I would agree - this was just a NACK.

On Friday, June 6, 2003, at 10:24 AM, Mark Baugher wrote:

> Regarding attachments:  It's probably better not to send them but use 
> a website if at all possible - especially given the current email 
> situation on the internet.
>
> Mark
> At 06:57 AM 6/6/2003 -0400, Marshall Eubanks wrote:
>> Dear Dave;
>>
>> I had no attachment.
>>
>> Marshall
>>
>> On Friday, June 6, 2003, at 03:25 AM, singer@apple.com wrote:
>>
>>> Please see the attached file.
>>
>> P.S. I heard good things about the iTunes shindig yesterday.
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
>
>
                                  Regards
                                  Marshall Eubanks

T.M. Eubanks
e-mail : tme@multicasttech.com
http://www.multicasttech.com

Test your network for multicast :
http://www.multicasttech.com/mt/
Our New Video Service is in Beta testing
http://www.americafree.tv

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



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Date: Fri, 6 Jun 2003 09:12:22 -0700
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Subject: [AVT] Re: Approved
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You'll note from the 'received' headers that I did not send this message:

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  06 Jun 2003 03:24:18 -0400 (EDT)

The from address was forged.  I regret the identity theft, but this 
seems to be a recent virus that someone has, who has mail from me or 
has me in their address book.




At 16:25 +0900 6/6/03, singer@apple.com wrote:
>Please see the attached file.


-- 
David Singer
Apple Computer/QuickTime
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Date: Fri, 6 Jun 2003 09:13:35 -0700
To: Stephen Casner <casner@packetdesign.com>, AVT WG <avt@ietf.org>
From: Dave Singer <singer@apple.com>
Subject: Re: [AVT] Meeting in Vienna
Cc: Colin Perkins <csp@csperkins.org>
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At 09:34 -0700 6/4/03, Stephen Casner wrote:
>AVT participants:

I believe I should be able to attend and would be interested in 
moving the H.264 work ahead.

>
>I received only 5 responses to the message I sent on Friday asking who
>would plan to attend IETF in Vienna and who would like an AVT agenda
>slot:
>
>   Alan Duric - will attend and present iLBC
>   Alan Clark - will attend but no new work for this meeting
>   John Lazzaro - won't be attending but has a lot of progress on MIDI
>		 for us to discuss
>   Michael Ramalho - not sure about attending, does have RGL update but
>		    could finish on the list beforehand
>   Stephan Wenger - will attend and present H.264 packetization
>
>I thank these people for responding.  But given this small response,
>it may not be worthwhile to hold an AVT session in Vienna.
>
>Shall we not meet?
>
>                                                         -- Steve
>
>_______________________________________________
>Audio/Video Transport Working Group
>avt@ietf.org
>https://www1.ietf.org/mailman/listinfo/avt


-- 
David Singer
Apple Computer/QuickTime
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



From mailnull@www1.ietf.org  Fri Jun  6 13:13:34 2003
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Date: Fri, 06 Jun 2003 10:06:51 -0700
To: Dave Singer <singer@apple.com>
From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [AVT] Re: Approved
Cc: avt@ietf.org
In-Reply-To: <p05210644bb066d91b215@[17.202.35.52]>
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At 09:12 AM 6/6/2003 -0700, Dave Singer wrote:
>You'll note from the 'received' headers that I did not send this message:
>
>Received: from HYUN (v33-195.icu.ac.kr [210.107.133.195])
>         by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA12826      for 
> <avt@ietf.org>; Fri,
>  06 Jun 2003 03:24:18 -0400 (EDT)
>
>The from address was forged.  I regret the identity theft, but this seems 
>to be a recent virus that someone has, who has mail from me or has me in 
>their address book.

It's happening to many people right now.

Mark





>At 16:25 +0900 6/6/03, singer@apple.com wrote:
>>Please see the attached file.
>
>
>--
>David Singer
>Apple Computer/QuickTime
>_______________________________________________
>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  Sat Jun  7 16:25:33 2003
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To: casner@packetdesign.com
Cc: csp@csperkins.org, Conrad.Parker@csiro.au, avt@ietf.org
Message-ID: <3EE2487E.4060400@csiro.au>
Date: Sun, 08 Jun 2003 06:18:06 +1000
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I've just submitted a set of I-Ds that could be interesting to AVT.

We've developed an audio/video markup language similar to HTML and a 
streamable container format for it. Together with the temporal URIs 
which I have previously posted, that stuff enables Web-like searching 
and surfing of time-continuous data. Check it out at http://www.annodex.net/

We'll be coming to Vienna with two people to make friends and learn our 
way around. It would be great if we could get to know some AVT people!

Cheers,

Silvia.

Stephen Casner wrote:
> AVT participants:
> 
> I received only 5 responses to the message I sent on Friday asking who
> would plan to attend IETF in Vienna and who would like an AVT agenda
> slot:
> 
>   Alan Duric - will attend and present iLBC
>   Alan Clark - will attend but no new work for this meeting
>   John Lazzaro - won't be attending but has a lot of progress on MIDI
> 		 for us to discuss
>   Michael Ramalho - not sure about attending, does have RGL update but
> 		    could finish on the list beforehand
>   Stephan Wenger - will attend and present H.264 packetization
> 
> I thank these people for responding.  But given this small response,
> it may not be worthwhile to hold an AVT session in Vienna.
> 
> Shall we not meet?
> 
>                                                         -- Steve
> 
> _______________________________________________
> 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 Jun  9 07:52:43 2003
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Subject: [AVT] I-D ACTION:draft-ietf-avt-rtp-retransmission-08.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-08.txt
	Pages		: 27
	Date		: 2003-6-6
	
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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_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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Dear Leonardo,

H.263 was one of the reference that members of the ad hoc brought up.
There were significant support for comparing the AVC baseline against
the H.263 + a few annexes which was claimed to be significantly
different from MPEG-4 SP.

And since H.26L for which AVC was derived

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Subject: Re: [AVT] WG last call: RTP Profile for RTCP-based Feedback (RTP/AVPF)
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Hi Joerg,

Responses below:

Joerg Ott wrote:
> 
> I believe the actions are clear.  The paragraph after 2c even explicitly
> mentions 2c (and all others).  I do not see any need for additional
> wording.
> 

Okay.

> 
>> First I think one should use the word suppress, as this is what has 
>> been used on all other places. Then the text should also mention that 
>> tn and tp, Allow_early shall be updated as if the packet has been 
>> sent. It should also be made clear that t_rr_last SHALL NOT be updated.
> 
> 
> 
> I have added the use of "supressed".  I think stating that "t_rr_last MUST
> remain unchanged" is pretty unambiguous.  tn and tp as well as allow_early
> are dealt with collectively for all four alternatives which I believe is
> a feature, is clear, and which I do not want to change (it would just
> increase wording).

Okay.

> 
> 
>> My remaining objection is that the usage of T_rr_interval does not 
>> match the behaviour that setting T_min for the old algorithm achieved. 
>> Setting T_rr_interval to 5.0 seconds will not mean that a receiver 
>> will send messages every 5 seconds on average, given enough bandwidth 
>> and no feedback messages. This algorithm will instead send messages 
>> every N*Td  > T_rr_interval, where N is the smallest integer that 
>> result in that the equality is true, and Td is the deterministic RTCP 
>> reporting interval produced by the algorithm in the RTP spec.
> 
> 
> 
> It has never been stated anywhere -- and was explicitly not the intention
> -- to enforce T_rr_interval as an "exact" value.  It should just be a lower
> bound to limit the amount of RTCP traffic as you requested.  We have had
> many  discussions among the authors regarding this issue (about a year
> ago or so)  and came to the conclusion that trying to create another fixed
> timer will just complicate things (as this timer's clock needs to run
> independently of the regular RTCP intervals in order not to reduce the
> amount of early feedback that can be sent).
> 
>> Taking the randomization into account one will have the possibility to 
>> send a regular RTCP packet in the interval 0.5*T_rr_interval to 
>> 1.5*T_rr_interval+1.5*Td. Also I can't  determine what the 
>> distribution within this interval will become, at least it is not 
>> equal probability over the whole interval. If this has any ill effects 
>> compared to using a equal distribution between 0.5*T_rr_interval and 
>> 1.5*T_rr_interval I don't know.
> 
> 
> 
> If you choose to use the T_rr_interval, you are likely to do so because
> Td is too short for you, probably much too short (e.g. an order of
> magnitude).  This means that 1.5*T_rr_interval will clearly dominate the
> above equation.  Even if T_rr_interval == Td, you get 0.5*Td to 3*Td
> (wondering whether the latter is really possible).
> 
> As I have argued before: if you start introducing T_rr_interval then you
> do so to _reduce_ RTCP traffic.  This means in particular that you care
> less about frequent feedback which makes RTCP less useful for any kind of
> closed loop algorithm.  With RTCP and its dithering interval of
> 0.5 - 1.5 you won't get equidistant regular feedback anyway, and its
> frequency further varies with the number of session members and the RTCP
> packet size.
> 
> Why is this irregularity bad?  From experimenting, I think we have learned
> that the members do not synchronize their RTCP transmissions.  They all
> will also transmit them at some point and they will stay below the
> calculated RTCP bandwidth.  I don't think we want to ask for more.

Okay, as long as you have considered this I am fine. It is not how I 
would how made it, however it works this way, so lets end the discussion.

> 
>>>> 4. Page 19, Section 3.6.1, first paragraph. A question. The ACK mode 
>>>> is it only possible to use in cases where one is in a point to point 
>>>> communications scenario. However is it really necessary to require 
>>>> that only two participants in this scenario? I the case of SSRC 
>>>> multiplexing for RTP retransmission it will not be a different 
>>>> situation even if one uses 3 or 4 SSRC in the session.
>>>
>>>
>>> This is an interesting point.  Let me ask a quick counter-question: how
>>> different is this, at the RTP level, from a multicast-unicast translator
>>> scenario? If we want to make this distinction we can probably only use
>>> CNAME bindings to determine the actual number of entities in a session.
>>
>>
>> Yes, you are correct, a client can't determine this reliable. I don't 
>> have any problem with leaving the restriction in.
> 
> 
> 
> They key question probably is the impact on the retransmission draft.

Yes, when using SSRC multiplexing it is not possible to use the ACK 
mode. I don't know if anyone has a problem with this. I don't think I 
have any problem with this restriction. The RTP retransmission format 
does recommend that one uses NACK.

> 
> 
> 
> How about this:
> 
> There MUST be exactly one Application Layer FB message contained in the 
> FCI field, unless the Application Layer FB message structure itself 
> allows for
> stacking (e.g. by means of a fixed size or explicit length indicator).
> 

sounds good.


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@ericsson.com

_______________________________________________
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Hi Magnus,

thanks for your reply -- looks like we can close it then.  And, again,
thanks a lot for your review in general.

>> They key question probably is the impact on the retransmission draft.
> 
> Yes, when using SSRC multiplexing it is not possible to use the ACK 
> mode. I don't know if anyone has a problem with this. I don't think I 
> have any problem with this restriction. The RTP retransmission format 
> does recommend that one uses NACK.


Ok, I'll prepare an updated draft (-07) tomorrow and submit it --
unless someone still has issues here.  Please speak up now.

Joerg

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Date: Wed, 11 Jun 2003 10:34:23 -0400
From: Colin Perkins <csp@csperkins.org>
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Subject: [AVT] Fw: Internet-Draft Cutoff Dates for Vienna, Austria (July 16-21,
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Note, in particular, that new working group drafts need to be approved by
16th June.  If you are planning on submitting a draft-ietf-avt-...-00.txt
draft, then please contact the chairs before this date.

Colin


------- Forwarded message:
>Date: Fri, 06 Jun 2003 09:05:04 -0400
>From: Internet-Drafts Administrator <internet-drafts@ietf.org>
>To: IETF-Announce: ;
>Subject: Internet-Draft Cutoff Dates for Vienna, Austria (July 16-21,
>2003) 
>
> 
> NOTE: There are two (2) Internet-Draft Cutoff dates
> 
> June 23rd: Cutoff for Initial Submissions (new documents)
> 
> All initial submissions(-00) must be submitted by Monday, June 23rd, 
> at 09:00 ET.  Initial submissions received after this time will NOT be
> made available in the Internet-Drafts directory, and will have to be
> resubmitted.
> 
>  
> As before, all initial submissions (-00.txt) with a filename beginning
> with a draft-ietf MUST be approved by the appropriate WG Chair prior to
> processing and announcing. WG Chair approval must be received by
> Monday, June 16th.
> 
>  Please do NOT wait until the last minute to submit.
> 
> Be advised: NO placeholders. Updates to initial submissions received
>             the week of June 23rd will NOT be accepted.
> 
> June 30st: FINAL Internet-Draft Cutoff
> 
> All revised Internet-Draft submissions must be submitted by Monday,
> June 30st, 2003 at 09:00 ET.  Internet-Drafts received after this
> time will NOT be announced NOR made available in the Internet-Drafts
> Directories.
> 
> We will begin accepting Internet-Draft submissions the week of the
> meeting, though announcements will NOT be sent until the IETF meeting
> is over.
> 
> Thank you for your understanding and cooperation. Please do not hesitate
> to contact us if you have any questions or concenrs.
> 
> FYI: These and other significant dates can be found at
>       http://www.ietf.org/meetings/cutoff_dates_57.html
> 
> 

------- End of Forwarded Message

-- 
Colin Perkins
csp@csperkins.org
_______________________________________________
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From: Phil Kerr <philkerr@elec.gla.ac.uk>
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Subject: [AVT] Updated draft-kerr-avt-vorbis-rtp I-D
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Hi all,

An updated Vorbis-RTP I-D was submitted to the IETF last night:

http://www.ietf.org/internet-drafts/draft-kerr-avt-vorbis-rtp-02.txt

Changes cover:

Codebook caching mechanism
Expanded SDP parameters
Expanded MIME section
Expanded introduction
Packet loss section
Minor tweaks and clarity changes to text

Comments and feedback is welcomed.  I'd also like to request WG status
for the draft.

Open issues still exist with congestion control, as there is ongoing
development in this area, and best path MTU size.


Regards

Phil


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Thread-Topic: TimeStamp in the RFC2833 packet.
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From: "Oren Peleg" <OrenP@audiocodes.com>
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Subject: [AVT] TimeStamp in the RFC2833 packet.
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How does the RFC2833 TimeStamp values come along with the RTCP report =
Jitter Calculation explained at the RFC1889?

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">How does the RFC2833 =
TimeStamp values come along with the RTCP report Jitter Calculation =
explained at the RFC1889?</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT></P>

<P DIR=3DLTR><FONT COLOR=3D"#000080" SIZE=3D4 FACE=3D"Monotype =
Corsiva">Oren Peleg</FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

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Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
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From: Allison Mankin <mankin@psg.com>
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Subject: [AVT] IESG review of draft-ietf-avt-srtp.08 - first comments
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Mark and other SRTP authors,

The IESG is considering SRTP at the telechat tomorrow.  The review will
continue over till the Jun 26th meeting, so some comments will not come in
yet, but here is a Discuss comment from Steve Bellovin that it would be good
to respond to (don't revise the draft yet, though).

Allison

------- Forwarded Message

>
>	Please return the full line with your position.
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>Steve Bellovin      [   ]     [   ]       [ x ]      [   ] 


SSRC should be expanded in the text the first time it's used.

The IV definition in 4.1.1 has me a bit nervous.  Right now, it's
(k_s << 16) ^ (ssrc << 64) ^ (i << 16), where k_s is the session key, 
ssrc is the 32-bit synchronization source, and i is the 48-bit packet 
index.  The low-order 16 bits are for the block number within the 
packet, which is fine.

The problem I have is that given the mandated 0-padding, the high-order 
32 bits of the IV are from k_s, unmodified by anything else.  Furthermore
ssrc and i are known to the attacker, and the block count is obvious.  
This means that the IV is a trivial function of most of the session key.
I don't *think* that that's a problem, but any extra use of keys makes me 
nervous.



		--Steve Bellovin, http://www.research.att.com/~smb (me)
		http://www.wilyhacker.com (2nd edition of "Firewalls" book)



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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		: Extended RTP Profile for RTCP-based Feedback(RTP/AVPF)
	Author(s)	: J. Ott, S. Wenger et al.
	Filename	: draft-ietf-avt-rtcp-feedback-07.txt
	Pages		: 51
	Date		: 2003-6-11
	
Real-time media streams that use RTP are, to some degree, resilient 
against packet losses.  Receivers may use the base mechanisms of 
RTCP to report packet reception statistics and thus allow a sender 
to adapt its transmission behavior in the mid-term as sole means 
for feedback and feedback-based error repair (besides a few codec-
specific mechanisms).  This document defines an extension to the 
Audio-visual Profile (AVP) that enables receivers to provide, 
statistically, more immediate feedback to the senders and thus 
allow for short-term adaptation and efficient feedback-based repair 
mechanisms to be implemented.  This early feedback profile (AVPF) 
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Subject: Re: [AVT] TimeStamp in the RFC2833 packet.
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TimeStamp in the RFC2833 packet.The solution I am following right now is =
the one from http://www.cs.columbia.edu/~hgs/rtp/ which says under "How =
is Jitter Computed"
=20
"If several packets, say, within a video frame, bear the same timestamp, =
it is advisable to only use the first packet in a frame to compute the =
jitter. (This issue may be addressed in a future version of the =
specification.) ..."

I hope the future version of spec. mentioned here is coming soon, =
because this aspect of timestamp issue is quite confusing in the current =
one.

Regards

Ali Awais
  ----- Original Message -----=20
  From: Oren Peleg=20
  To: schulzrinne@cs.columbia.edu=20
  Cc: avt@ietf.org=20
  Sent: Wednesday, June 11, 2003 9:46 AM
  Subject: [AVT] TimeStamp in the RFC2833 packet.


  How does the RFC2833 TimeStamp values come along with the RTCP report =
Jitter Calculation explained at the RFC1889?

  Thanks,

  Oren Peleg

  Work phone No. : 03 - 5394167

  Home phone No. : 03 - 5335913

  Cell phone No. : 055 - 669613

  Email : Oren.Peleg@audiocodes.com


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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD><TITLE>TimeStamp in the RFC2833 packet.</TITLE>
<META content=3D"text/html; charset=3Dwindows-1255" =
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<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>
<DIV><FONT face=3DArial size=3D2>The solution I am following right now =
is the one=20
from <A=20
href=3D"http://www.cs.columbia.edu/~hgs/rtp/">http://www.cs.columbia.edu/=
~hgs/rtp/</A>&nbsp;which=20
says under "How is Jitter Computed"</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>"If several packets, say, within a =
video frame,=20
bear the same timestamp, it is advisable to only use the first packet in =
a frame=20
to compute the jitter. (This issue may be addressed in a future version =
of the=20
specification.) ..."</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>I hope the future version of spec. =
mentioned here=20
is coming soon, because this aspect of timestamp issue is quite =
confusing in the=20
current one.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Regards</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2>Ali =
Awais</FONT></DIV></FONT></DIV></FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"BORDER-LEFT: #000000 2px solid; MARGIN-LEFT: 5px; MARGIN-RIGHT: =
0px; PADDING-LEFT: 5px; PADDING-RIGHT: 0px">
  <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
  <DIV=20
  style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
  <A href=3D"mailto:OrenP@audiocodes.com" =
title=3DOrenP@audiocodes.com>Oren=20
  Peleg</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
  href=3D"mailto:schulzrinne@cs.columbia.edu"=20
  title=3Dschulzrinne@cs.columbia.edu>schulzrinne@cs.columbia.edu</A> =
</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
href=3D"mailto:avt@ietf.org"=20
  title=3Davt@ietf.org>avt@ietf.org</A> </DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, June 11, 2003 =
9:46=20
  AM</DIV>
  <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [AVT] TimeStamp in the =
RFC2833=20
  packet.</DIV>
  <DIV><BR></DIV><!-- Converted from text/rtf format -->
  <P dir=3Dltr><FONT face=3DArial size=3D2>How does the RFC2833 =
TimeStamp values come=20
  along with the RTCP report Jitter Calculation explained at the=20
  RFC1889?</FONT></P>
  <P dir=3Dltr><FONT face=3DArial size=3D2>Thanks,</FONT></P>
  <P dir=3Dltr><FONT color=3D#000080 face=3D"Monotype Corsiva" =
size=3D4>Oren=20
  Peleg</FONT></P>
  <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
  size=3D2>Work phone No. : 03 - 5394167</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
  size=3D2>Home phone No. : 03 - 5335913</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
  size=3D2>Cell phone No. : 055 - 669613</FONT></SPAN></P>
  <P dir=3Dltr><SPAN lang=3Den-us><FONT face=3D"Book Antiqua" =
size=3D2>Email :<U>=20
  </U></FONT><U><FONT color=3D#0000ff face=3D"Book Antiqua" size=3D2><A=20
  =
href=3D"mailto:Oren.Peleg@audiocodes.com">Oren.Peleg@audiocodes.com</A></=
FONT></U></SPAN><U><SPAN=20
  lang=3Dhe></SPAN></U><SPAN =
lang=3Dhe></SPAN></P></BLOCKQUOTE></BODY></HTML>

------=_NextPart_000_0222_01C330DD.31671250--

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



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CC: mbaugher@cisco.com, avt@ietf.org, Steve Bellovin
 <smb@research.att.com>
Subject: Re: [AVT] IESG review of draft-ietf-avt-srtp.08 - first comments
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Dear all,

Since I haven't seen any other reply, I will give a quick
response to Steve's concern (forwarded by Allison below).
See inline.


Allison Mankin wrote:
 > Mark and other SRTP authors,
 >
 > The IESG is considering SRTP at the telechat tomorrow.  The review will
 > continue over till the Jun 26th meeting, so some comments will not 
come in
 > yet, but here is a Discuss comment from Steve Bellovin that it would 
be good
 > to respond to (don't revise the draft yet, though).
 >
 > Allison
 >
 > ------- Forwarded Message
 >
 >
 >>	Please return the full line with your position.
 >>
 >>                   Yes    No-Objection  Discuss *  Abstain
 >>
 >>
 >>Steve Bellovin      [   ]     [   ]       [ x ]      [   ]
 >
 >
 >
 > SSRC should be expanded in the text the first time it's used.

OK.

 >
 > The IV definition in 4.1.1 has me a bit nervous.  Right now, it's
 > (k_s << 16) ^ (ssrc << 64) ^ (i << 16), where k_s is the session key,

Note: k_s is the "session salt", not the "session key".
I am not sure if this is a typo or a missinterpretation of the spec.
In the latter case (you've misstaken the salt for the key), I can
certainly see your concerns...

Let me clarify why it is (we are convinced) not only OK, but rather
REQUIRED to use the *salt*, k_s, in this way.

 > ssrc is the 32-bit synchronization source, and i is the 48-bit packet
 > index.  The low-order 16 bits are for the block number within the
 > packet, which is fine.
 >
 > The problem I have is that given the mandated 0-padding, the high-order
 > 32 bits of the IV are from k_s, unmodified by anything else.
 > Furthermore

Here, I think you have "reversed" the security issue that
motivates the presence of k_s in the IV: without the
k_s salt, the high order bits of the IV would be 00....0. Hence,
the k_s is there for the sole purpose of randomizng these zeros,
there is no need for the k_s to be randomized by something else
(more below).


 > ssrc and i are known to the attacker, and the block count is obvious.

Absolutely! *Without* the k_s salt, the IV would be highly predictable,
opening up for pre-computation attacks etc. By XORing k_s onto the
IV, we randomize the IV and fight off such attacks.

 > This means that the IV is a trivial function of most of the session key.
 > I don't *think* that that's a problem, but any extra use of keys 
makes me
 > nervous.

Recall how keys and salts are generated in SRTP (Sect 4.3 of the ID):

Starting from a master key and a master salt, we derive (among other
things):

* the session key, k_e (which is the AES counter mode key)
* the session salt, k_s (as above)

k_e and k_s are derived by applying a pseudo random function to
said master key and master salt. Hence, k_s and k_e are
"cryptographically isolated" (sorry for imprecise language,
but I think you understand what I mean), both from each other,
and from the master key. The session key, k_e, is kept secret
and is not part of any IV (but k_s is part of it as you noted).

Note: neither the master salt, nor the session salt (k_s) needs to be
secret, though as typically implemented, k_s will be secret. The
only property needed from the "salts" is that they are unpredictable.


Hope this sorts out things.

Best,

/mats


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



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To: =?ISO-8859-1?Q?Mats_N=E4slund?= <mats.naslund@era.ericsson.se>
Cc: mankin@psg.com, mbaugher@cisco.com, avt@ietf.org, housley@vigilsec.com
Subject: Re: [AVT] IESG review of draft-ietf-avt-srtp.08 - first comments 
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Date: Fri, 13 Jun 2003 22:59:31 -0400
From: "Steven M. Bellovin" <smb@research.att.com>
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In message <3EE9C121.80909@era.ericsson.se>, =?ISO-8859-1?Q?Mats_N=E4slund?= wr
ites:
>
>Dear all,
>
>Since I haven't seen any other reply, I will give a quick
>response to Steve's concern (forwarded by Allison below).
>See inline.
>
>
>Allison Mankin wrote:
> > Mark and other SRTP authors,
> >
> > The IESG is considering SRTP at the telechat tomorrow.  The review will
> > continue over till the Jun 26th meeting, so some comments will not 
>come in
> > yet, but here is a Discuss comment from Steve Bellovin that it would 
>be good
> > to respond to (don't revise the draft yet, though).
> >
> > Allison
> >
> > ------- Forwarded Message
> >
> >
> >>	Please return the full line with your position.
> >>
> >>                   Yes    No-Objection  Discuss *  Abstain
> >>
> >>
> >>Steve Bellovin      [   ]     [   ]       [ x ]      [   ]
> >
> >
> >
> > SSRC should be expanded in the text the first time it's used.
>
>OK.
>
> >
> > The IV definition in 4.1.1 has me a bit nervous.  Right now, it's
> > (k_s << 16) ^ (ssrc << 64) ^ (i << 16), where k_s is the session key,
>
>Note: k_s is the "session salt", not the "session key".
>I am not sure if this is a typo or a missinterpretation of the spec.
>In the latter case (you've misstaken the salt for the key), I can
>certainly see your concerns...

You're right -- I misread the spec.  I withdraw my DISCUSS; the 
SSRC change is a trivial RFC editor's note.

		--Steve Bellovin, http://www.research.att.com/~smb (me)
		http://www.wilyhacker.com (2nd edition of "Firewalls" book)


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



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Subject: RE: [AVT] TimeStamp in the RFC2833 packet.
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From: "Oren Peleg" <OrenP@audiocodes.com>
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Thanks. It seems like a good idea. Probably should be added to the =
RTP-new draft.
=20
Regards,
=20
Oren P.

-----Original Message-----
From: Awais Ali [mailto:aali@i2telecom.com]
Sent: Thursday, June 12, 2003 10:22 PM
To: Oren Peleg; schulzrinne@cs.columbia.edu
Cc: avt@ietf.org
Subject: Re: [AVT] TimeStamp in the RFC2833 packet.




The solution I am following right now is the one from =
http://www.cs.columbia.edu/~hgs/rtp/ which says under "How is Jitter =
Computed"
=20
"If several packets, say, within a video frame, bear the same timestamp, =
it is advisable to only use the first packet in a frame to compute the =
jitter. (This issue may be addressed in a future version of the =
specification.) ..."
=20
I hope the future version of spec. mentioned here is coming soon, =
because this aspect of timestamp issue is quite confusing in the current =
one.
=20
Regards
=20
Ali Awais

----- Original Message -----=20
From: Oren  <mailto:OrenP@audiocodes.com> Peleg=20
To: schulzrinne@cs.columbia.edu=20
Cc: avt@ietf.org=20
Sent: Wednesday, June 11, 2003 9:46 AM
Subject: [AVT] TimeStamp in the RFC2833 packet.


How does the RFC2833 TimeStamp values come along with the RTCP report =
Jitter Calculation explained at the RFC1889?

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167

Home phone No. : 03 - 5335913

Cell phone No. : 055 - 669613

Email : Oren.Peleg@audiocodes.com


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charset=3Dwindows-1255">
<TITLE>TimeStamp in the RFC2833 packet.</TITLE>

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<BODY bgColor=3D#ffffff>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D356244708-15062003>Thanks. It seems like a good idea. Probably =
should be=20
added to the RTP-new draft.</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D356244708-15062003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D356244708-15062003>Regards,</SPAN></FONT></DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN=20
class=3D356244708-15062003></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT color=3D#0000ff face=3DArial size=3D2><SPAN =
class=3D356244708-15062003>Oren=20
P.</SPAN></FONT></DIV>
<BLOCKQUOTE dir=3Dltr style=3D"MARGIN-RIGHT: 0px">
  <DIV align=3Dleft class=3DOutlookMessageHeader dir=3Dltr><FONT =
face=3DTahoma=20
  size=3D2>-----Original Message-----<BR><B>From:</B> Awais Ali=20
  [mailto:aali@i2telecom.com]<BR><B>Sent:</B> Thursday, June 12, 2003 =
10:22=20
  PM<BR><B>To:</B> Oren Peleg; schulzrinne@cs.columbia.edu<BR><B>Cc:</B> =

  avt@ietf.org<BR><B>Subject:</B> Re: [AVT] TimeStamp in the RFC2833=20
  packet.<BR><BR></DIV></FONT>
  <DIV><FONT face=3DArial size=3D2>
  <DIV><FONT face=3DArial size=3D2>
  <DIV><FONT face=3DArial size=3D2>The solution I am following right now =
is the one=20
  from <A=20
  =
href=3D"http://www.cs.columbia.edu/~hgs/rtp/">http://www.cs.columbia.edu/=
~hgs/rtp/</A>&nbsp;which=20
  says under "How is Jitter Computed"</FONT></DIV>
  <DIV><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>"If several packets, say, within a =
video frame,=20
  bear the same timestamp, it is advisable to only use the first packet =
in a=20
  frame to compute the jitter. (This issue may be addressed in a future =
version=20
  of the specification.) ..."</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>I hope the future version of spec. =
mentioned here=20
  is coming soon, because this aspect of timestamp issue is quite =
confusing in=20
  the current one.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Regards</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Ali =
Awais</FONT></DIV></FONT></DIV></FONT></DIV>
  <BLOCKQUOTE dir=3Dltr=20
  style=3D"BORDER-LEFT: #000000 2px solid; MARGIN-LEFT: 5px; =
MARGIN-RIGHT: 0px; PADDING-LEFT: 5px; PADDING-RIGHT: 0px">
    <DIV style=3D"FONT: 10pt arial">----- Original Message ----- </DIV>
    <DIV=20
    style=3D"BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: =
black"><B>From:</B>=20
    <A href=3D"mailto:OrenP@audiocodes.com" =
title=3DOrenP@audiocodes.com>Oren=20
    Peleg</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>To:</B> <A=20
    href=3D"mailto:schulzrinne@cs.columbia.edu"=20
    title=3Dschulzrinne@cs.columbia.edu>schulzrinne@cs.columbia.edu</A> =
</DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Cc:</B> <A =
href=3D"mailto:avt@ietf.org"=20
    title=3Davt@ietf.org>avt@ietf.org</A> </DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Sent:</B> Wednesday, June 11, =
2003 9:46=20
    AM</DIV>
    <DIV style=3D"FONT: 10pt arial"><B>Subject:</B> [AVT] TimeStamp in =
the RFC2833=20
    packet.</DIV>
    <DIV><BR></DIV><!-- Converted from text/rtf format -->
    <P dir=3Dltr><FONT face=3DArial size=3D2>How does the RFC2833 =
TimeStamp values=20
    come along with the RTCP report Jitter Calculation explained at the=20
    RFC1889?</FONT></P>
    <P dir=3Dltr><FONT face=3DArial size=3D2>Thanks,</FONT></P>
    <P dir=3Dltr><FONT color=3D#000080 face=3D"Monotype Corsiva" =
size=3D4>Oren=20
    Peleg</FONT></P>
    <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
    size=3D2>Work phone No. : 03 - 5394167</FONT></SPAN></P>
    <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
    size=3D2>Home phone No. : 03 - 5335913</FONT></SPAN></P>
    <P dir=3Dltr><SPAN lang=3Den-us><FONT color=3D#000000 face=3D"Book =
Antiqua"=20
    size=3D2>Cell phone No. : 055 - 669613</FONT></SPAN></P>
    <P dir=3Dltr><SPAN lang=3Den-us><FONT face=3D"Book Antiqua" =
size=3D2>Email :<U>=20
    </U></FONT><U><FONT color=3D#0000ff face=3D"Book Antiqua" =
size=3D2><A=20
    =
href=3D"mailto:Oren.Peleg@audiocodes.com">Oren.Peleg@audiocodes.com</A></=
FONT></U></SPAN><U><SPAN=20
    lang=3Dhe></SPAN></U><SPAN=20
lang=3Dhe></SPAN></P></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>

------_=_NextPart_001_01C3331C.335C4814--
_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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	(envelope-from csp@csperkins.org)
Date: Sun, 15 Jun 2003 09:56:37 -0400
From: Colin Perkins <csp@csperkins.org>
To: "Oren Peleg" <OrenP@audiocodes.com>
Cc: aali@i2telecom.com, avt@ietf.org, schulzrinne@cs.columbia.edu
Subject: Re: [AVT] TimeStamp in the RFC2833 packet.
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Section 6.4.4 of rtp-new discusses these issues.
Colin

--> "Oren Peleg" <OrenP@audiocodes.com>
writes:
> Thanks. It seems like a good idea. Probably should be added to the
> RTP-new draft.
>  
> Regards,
>  
> Oren P.
> 
> -----Original Message-----
> From: Awais Ali [mailto:aali@i2telecom.com]
> Sent: Thursday, June 12, 2003 10:22 PM
> To: Oren Peleg; schulzrinne@cs.columbia.edu
> Cc: avt@ietf.org
> Subject: Re: [AVT] TimeStamp in the RFC2833 packet.
> 
> 
> 
> 
> The solution I am following right now is the one from
> http://www.cs.columbia.edu/~hgs/rtp/ which says under "How is Jitter
> Computed"
>  
> "If several packets, say, within a video frame, bear the same timestamp,
> it is advisable to only use the first packet in a frame to compute the
> jitter. (This issue may be addressed in a future version of the
> specification.) ..."
>  
> I hope the future version of spec. mentioned here is coming soon, because
> this aspect of timestamp issue is quite confusing in the current one.
>  
> Regards
>  
> Ali Awais
> 
> ----- Original Message ----- 
> From: Oren  <mailto:OrenP@audiocodes.com> Peleg 
> To: schulzrinne@cs.columbia.edu 
> Cc: avt@ietf.org 
> Sent: Wednesday, June 11, 2003 9:46 AM
> Subject: [AVT] TimeStamp in the RFC2833 packet.
> 
> 
> How does the RFC2833 TimeStamp values come along with the RTCP report
> Jitter Calculation explained at the RFC1889?
> 
> Thanks,
> 
> Oren Peleg
> 
> Work phone No. : 03 - 5394167
> 
> Home phone No. : 03 - 5335913
> 
> Cell phone No. : 055 - 669613
> 
> Email : Oren.Peleg@audiocodes.com
> 
> 


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



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Subject: [AVT] REMINDER: DEADLINE * ACM Multimedia Computing & Networking 2004
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**********************************************************************
*                                                                    *
*                           CALL FOR PAPERS                          *
*                                                                    *
*                             MMCN 2004                              *
*                                                                    *
*              MULTIMEDIA COMPUTING AND NETWORKING 2004 (EI22)       *
*                                                                    *
*                     http://www.ics.uci.edu/~mmcn2004               *
*                                                                    *
*                          January 21-22, 2004                       *
*                          San Jose, CA, USA                         *
*                                                                    *
*                Held in cooperation with ACM SIG Multimedia         *
*                                                                    *
**********************************************************************


MMCN 2004 will be held January 21-22, 2004, at the San Jose Marriott
and San Jose Convention Center, San Jose, California, USA.

Present your work at MMCN 2004 and define the future of multimedia!

The paper submission deadline is June 23, 2003.
For paper submissions please check the MMCN 2004 
webpage at http://www.ics.uci.edu/~mmcn2004

Recent advances in telecommunication, storage and information
technologies along with enhancements in consumer electronics have
stimulated the development of pervasive multimedia applications.
New and upcoming standards (MPEG-4, MPEG-7 and MPEG-21) facilitate
the seamless integration of multiple modalities of information
into truly interoperable multimedia frameworks, creating new
experiences for consumers. Such powerful multimedia solutions can
have significant impact across a wide spectrum of commercial,
consumer, non-profit and governmental domains.

The objective of this conference is to bring together researchers,
developers, and practitioners contributing to all facets of
multimedia computing and networking. The conference will serve as
an invigorating forum for the dissemination of state-of-the-art
research, development, and implementations of multimedia systems,
technologies, and applications. Presenters will be encouraged to
make multimedia presentations and demonstrate their solutions in
person.

We especially encourage papers on emerging technologies such as
multimedia and QoS support for 3G and UWB networks, power-aware
computing and communications, mobile and fixed wireless multimedia
networks, multimedia in P2P environments, network processors,
content distribution networks, home networking and digital
appliances. An exclusive industrial track will feature industrial
design experiences and showcase tools for next-generation
multimedia systems and applications.

Papers are solicited in all areas of multimedia, including, but
not limited to:

Multimedia Computing

    * hardware accelerators
    * multimedia OS services
    * power-aware systems
    * video-on-demand services
    * peer-to-peer media systems


Multimedia and the Internet

    * web-based services
    * push technologies and content distribution
    * wide-area caching
    * data streaming and delivery

Multimedia Networking

    * mobile and wireless networks
    * broadband networks
    * network-processor systems
    * home networking
    * QoS control and scheduling
    * access technologies

Measurement and Modeling

    * performance measurement of multimedia systems
    * statistical modeling of server traffic and server software
    * multimedia system simulations
    * benchmark comparisons

Multimedia Middleware
    * proxy services 
    * transcoding 
    * multimedia data management 
    * security and multimedia 

Case Studies and Applications

    * synthetic animation	
    * multimedia search engines 
    * entertainment and games
    * distributed virtual reality
    * multimedia authoring


We specifically solicit submissions in new and emerging areas in
the field.

AWARDS will be given to the best paper and the best student paper.

PANELS will consist of discussions on timely and controversial
topics.


Author Information

Submissions should not exceed 15 single-spaced pages including
figures, tables, and references, using a typeface no smaller than
10 point. Papers must be electronically submitted in PDF or
postscript format (using standard fonts and "US letter" size).
Detailed instructions on the electronic submission process can be
found on the conference web page at
http://www.ics.uci.edu/~mmcn2004/. Please also submit
a cover page electronically, (in plain text format) to
mmcn2004@ics.uci.edu. Each cover page should contain paper title,
author names and affiliations, name and address (both postal and
electronic) of contact author, abstract (less than 500 words),
keywords, and submission area.

Please direct questions about paper submission to the Program
Co-Chair at mmcn2004@ics.uci.edu


IMPORTANT DATES

Paper submission deadline    :  June 23, 2003
Notification of acceptance   :  Aug 24, 2003
Camera-ready manuscripts due :  Oct 27, 2003
200-word Summary to SPIE     :  Nov 17, 2003


CONFERENCE ORGANIZERS

General Chair
    Ragunathan Rajkumar, Carnegie Mellon University

Program Chair
    Nalini Venkatasubramanian, University of California, Irvine.

Publicity Chairs
    Americas: Ketan Mayer-Patel, University of North Carolina, 
                                 Chapel Hill
    Europe:   Carsten Griwodz, University of Oslo, Norway
    Far East: Wei-Ying Ma, Microsoft Research, China

Program Committee

    Tarek F. Abdelzaher, University of Virginia
    Kevin Almeroth, University of California, Santa Barbara
    Riccardo Bettati, Texas A&M University
    Nina Bhatti, Hewlett-Packard Labs
    Sajal K. Das, University of Texas, Arlington
    Magda El Zarki, University of California, Irvine
    Carsten Griwodz, University of Oslo (Norway)
    Martin G. Kienzle, IBM Research
    Ketan D. Mayer-Patel, University of North Carolina, Chapel Hill
    Klara Nahrstedt, University of Illinois, Urbana-Champaign
    Scott Brandt, University of California, Santa Cruz
    Douglas C. Schmidt, DARPA/Vanderbilt University
    Prashant J. Shenoy, University of Massachusetts, Amherst
    Cormac J. Sreenan, University College Cork (Ireland)
    Michael Zink, Technical University of Darmstadt (Germany)
    Roger Zimmermann, University of Southern California
    Dongyan Xu, Purdue University
    Hong Jiang Zhang, Microsoft Research
    Wei-Ying Ma, Microsoft Research, China
    Reza Rejaie, University of Oregon
    
--------------------------------------------------------------------------------

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



From mailnull@www1.ietf.org  Sun Jun 15 16:38:58 2003
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Subject: RE: [AVT] TimeStamp in the RFC2833 packet.
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From: "Oren Peleg" <OrenP@audiocodes.com>
To: "Colin Perkins" <csp@csperkins.org>
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I read this section before, & I think that it should still be added. The RTCP Jitter value is used for Call Controls (e.g. MGCP) to know the quality of the call when finished. If for example, an RFC2833 ABCD Signaling packet will be transmitted every 5 seconds with the TimeStamp value of the beginning of the call, the Jitter calculation will be damaged significantly.

Regards,

Oren P. 

-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Sunday, June 15, 2003 4:57 PM
To: Oren Peleg
Cc: aali@i2telecom.com; avt@ietf.org; schulzrinne@cs.columbia.edu
Subject: Re: [AVT] TimeStamp in the RFC2833 packet.


Section 6.4.4 of rtp-new discusses these issues.
Colin

--> "Oren Peleg" <OrenP@audiocodes.com>
writes:
> Thanks. It seems like a good idea. Probably should be added to the
> RTP-new draft.
>  
> Regards,
>  
> Oren P.
> 
> -----Original Message-----
> From: Awais Ali [mailto:aali@i2telecom.com]
> Sent: Thursday, June 12, 2003 10:22 PM
> To: Oren Peleg; schulzrinne@cs.columbia.edu
> Cc: avt@ietf.org
> Subject: Re: [AVT] TimeStamp in the RFC2833 packet.
> 
> 
> 
> 
> The solution I am following right now is the one from
> http://www.cs.columbia.edu/~hgs/rtp/ which says under "How is Jitter
> Computed"
>  
> "If several packets, say, within a video frame, bear the same timestamp,
> it is advisable to only use the first packet in a frame to compute the
> jitter. (This issue may be addressed in a future version of the
> specification.) ..."
>  
> I hope the future version of spec. mentioned here is coming soon, because
> this aspect of timestamp issue is quite confusing in the current one.
>  
> Regards
>  
> Ali Awais
> 
> ----- Original Message ----- 
> From: Oren  <mailto:OrenP@audiocodes.com> Peleg 
> To: schulzrinne@cs.columbia.edu 
> Cc: avt@ietf.org 
> Sent: Wednesday, June 11, 2003 9:46 AM
> Subject: [AVT] TimeStamp in the RFC2833 packet.
> 
> 
> How does the RFC2833 TimeStamp values come along with the RTCP report
> Jitter Calculation explained at the RFC1889?
> 
> Thanks,
> 
> Oren Peleg
> 
> Work phone No. : 03 - 5394167
> 
> Home phone No. : 03 - 5335913
> 
> Cell phone No. : 055 - 669613
> 
> Email : Oren.Peleg@audiocodes.com
> 
> 


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



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From: Stephen Casner <casner@acm.org>
To: AVT WG <avt@ietf.org>
cc: Colin Perkins <csp@csperkins.org>,
        Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Allison Mankin <mankin@psg.com>
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Subject: [AVT] AVT will meet in Vienna; new co-chair
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Two important announcements:

1.  AVT will meet in Vienna.  Given the smaller-than-usual number of
    topics to discuss, we plan to meet for just one session.  We don't
    know yet what day and time that session will be.  Anyone who would
    like a time slot and has not yet made a request should do so
    promptly.

2.  I will step down as co-chair of AVT after the Vienna meeting.
    Colin Perkins will continue and Magnus Westerlund has accepted our
    invitation to join now as a new co-chair.  The Transport Area
    Directors have approved adding Magnus.

I began as chair of AVT when it was formed at the Santa Fe IETF in
November, 1991 (the first regular meeting was March, 1992).  Colin
joined as co-chair in August, 1998.  The reason for my transition at
this time is that my real job is no longer related to AVT work, nor is
it academia/research.  I have not been able to put as much energy into
AVT work as it deserves, and that slows down progress.

Meanwhile, Magnus has demonstrated his commitment to AVT work by
carefully reviewing and commenting on many of the drafts, as well as
authoring some of this own.  In my opinion, Magnus has shown a good
sense of direction in AVT-related protocol design and a strong
motivation to get IETF work done and done right.  I am confident that
Colin and Magnus can provide the leadership needed for AVT to continue
its work.

See you in Wien!

                                                        -- Steve

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



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	(envelope-from casner@acm.org)
Date: Sun, 15 Jun 2003 17:35:13 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Jose Rey <rey@panasonic.de>
cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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On Wed, 4 Jun 2003, Magnus Westerlund wrote:

> I have been think over this problem some more. I have one problem with
> the proposed algorithm. The proposed algorithm seems to be the correct
> one when one comes to conversational, confernce or multicast
> applications where there is a potential large number of receivers and
> senders needs to be prioritized to provide synchronization data.

So far I agree.

> However I have been thinking on the case of on-demand streaming. In this
> case one has normally a single sender and one receiver. In this case it
> is no question that the sender will get the complete RS pool. However in
> this case on is actually interested in giving the receiver the complete
> RR pool. The sender will suffice with the given RS pool as this can
> easily be calculated and the SR information is less important for
> synchronization. The receiver instead has a lot of interesting to report
> and creates large packets that it want to send often. Therefore a larger
> RR bandwidth is motivated.

I don't see why the synchronization information is any less important
in this case than for the earlier ones.

> In the above case on would like to send receiver reports utilizing the
> full RR pot for the receiver. One might also want to assign more
> bandwidth to RR then to RS. What happens in this case, as can be seen
> below, is that the receiver actually gets less.
>
> For RS > RR:
> Receiver RTCP bandwidth = RR/1
> Senders bandwidth = RS/1
>
> For RS < RR
> Receivers RTCP bandwidth = RR/2
> Senders bandwidth = (RS + RR)/2

While this arithmetic is correct, I'm not sure it matters.  There are
two considerations you have not included in your analysis:

1.  The reporting interval is determined by dividing the average
    packet size by the calculated bandwidth.  Since the average packet
    size is calculated using all the RTCP packets from both senders
    and receivers, then receivers will be getting a larger share of
    the bandwidth because their RTCP packets are larger.

2.  For narrow channels, the 5 second minimum will constrain the
    interval calculation more than the default 5% (or 3.75%) RTCP bw,
    so unless the application needs to severely constrain the bw to
    make the interval even longer, it won't make a difference.

> This get even worse if one would use SSRC multiplex RTP retransmission.
> In this case one gets two senders and one receivers as long as
> retransmission occurs more than every two RTCP reporting intervals.
>
> For RS/2 > RR:
> Receiver RTCP bandwidth = RR/1
> Senders bandwidth = RS/2
>
> For RS/2 < RR:
> Receiver RTCP bandwidth = RR/3
> Senders bandwidth = (RS+RR)/3

If RTP retransmission is used, I would assume that AVPF is also being
used, and therefore the interval calculation is modified.  Given the
variability in both data bw and RTCP bw depending upon the error rate,
there has got to be significant headroom between the base data rate
and the channel capacity, so precisely controlling the RTCP bw should
not be an issue.


On Fri, 6 Jun 2003, Jose Rey wrote:

> in my understanding the problem is the following: RTCP BW modifiers
> don't work for RTP unicast with the algorithm in RTP draft annex
> A.7. And since unicast is a very important application which will be
> using these modifiers (at least 3GPP will be using them), it is
> necessary to do something about this.

Unicast is not the right characterization.  For conversational
unicast, which has symmetric requirements, there should be no problem.
Unicast streaming, where the data transmission is unidirectional, is
where the questions arise.  Furthermore, it is the inclusion of only
two participants, not the use of unicast transmission, that might make
this case different.  Note that there can be RTP sessions with more
than two participants using unicast through an RTP mixer.

> An added side problem is that you cannot turn the receivers' RTCP
> off, because (as explained below) all session participants are
> treated equal.
> Now in detail, applying the algorithm in RTP-draft section A.7 for
> unicast, the end result is that senders AND receivers equally share
> the bandwidth, each getting a value equal to (RR+RS)/members,
> because the if-clause is always false.

Assuming a modification to use RS/(RS+RR) as the
RTCP_SENDER_BW_FRACTION rather than a fixed .25, then the algorithm in
A.7 gives non-data-senders zero RTCP bw when RR is zero because the
number of senders is less than the number of members.  Part of my
proposal was to remove the "senders > 0" term so that there isn't a
problem when there are no senders.

> This gets even worse if RTP retransmission is used (where different
> SSRCs might be used to multiplex media and retransmissions) as the
> bandwidth that one wants to dedicate to the receiver is actually
> shared equally between an increased number of SSRCs.

Yes, although not really equally because of the difference in packet
sizes.  But given the considerations I have outlined above, is it a
problem that we need to address?

> If changing the algorithm is too much of a change at this stage AND
> this shall not be solved by draft-ietf-avt-rtcp-bw-05.txt, then the
> only solution I see is to tell RTCP through SDP who uses which
> bandwidth values, but this needs also of another rtcp_interval()
> algorithm that accepts these values as parameters (as Magnus
> proposed in his code). Makes this sense to all?
>
> Would another draft be the rigth place for this other algorithm?
> Would that constitute a new profile?

A new profile could change the algorithm if that is necessary.  New
SDP parameters could be defined if that was necessary.  I suggest that
you guys run some numbers to see if it is a problem.

                                                        -- Steve

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



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	(envelope-from casner@acm.org)
Date: Sun, 15 Jun 2003 23:46:56 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Jose Rey <rey@panasonic.de>
cc: IETF AVT WG <avt@ietf.org>
Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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Some more thoughts on this:

To simplify the unicast streaming application, we could specify that
when RTCP bw is specified with RS and RR those values are used
separately for senders and non-senders without any consideration of
treating senders and receivers together when the number of senders is
greater than RS/(RS+RR).

However, that would not be good for conversational applications
(unicast or multicast) because the classification as sender changes
over time.  At times when both ends have sent within two intervals,
they would be using the RS bw only and ignoring the RR bw.  To get
around this we could set RS to be the total, and RR zero, but then if
one was sending and the other just listening for a long time, the
listener would get nothing.  For these applications, you need the
smooth transision that the existing algorithm was intended to
provide.

                                                        -- Steve

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



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        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Subject: RE: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
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Hi,

> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of Stephen
> Casner
> Sent: Monday, June 16, 2003 8:47 AM
> To: Magnus Westerlund; Jose Rey
> Cc: IETF AVT WG
> Subject: Re: [AVT] Re: Errors in draft-ietf-avt-rtcp-bw-05.txt
>
>
> Some more thoughts on this:
>
> To simplify the unicast streaming application, we could specify that
> when RTCP bw is specified with RS and RR those values are used
> separately for senders and non-senders without any consideration of
> treating senders and receivers together when the number of senders is
> greater than RS/(RS+RR).

I think this is the best and easiest solution. In 3GPP unicast streaming it is
the server that sets the RS and RR values, that means the server should consider
well how much bandwidth it is needed for own reporting and for the sender's.
There's no case where the algorithm must adapt, the scenario is rather static.

>
> However, that would not be good for conversational applications
> (unicast or multicast) because the classification as sender changes
> over time.  At times when both ends have sent within two intervals,
> they would be using the RS bw only and ignoring the RR bw.  To get
> around this we could set RS to be the total, and RR zero, but then if
> one was sending and the other just listening for a long time, the
> listener would get nothing.  For these applications, you need the
> smooth transision that the existing algorithm was intended to
> provide.
>
I see. I was referring more to the streaming unicast scenario. I guess my
characterization was not accure enough.

cheers,

Jose

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

_______________________________________________
Audio/Video Transport Working Group
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Date: Mon, 16 Jun 2003 13:27:57 -0700
From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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Subject: Re:  [AVT] AVT will meet in Vienna; new co-chair 
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> Stephen Casner <casner@acm.org> writes:
>
> 2.  I will step down as co-chair of AVT after the Vienna meeting.
>     Colin Perkins will continue and Magnus Westerlund has accepted our
>     invitation to join now as a new co-chair.  The Transport Area
>     Directors have approved adding Magnus.

We've all been incredibly lucky to have Steve at the co-helm of the 
ship ... and we're also incredibly lucky to have Magnus step up to
share the wheel with Colin going forward. 

-------------------------------------------------------------------------
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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From: John Lazzaro <lazzaro@CS.Berkeley.EDU>
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> You don't sail, do you?

No, but I went to see Finding Nemo a few days ago, and my
brain is still in maritime mode :-).

-------------------------------------------------------------------------
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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On Mon, 16 Jun 2003, John Lazzaro wrote:

> > Stephen Casner <casner@acm.org> writes:
> >
> > 2.  I will step down as co-chair of AVT after the Vienna meeting.
> >     Colin Perkins will continue and Magnus Westerlund has accepted our
> >     invitation to join now as a new co-chair.  The Transport Area
> >     Directors have approved adding Magnus.
>
> We've all been incredibly lucky to have Steve at the co-helm of the
> ship ... and we're also incredibly lucky to have Magnus step up to
> share the wheel with Colin going forward.

You don't sail, do you?

L.

<http://www.ee.surrey.ac.uk/Personal/L.Wood/><L.Wood@ee.surrey.ac.uk>
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Thread-Topic: Q about draft-ietf-avt-rtp-new-12
Thread-Index: AcM0t6PBx11hoJvyEdeSdQDQt7+w2w==
From: "Oren Peleg" <OrenP@audiocodes.com>
To: <schulzrinne@cs.columbia.edu>, "Stephen Casner" <casner@acm.org>
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Subject: [AVT] Q about draft-ietf-avt-rtp-new-12
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This is a multi-part message in MIME format.

------_=_NextPart_001_01C334B7.A278F944
Content-Type: text/plain;
	charset="windows-1255"
Content-Transfer-Encoding: quoted-printable

In the draft-ietf-avt-rtp-new-12 it is mentioned that the RTCP packet =
MUST end on a 32-bit boundary. At the RTCP Padding section, it is =
mentioned that the padding is done octet based - length value is at  the =
last padding octet:=20

"If the padding bit is set, this individual RTCP packet
             contains some additional padding octets at the end which
             are not part of the control information but are included in
             the length field.  The last octet of the padding is a count
             of how many padding octets should be ignored, including
             itself (it will be a multiple of four)."

As I understand it, the Padding MUST be 32-bit aligned. Is my assumption =
correct? if it is, shouldn't it be mentioned at the Padding part? If =
this will be misunderstood, it could cause reading of a garbage octet as =
the padding length ...

Regards,=20

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<TITLE>Q about draft-ietf-avt-rtp-new-12</TITLE>
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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">In the =
draft-ietf-avt-rtp-new-12 it is mentioned that the RTCP packet MUST end =
on a 32-bit boundary. At the RTCP Padding<SPAN LANG=3D"en-us"> section, =
it is mentioned that the padding is done octet based - length value is =
at&nbsp; the last padding octet: </SPAN></FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">&quot;If =
the padding bit is set, this individual RTCP packet</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; contains some additional padding octets at the end =
which</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; are not part of the control information but are included =
in</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; the length field.&nbsp; The last octet of the padding is =
a count</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; of how many padding octets should be ignored, =
including</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; itself (it will be a multiple of =
four).&quot;</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Arial">As I =
understand it, the Padding MUST be 32-bit aligned. Is my assumption =
correct? if it is, shouldn't it be mentioned at the Padding part? If =
this will be misunderstood, it could cause reading of a garbage octet as =
the padding length ...</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 =
FACE=3D"Arial">Regards,</FONT></SPAN><SPAN LANG=3D"he"> </SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"he"><FONT COLOR=3D"#000080" SIZE=3D4 =
FACE=3D"Monotype Corsiva">Oren Peleg</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Book =
Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Book =
Antiqua">Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN =
LANG=3D"he"></SPAN></U><SPAN LANG=3D"he"></SPAN></P>

</BODY>
</HTML>
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_______________________________________________
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On Tue, 17 Jun 2003, Oren Peleg wrote:

> In the draft-ietf-avt-rtp-new-12 it is mentioned that the RTCP
> packet MUST end on a 32-bit boundary. At the RTCP Padding section,
> it is mentioned that the padding is done octet based - length value
> is at the last padding octet:
>
> "If the padding bit is set, this individual RTCP packet
>              contains some additional padding octets at the end which
>              are not part of the control information but are included in
>              the length field.  The last octet of the padding is a count
>              of how many padding octets should be ignored, including
>              itself (it will be a multiple of four)."
>
> As I understand it, the Padding MUST be 32-bit aligned. Is my
> assumption correct? if it is, shouldn't it be mentioned at the
> Padding part? If this will be misunderstood, it could cause reading
> of a garbage octet as the padding length ...

This text seems clear to me.  The padding must fit within the length
field, which is in units of 32-bit words; it is appended to the
contents of an RTCP packet, which must end on a 32-bit boundary for
all packet types; and the padding count is a multiple of four.

What construction did you envision that complies with those
requirements and is not 32-bit aligned?

                                                        -- Steve


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



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From: "Oren Peleg" <OrenP@audiocodes.com>
To: "Stephen Casner" <casner@acm.org>
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It is mentioned that it is "included in the length field". This could be interpreted as if padding exists only for 1 octet - still the Length will be increased by one - rounded to 32bit size. It could be that Im just blowing some wind, but it did cause my suspicious level to rise.

Thanks for clearing it up to me,

Oren P.

-----Original Message-----
From: Stephen Casner [mailto:casner@acm.org]
Sent: Tuesday, June 17, 2003 7:46 PM
To: Oren Peleg
Cc: schulzrinne@cs.columbia.edu; avt@ietf.org
Subject: Re: Q about draft-ietf-avt-rtp-new-12


On Tue, 17 Jun 2003, Oren Peleg wrote:

> In the draft-ietf-avt-rtp-new-12 it is mentioned that the RTCP
> packet MUST end on a 32-bit boundary. At the RTCP Padding section,
> it is mentioned that the padding is done octet based - length value
> is at the last padding octet:
>
> "If the padding bit is set, this individual RTCP packet
>              contains some additional padding octets at the end which
>              are not part of the control information but are included in
>              the length field.  The last octet of the padding is a count
>              of how many padding octets should be ignored, including
>              itself (it will be a multiple of four)."
>
> As I understand it, the Padding MUST be 32-bit aligned. Is my
> assumption correct? if it is, shouldn't it be mentioned at the
> Padding part? If this will be misunderstood, it could cause reading
> of a garbage octet as the padding length ...

This text seems clear to me.  The padding must fit within the length
field, which is in units of 32-bit words; it is appended to the
contents of an RTCP packet, which must end on a 32-bit boundary for
all packet types; and the padding count is a multiple of four.

What construction did you envision that complies with those
requirements and is not 32-bit aligned?

                                                        -- Steve


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



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Date: Tue, 17 Jun 2003 14:31:01 -0700 (PDT)
From: MMCN-2004 <mmcn2004@ics.uci.edu>
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Subject: [AVT] REMINDER: paper submission deadline EXTENDED FOR MMCN-2004 (new
 deadline July 5, 2003) *  ACM Multimedia Computing & Networking 2004 *
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Hi all, 

Please note that the final paper submission deadline for ACM Multimedia 
Computing & Networking (MMCN 2004) has now been extended to July 5,
2003. 


**********************************************************************
*                                                                    *
*                           CALL FOR PAPERS                          *
*                                                                    *
*                             MMCN 2004                              *
*                                                                    *
*              MULTIMEDIA COMPUTING AND NETWORKING 2004 (EI22)       *
*                                                                    *
*                     http://www.ics.uci.edu/~mmcn2004               *
*                                                                    *
*                          January 21-22, 2004                       *
*                          San Jose, CA, USA                         *
*                                                                    *
*                Held in cooperation with ACM SIG Multimedia         *
*                                                                    *
**********************************************************************


MMCN 2004 will be held January 21-22, 2004, at the San Jose Marriott
and San Jose Convention Center, San Jose, California, USA.

Present your work at MMCN 2004 and define the future of multimedia!

The paper submission deadline is July 5, 2003, 11:59 PM Pacific Standard
Time (PST).For paper submissions please check the MMCN webpage at 
http://www.ics.uci.edu/~mmcn2004

Recent advances in telecommunication, storage and information
technologies along with enhancements in consumer electronics have
stimulated the development of pervasive multimedia applications.
New and upcoming standards (MPEG-4, MPEG-7 and MPEG-21) facilitate
the seamless integration of multiple modalities of information
into truly interoperable multimedia frameworks, creating new
experiences for consumers. Such powerful multimedia solutions can
have significant impact across a wide spectrum of commercial,
consumer, non-profit and governmental domains.

The objective of this conference is to bring together researchers,
developers, and practitioners contributing to all facets of
multimedia computing and networking. The conference will serve as
an invigorating forum for the dissemination of state-of-the-art
research, development, and implementations of multimedia systems,
technologies, and applications. Presenters will be encouraged to
make multimedia presentations and demonstrate their solutions in
person.

We especially encourage papers on emerging technologies such as
multimedia and QoS support for 3G and UWB networks, power-aware
computing and communications, mobile and fixed wireless multimedia
networks, multimedia in P2P environments, network processors,
content distribution networks, home networking and digital
appliances. An exclusive industrial track will feature industrial
design experiences and showcase tools for next-generation
multimedia systems and applications.

Papers are solicited in all areas of multimedia, including, but
not limited to:

Multimedia Computing

    * hardware accelerators
    * multimedia OS services
    * power-aware systems
    * video-on-demand services
    * peer-to-peer media systems


Multimedia and the Internet

    * web-based services
    * push technologies and content distribution
    * wide-area caching
    * data streaming and delivery

Multimedia Networking

    * mobile networks
    * wireless networks
    * broadband networks
    * network-processor systems
    * home networking
    * QoS control and scheduling
    * access technologies


Measurement and Modeling

    * performance measurement of multimedia systems
    * statistical modeling of server traffic and server software
    * multimedia system simulations
    * benchmark comparisons

Applications

    * multimedia search engines
    * entertainment and games
    * reflective applications

Multimedia Middleware
    * proxy services 
    * transcoding 
    * multimedia data management 
    * security and multimedia 

Case Studies

    * synthetic animation
    * distributed virtual reality
    * multimedia DBs and authoring


We specifically solicit submissions in new and emerging areas in
the field.

AWARDS will be given to the best paper and the best student paper.

PANELS will consist of discussions on timely and controversial
topics.


Author Information

Submissions should not exceed 15 single-spaced pages including
figures, tables, and references, using a typeface no smaller than
10 point. Papers must be electronically submitted in PDF or
postscript format (using standard fonts and "US letter" size).
Detailed instructions on the electronic submission process can be
found on the conference web page at
http://www.ics.uci.edu/~mmcn2004/. Please also submit
electronically (in plain text format) a cover page to
mmcn2004@ics.uci.edu. Each cover page should contain paper title,
author names and affiliations, name and address (both postal and
electronic) of contact author, abstract (less than 500 words),
keywords, and submission area.

Please direct questions about paper submission to the Program
Co-Chair at mmcn2004@ics.uci.edu


IMPORTANT DATES

Paper submission deadline    :  July 5, 2003
Notification of acceptance   :  Aug 24, 2003
Camera-ready manuscripts due :  Oct 27, 2003
200-word Summary to SPIE     :  Nov 17, 2003


CONFERENCE ORGANIZERS

General Chair
    Raj Rajkumar, Carnegie Mellon University

Program Chair
    Nalini Venkatasubramanian, University of California, Irvine.

Publicity Chairs
    Americas: Ketan Mayer-Patel, University of North Carolina, 
                                 Chapel Hill
    Europe:   Carsten Griwodz, University of Oslo, Norway
    Far East: Wei-Ying Ma, Microsoft Research, China

Program Committee

    Tarek Abdelzaher, University of Virginia
    Kevin Almeroth, University of California, Santa Barbara
    Riccardo Bettati, Texas A&M University
    Nina Bhatti, HP Labs
    Sajal Das, University of Texas, Arlington
    Magda El Zarki, University of California, Irvine
    Carsten Griwodz, University of Oslo
    Martin Kienzle, IBM Research
    Ketan Mayer-Patel, University of North Carolina, Chapel Hill
    Klara Nahrstedt, University of Illinois, Urbana-Champaign
    Douglas C. Schmidt, DARPA/Vanderbilt University
    Prashant Shenoy, University of Massachusetts, Amherst
    Cormac Sreenan, University College Cork
    Michael Zink, Technical University of Darmstadt
    Dongyan Xu, Purdue University
    Roger Zimmerman, University of Southern California
    Scott Brandt, University of California, Santa Cruz
    Hong Jiang Zhang, Microsoft Research
    Reza Rejaie, University of Oregon
    Wei-Ying Ma, Microsoft Research, China


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



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



From exim@www1.ietf.org  Wed Jun 18 15:53:07 2003
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Subject: [AVT] Re: Questions about draft-ietf-avt-rfc2833bis-02 and sub-events
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Rhodes, Ann wrote:
> These questions are based on the draft-ietf-avt-rfc2833bis-02
> document, specifically in the area of sub-events for DTMF digits
> lasting longer than 8.192 seconds.
> 
> 1. First, this draft of the specification indicates it "expires" in
> April, 2003. Does that mean that this draft is now the accepted
> standard since the comment period has expired?

I plan to issue a new draft shortly, but have been waylaid with other items.

> 
> 2. Second, it would be helpful to have an example of what durations,
> packet intervals and timestamps look like at the rollover points
> where sub-events are created. For example, suppose the Named Event
> began at timestamp 0, has duration 72,000 (9 seconds), and RFC2833
> packets are sent out at 50msec intervals. Assume that associated
> audio packets are muted during the event. Is this sequence of packets
> correct?

Please see if the new edition will help.

> 
> Relative     Sequence      Time Time (ms)    Number        Stamp
> Duration     Comment 
> ---------------------------------------------------------- 0
> X             0         400        this is the first packet of the
> event 50         X+1           0         800        this is the
> second packet of the event .....
> 8.xxx seconds pass with packets sent 8100         X+162         0
> 65200       last packet before rollover/subevent 8141.875     X+163
> 0        65535       last packet of event with duration FFFF note
> that it comes at non-periodic interval 8150         X+164     65535
> 65       this is the first packet of the subevent 8200         X+165
> 65535          465       now the sequence continues from here... 
> .....
> 
> What if, because of DSP sampling intervals, it is impossible for the
> transmitting gateway to send packet X+163 at the rollover instant?
> What should the transmitting gateway do to minimize jitter perceived
> at the receiving gateway?
> 
> If this example is incorrect, what is the appropriate sequence?
> 
> 
> Regards,
> 
> Ann Rhodes



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



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Date: Wed, 18 Jun 2003 16:18:12 -0400
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Folks,

The AVT working group is scheduled for one slot at the 57th IETF meeting in
Vienna.  On the tentative agenda, this is from 9:00 to 11:30 on Monday 14th
July.  This date is subject to change - DO NOT USE IT TO PLAN TRAVEL.

Those wanting agenda time at the meeting should contact the working group
chairs by 1st July. Submitting a draft does not automatically provide
agenda time. 

You are reminded that the cut-off date for -00 Internet Draft submissions
is Monday, 23rd June at 9:00am EDT (GMT -04:00). Unless you have prior
approval from the chairs, -00 submissions should have filenames with the
format draft-yourname-avt-...-00.txt. Be advised that updates to -00 drafts
received after 23rd June will not be accepted. 

All other Internet-Draft submissions must be made by Monday, 30th June at
9:00am EDT.

Drafts which are submitted late and do not make the cutoff date will not be
discussed at the meeting.

-- 
Colin Perkins
csp@csperkins.org

_______________________________________________
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Thread-Topic: Question about RFC 2833
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From: "Rhodes, Ann" <ann.rhodes@intel.com>
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Subject: [AVT] Questions about draft-ietf-avt-rfc2833bis-02 and sub-events
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These questions are based on the draft-ietf-avt-rfc2833bis-02 document, specifically in the area of sub-events for DTMF digits lasting longer than 8.192 seconds.

1. First, this draft of the specification indicates it "expires" in April, 2003. Does that mean that this draft is now the accepted standard since the comment period has expired?

2. Second, it would be helpful to have an example of what durations, packet intervals and timestamps look like at the rollover points where sub-events are created. For example, suppose the Named Event began at timestamp 0, has duration 72,000 (9 seconds), and RFC2833 packets are sent out at 50msec intervals. Assume that associated audio packets are muted during the event. Is this sequence of packets correct?

Relative     Sequence      Time     
Time (ms)    Number        Stamp     Duration     Comment
----------------------------------------------------------
     0         X             0         400        this is the first packet of the event
    50         X+1           0         800        this is the second packet of the event
 .....                                            8.xxx seconds pass with packets sent
  8100         X+162         0        65200       last packet before rollover/subevent
  8141.875     X+163         0        65535       last packet of event with duration FFFF
                                                  note that it comes at non-periodic interval
  8150         X+164     65535           65       this is the first packet of the subevent
  8200         X+165     65535          465       now the sequence continues from here...
 .....

What if, because of DSP sampling intervals, it is impossible for the transmitting gateway to send packet X+163 at the rollover instant?  What should the transmitting gateway do to minimize jitter perceived at the receiving gateway?

If this example is incorrect, what is the appropriate sequence?      

Regards,

Ann Rhodes


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Subject: [AVT] I-D ACTION:draft-chen-rtp-bv-00.txt
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: RTP Payload Format for BroadVoice Speech Codecs
	Author(s)	: J. Chen et al.
	Filename	: draft-chen-rtp-bv-00.txt
	Pages		: 12
	Date		: 2003-6-18
	
This document describes the RTP payload format for the 
BroadVoice(TM) narrowband and wideband speech codecs developed by 
Broadcom Corporation.  The document also provides specifications for 
the use of BroadVoice with MIME and SDP.

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http://www.ietf.org/internet-drafts/draft-chen-rtp-bv-00.txt

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



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---------------
The IESG has received a request from Audio/Video Transport WG to consider 
the following Internet-Draft(s) as a Proposed Standard. 
o Audio/Video Transport (AVT): RTP Control Protocol Extended Reports 
  (RTCP XR) 
    <draft-ietf-avt-rtcp-report-extns-06.txt>
    Proposed Standard
                                                                                       

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2003-07-03.
                                                                                       
Files can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-avt-rtcp-report-extns-06.txt

_______________________________________________
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avt@ietf.org
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.


	Title		: RTP Payload Format for 3GPP Timed Text
	Author(s)	: J. Rey et al.
	Filename	: draft-rey-avt-3gpp-timed-text-00.txt
	Pages		: 11
	Date		: 2003-6-20
	
This document specifies an RTP payload format for the transmission of 
3GPP timed text.  Timed text is defined as a part of 3GP file format.  
It is currently used for downloading timed text contents with or 
without audio/video contents.  In the following sections the problems 
of streaming timed text are addressed and a means for streaming 3GPP 
timed text over RTP is specified.

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http://www.ietf.org/internet-drafts/draft-rey-avt-3gpp-timed-text-00.txt

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Hi everyone,

        Looks like the I-D announcement below probably should
have been posted to the list but wasn't ... 

----



A New Internet-Draft is available from the on-line Internet-Drafts directories.


        Title           : RTP Payload Format for 1394/61883 Isochronous Streams
        Author(s)       : B. Fairman
        Filename        : draft-fairman-rtp-61883-00.txt
        Pages           : 11
        Date            : 2003-6-19
        
This document describes a payload format for transporting IEC 61883-1[2]
(CIP) compliant IEEE1394 isochronous transport data using RTP. A 1394[3]
CIP isochronous transport may carry a stream format, such as DV[4],
AM824[5] or MPEG[6], that has been packetized for isochronous transport
by the source. The format is opaque to the transport mechanism. The
isochronous transport clock is derived from the 1394 cycle timer clock.
This protocol may be used to transport 1394 61883 streams between 1394
buses by IP, specifically Ethernet/IP.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-fairman-rtp-61883-00.txt

----


-------------------------------------------------------------------------
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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Subject: [AVT] timed text RTP payload
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Remarks & questions

[
        2.  Transmit the text sample size, sample duration and sample
   description index in-band.  In 3GP format this information is
   included in the header part.  In RTP it is important to transmit it
   in-band because this information might change from sample to sample.
]

I think you should explicit that it is not a dangerous thing to do because
this information affects only the data in the packet
so when you loose a packet you do not loose the context...

it is rather obviously true for size and duration,
but less obvious for description index, i.e. as you
mention it in the section 1.2 a timed text stream
may transport samples using several different "descriptions"
(i.e. configurations) and it is therefore required
to indicate for each sample the "number" (index) of the
corresponding description.

BTW the sentence "In 3GP format this information is
   included in the header part" is misleading in this respect...

**************


[

      4.  Enable the multiplexing of text samples in one single RTP
   packet.  In a mobile communication environment a typical text sample
   size is around 100 bytes.  Thus, multiplexing several text samples
   makes the transport over RTP more efficient.
]


I think you should replace "multiplex" with "aggregation",
multiplex hints at something much more "drastic" than what you
want to achieve i.e. mixing several different streams together,
which comes with a whole lot of difficult issues etc :-)

**************

Questions

*  How does a receiver know if the first element of the payload is a FHDR or a THDR ?

* Figure 3.2 has two cases, both are 32 bits aligned, is it a requirement ?
otherwise how is the header aligned (with padding ?)

* section 4 *recommends* redundancy, I would leave that more open ...
anyway would you consider the possibility that the payload format itself
would enable redundancy by simply allowing "repeat"
(of, say some highly valuable "sentences" ?)


regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com



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



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To: <philippe.gentric@philips.com>
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Subject: RE: [AVT] timed text RTP payload
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Thanks Philippe for your comments,

please see my answers inline...




> -----Original Message-----
> From: avt-admin@ietf.org [mailto:avt-admin@ietf.org]On Behalf Of
> philippe.gentric@philips.com
> Sent: Monday, June 23, 2003 11:37 AM
> To: avt@ietf.org
> Subject: [AVT] timed text RTP payload
>
>
>
> Remarks & questions
>
> [
>         2.  Transmit the text sample size, sample duration and sample
>    description index in-band.  In 3GP format this information is
>    included in the header part.  In RTP it is important to transmit it
>    in-band because this information might change from sample
> to sample.
> ]
>
> I think you should explicit that it is not a dangerous thing
> to do because
> this information affects only the data in the packet
> so when you loose a packet you do not loose the context...
>

yes. this is explained in the section 1.2 with

"..and the Sample to Chunk Box is mapped to the field SIDX (Text Sample
Entry Index).  The Sample to Chunk Box (stsc) associates the text sample
and its corresponding sample description entry in the Sample Description
Box (stsd, see below).  The Sample to Chunk Box can be used to associate
a text sample with a sample description entry...
"

If more than that is needed, I can add some comment when ellaborating on
the resilient transport section.

> it is rather obviously true for size and duration,
> but less obvious for description index, i.e. as you
> mention it in the section 1.2 a timed text stream
> may transport samples using several different "descriptions"
> (i.e. configurations) and it is therefore required
> to indicate for each sample the "number" (index) of the
> corresponding description.
>

exactly..

> BTW the sentence "In 3GP format this information is
>    included in the header part" is misleading in this respect...
>
> **************

Not only that... it is even incorrect :)

Thanks for the catch!

>
>
> [
>
>       4.  Enable the multiplexing of text samples in one single RTP
>    packet.  In a mobile communication environment a typical
> text sample
>    size is around 100 bytes.  Thus, multiplexing several text samples
>    makes the transport over RTP more efficient.
> ]
>
>
> I think you should replace "multiplex" with "aggregation",
> multiplex hints at something much more "drastic" than what you
> want to achieve i.e. mixing several different streams together,
> which comes with a whole lot of difficult issues etc :-)
>

"issue" is a word I don't like :) I'll change it to "aggregation".

> **************
>
> Questions
>
> *  How does a receiver know if the first element of the
> payload is a FHDR or a THDR ?
>

You look at the payload type field in the RTP header:

Section 3:

"The RTP sender implementing this payload format sends fragmented and
non-fragmented text samples using two different payload types, i.e.
payload type multiplexing, which are mapped dynamically.  For this
purpose, a new parameter is specified in this document for SDP,
"fragment", see Section SDP.  The receiver recognises a fragmented text
sample by the payload type value.  Note that this fact does not conflict
with Section 5.2 of RTP [5] because it is the same media that is being
transmitted.
"

If the RTP packet cotnains non-fragmented samples then there's only
THDRs, otherwise FHDR is always before THDR (see Fig. 1.2.)


> * Figure 3.2 has two cases, both are 32 bits aligned, is it a
> requirement ?
> otherwise how is the header aligned (with padding ?)

Not at all. For example if Length(SIDX)=Length(SLEN)=Length(SDUR)=8
bits, the header is 24bits long and this is a valid combination.

>
> * section 4 *recommends* redundancy, I would leave that more open ...
> anyway would you consider the possibility that the payload
> format itself
> would enable redundancy by simply allowing "repeat"
> (of, say some highly valuable "sentences" ?)
>

how about something like:

"3GPP Timed Text operates at low bit rates.  For this reason the use of
redundancy is RECOMMENDED.  Redundancy can be achieved in different
manners. For example, redundant information can be piggybacked as per
RFC 2198 [6], coded into new packets achieving FEC (Forward Error
Correction) as per RFC 2733 [7] or the sending application MAY decide to
repeat some text sample fragments it deems important to provide packet
loss resiliency.  Furthermore, other redundancy schemes not listed here
might be applicable and defined in other documents."

looking forward to your comments,

Jose

>
> regards,
>
>
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt



Jose Rey____ mailto:rey@panasonic.de
Mobile Multimedia Team -  MCOM Group
Panasonic European Laboratories GmbH
Tel: +49(0)6103-766-134
Fax: +49(0)6103-766-166
http://www.pel.panasonic.de
____________________________________


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



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Subject: [AVT] IESG Review of draft-ietf-srtp-08.txt - another set of comments
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SRTP is on the agenda of the IESG again this week.
Some more comments on SRTP may still come in, but I thought
it would be worth sending on those of Russ Housley, the second Security
AD.  Steve Bellovin has supported the draft.  Russ has comments that 
seem straightforward to address.  Wait for more comments before 
sending in a revision, but so far things are going well.

Allison

> 
> >                     Yes    No-Objection  Discuss *  Abstain
> >
> >Russ Housley        [   ]     [   ]       [ X ]      [   ]
> 
> I have six comments.
> 
> 1.  In section 1, spell out the first use of RTCP.
> 
> 2.  I find the structure of section 2 confusing.  I had to read it twice to 
> understand it.  I think that a second level of indenting would be one way 
> to fix it.  I am sure there are others.
> 
> 3.  In section 3.1, in the paragraph after figure 1, please delete:
> 
>     "It is exact for the pre-defined transforms."
> 
> This point is made more clearly later in the paragraph.  Then, at the end 
> of the same paragraph, the document says:
> 
>     "While it could seem more attractive to specify a fixed padding
>     scheme for all transforms, security and flexibility of transform
>     specifications REQUIRE that each transform specify a secure
>     padding method."
> 
> I disagree.  IPsec and S/MIME both specify padding schemes that are 
> employed by all of the ciphers.  Please reword.  Do not use "REQUIRE" in 
> the replacement.
> 
> 4.  In section 3.2.1, the document says: "the master key(s), which MUST be 
> random and kept secret."  This is quite a high bar.  I suggest that 
> pseudo-random is sufficient.  Please see the RFC 2828 definitions of 
> "random" and "pseudo-random."  The document says that it is following these 
> terms.  Also, see RFC 1750.  Similarly, the requirement on the master salt 
> should also be reduced to pseudo-random.
> 
> 5.  The last paragraph is section 3.2.3 says:
> 
>    "If no valid context can be found for a packet corresponding to a
>     certain context identifier, that packet MUST be discarded from
>     further SRTP processing."
> 
> Elsewhere, "discarded from further processing" is used.  This seems better 
> to me.  It is discarded, which seems different that no further SRTP 
> processing.  Also, SRTCP should be covered by this statement.
> 
> 6.  In section 4.1.1, I am confused by the term "fixed key" in the following:
> 
>     "For a fixed Counter Mode key, each IV value used as an input MUST be
>     distinct, in order to avoid the security exposure of a two-time pad
>     situation (Section 9.1).  To satisfy this constraint, an
>     implementation MUST ensure that the values of the SRTP packet index
>     of ROC || SEQ, and the SSRC used in the construction of the IV are
>     distinct for any fixed key.  The failure to ensure this uniqueness
>     could  be catastrophic for Secure RTP.  This is in contrast to the
>     situation for RTP itself, which may be able to tolerate such
>     failures.  It is RECOMMENDED that, if a dedicated security module is
>     present, the RTP sequence numbers and SSRC either be generated or
>     checked by that module (i.e., sequence-number and SSRC processing in
>     an SRTP system needs to be protected as well as the key). "
> 
> I think that "fixed" means "any particular key" but I am not sure.
> 
> 
> 
> 

_______________________________________________
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>
To: Allison Mankin <mankin@psg.com>
Cc: mats.naslund@era.ericsson.se, mbaugher@cisco.com,
        rolf.blom@era.ericsson.se, Elisabetta.Carrara@era.ericsson.se,
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        avt@ietf.org
Subject: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another set of
 comments
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[inline]

--> Allison Mankin <mankin@psg.com> writes:
> SRTP is on the agenda of the IESG again this week.
> Some more comments on SRTP may still come in, but I thought
> it would be worth sending on those of Russ Housley, the second Security
> AD.  Steve Bellovin has supported the draft.  Russ has comments that 
> seem straightforward to address.  Wait for more comments before 
> sending in a revision, but so far things are going well.
> 
> Allison
> 
> > 
> > >                     Yes    No-Objection  Discuss *  Abstain
> > >
> > >Russ Housley        [   ]     [   ]       [ X ]      [   ]
> > 
> > I have six comments.
> > 
> > 1.  In section 1, spell out the first use of RTCP.
> > 
> > 2.  I find the structure of section 2 confusing.  I had to read it
> > twice to understand it.  I think that a second level of indenting would
> > be one way to fix it.  I am sure there are others.
> > 
> > 3.  In section 3.1, in the paragraph after figure 1, please delete:
> > 
> >     "It is exact for the pre-defined transforms."
> > 
> > This point is made more clearly later in the paragraph.  Then, at the
> > end of the same paragraph, the document says:
> > 
> >     "While it could seem more attractive to specify a fixed padding
> >     scheme for all transforms, security and flexibility of transform
> >     specifications REQUIRE that each transform specify a secure
> >     padding method."
> > 
> > I disagree.  IPsec and S/MIME both specify padding schemes that are 
> > employed by all of the ciphers.  Please reword.  Do not use "REQUIRE"
> > in the replacement.

If IPsec and S/MIME both define a standard padding mechanism, why cannot
SRTP do the same?  And, perhaps, use the standard RTP padding mechanism?

-- 
Colin Perkins
csp@csperkins.org

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



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Subject: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another set of
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Hi,

Colin Perkins wrote:
> [inline]
> 
> --> Allison Mankin <mankin@psg.com> writes:
> 
>>SRTP is on the agenda of the IESG again this week.
>>Some more comments on SRTP may still come in, but I thought
>>it would be worth sending on those of Russ Housley, the second Security
>>AD.  Steve Bellovin has supported the draft.  Russ has comments that 
>>seem straightforward to address.  Wait for more comments before 
>>sending in a revision, but so far things are going well.
>>
>>Allison
>>
>>
>>>>                    Yes    No-Objection  Discuss *  Abstain
>>>>
>>>>Russ Housley        [   ]     [   ]       [ X ]      [   ]
>>>
>>>I have six comments.
>>>
>>>1.  In section 1, spell out the first use of RTCP.
>>>
>>>2.  I find the structure of section 2 confusing.  I had to read it
>>>twice to understand it.  I think that a second level of indenting would
>>>be one way to fix it.  I am sure there are others.
>>>
>>>3.  In section 3.1, in the paragraph after figure 1, please delete:
>>>
>>>    "It is exact for the pre-defined transforms."
>>>
>>>This point is made more clearly later in the paragraph.  Then, at the
>>>end of the same paragraph, the document says:
>>>
>>>    "While it could seem more attractive to specify a fixed padding
>>>    scheme for all transforms, security and flexibility of transform
>>>    specifications REQUIRE that each transform specify a secure
>>>    padding method."
>>>
>>>I disagree.  IPsec and S/MIME both specify padding schemes that are 
>>>employed by all of the ciphers.  Please reword.  Do not use "REQUIRE"
>>>in the replacement.
>>
> 
> If IPsec and S/MIME both define a standard padding mechanism, why cannot
> SRTP do the same?  And, perhaps, use the standard RTP padding mechanism?
> 


To comply with Russ' comment, I have no objection to deleting
"REQUIRE". I would hesitate to go beyond that.

We will open up new issues in doing so. Do we use the IPsec pad?
The S/MIME pad? Perhaps none of them is general enough? What about
new modes? (If I remember correctly, e.g. XCBC had some padding issues.)
David referenced a paper showing that some common pad schemes
were "weak". (Even if these weaknesses are "theoretical", we have
made great effort to make SRTP "maximally" secure, and I would hate
to sub-optimize at this point.)

It seems to me that the only future proof approach is to let each
transform specify its own padding, or at least point to were a
padding spec. can be found.

/Mats


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



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To: Mats =?iso-8859-1?Q?N=E4slund?= <mats.naslund@era.ericsson.se>,
        Colin Perkins <csp@csperkins.org>
From: David Mcgrew <mcgrew@cisco.com>
Subject: Re: [AVT] IESG Review of draft-ietf-srtp-08.txt - another set
  of comments
Cc: Allison Mankin <mankin@psg.com>, mbaugher@cisco.com,
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Mats, Colin,

The RTP padding is simple, and I can definitely see the value of using a=20
standard mechanism.  However, it is not clear to me that we actually need a=
=20
padding method.  AFAICT CBC is the only interesting mode that needs=20
padding.  The other 'NIST standard' encryption modes (OFB, CFB,  ECB, and=20
counter) and the new and interesting OCB mode don't.  Also, there is a CBC=
=20
encryption variant that avoids padding by switching to CFB mode to encrypt=
=20
the last block; it's been proven secure, though unfortunately NIST hasn't=20
included it in a standard, at least not explicitly.   If we used this CBC=20
variant - it would be the variant that would make the most sense for voice=
=20
traffic - then AFAICT we don't need a padding method at all.

I think that a brief discussion of the security details would be=20
useful.  If we allow chosen-plaintext attacks (e.g., the attacker can=20
manipulate the plaintexts directly or indirectly, causing the encryption of=
=20
data of her choice), then the padding method can affect=20
confidentiality.  Otherwise, I believe that it only affects message=20
authentication.  If message authentication is used (before decryption, as=20
required in SRTP), it defeats Vaudenay's attacks against CBC padding [1],=20
and likely defeats all similar attacks.  The threat is that when message=20
authentication isn't used, there are attacks that can violate=20
confidentiality.   Using the RTP padding method with CBC would provide the=
=20
same level of security as does the ESP padding: it's perfectly good unless=
=20
there's no authentication, in which case there are some=20
confidentiality-violating attacks.  At present, the two ciphers defined for=
=20
SRTP don't require padding, and aren't vulnerable to any=20
confidentiality-violating attacks even when message authentication is not=20
used, as you (Mats) point out.

One of the complicating factors is algorithms that provide message=20
authentication.  Block cipher modes of operation that provide that=20
service  (CBC MAC, XCBC MAC, OCB [2]) use their own padding schemes.   In=20
each of these modes, the padding method that is used is vital to security,=
=20
or at least is used in the security proof.   XCBC is widely regarded as the=
=20
'right' way to do CBC MAC, and is expected to be adopted by the industry,=20
and it is closely tied to its particular padding scheme.  So if we mandate=
=20
the RTP padding method for SRTP, we need to make clear that it is just for=
=20
encryption modes (that need padding), and not for authentication=20
modes.  Perhaps it would make more sense to state that encryption modes=20
that need a padding scheme SHOULD use the RTP padding method, unless they=20
need to use another method for security reasons.  OTOH, I personally would=
=20
rather see the pad-less variant of CBC, though crypto hardware vendors with=
=20
existing products might choose otherwise.

On a side note, the RTP padding could be used in conjunction with=20
encryption as a length-hiding mechanism.  The sender can hide the length of=
=20
a packet by adding padding, which the receiver will discard.  The=20
encryption will hide the last padding byte, so that an adversary cannot=20
tell by looking at a ciphertext packet what the length of the payload of=20
the corresponding plaintext packet is.  I don't think that we stated this=20
anywhere in the draft, though it seems like a legal use of RTP and SRTP to=
 me.

David

[1] Security Flaws Induced by CBC Padding - Applications to SSL, IPsec, and=
=20
WTLS, Serge Vaudenay, online at=20
http://lasecwww.epfl.ch/php_code/publications/search.php?ref=3DVau02a

[2] NIST Table of Submitted Modes of Operation, online at=20
http://csrc.nist.gov/CryptoToolkit/modes/proposedmodes/

At 11:00 AM 6/24/2003 +0200, Mats N=E4slund wrote:

>Hi,
>
>Colin Perkins wrote:
>>[inline]
>>--> Allison Mankin <mankin@psg.com> writes:
>>
>>>SRTP is on the agenda of the IESG again this week.
>>>Some more comments on SRTP may still come in, but I thought
>>>it would be worth sending on those of Russ Housley, the second Security
>>>AD.  Steve Bellovin has supported the draft.  Russ has comments that=20
>>>seem straightforward to address.  Wait for more comments before sending=
=20
>>>in a revision, but so far things are going well.
>>>
>>>Allison
>>>
>>>
>>>>>                    Yes    No-Objection  Discuss *  Abstain
>>>>>
>>>>>Russ Housley        [   ]     [   ]       [ X ]      [   ]
>>>>
>>>>I have six comments.
>>>>
>>>>1.  In section 1, spell out the first use of RTCP.
>>>>
>>>>2.  I find the structure of section 2 confusing.  I had to read it
>>>>twice to understand it.  I think that a second level of indenting would
>>>>be one way to fix it.  I am sure there are others.
>>>>
>>>>3.  In section 3.1, in the paragraph after figure 1, please delete:
>>>>
>>>>    "It is exact for the pre-defined transforms."
>>>>
>>>>This point is made more clearly later in the paragraph.  Then, at the
>>>>end of the same paragraph, the document says:
>>>>
>>>>    "While it could seem more attractive to specify a fixed padding
>>>>    scheme for all transforms, security and flexibility of transform
>>>>    specifications REQUIRE that each transform specify a secure
>>>>    padding method."
>>>>
>>>>I disagree.  IPsec and S/MIME both specify padding schemes that are=20
>>>>employed by all of the ciphers.  Please reword.  Do not use "REQUIRE"
>>>>in the replacement.
>>If IPsec and S/MIME both define a standard padding mechanism, why cannot
>>SRTP do the same?  And, perhaps, use the standard RTP padding mechanism?
>
>
>To comply with Russ' comment, I have no objection to deleting
>"REQUIRE". I would hesitate to go beyond that.
>
>We will open up new issues in doing so. Do we use the IPsec pad?
>The S/MIME pad? Perhaps none of them is general enough? What about
>new modes? (If I remember correctly, e.g. XCBC had some padding issues.)
>David referenced a paper showing that some common pad schemes
>were "weak". (Even if these weaknesses are "theoretical", we have
>made great effort to make SRTP "maximally" secure, and I would hate
>to sub-optimize at this point.)
>
>It seems to me that the only future proof approach is to let each
>transform specify its own padding, or at least point to were a
>padding spec. can be found.
>
>/Mats
>
>
>_______________________________________________
>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: [AVT] The Secure Real-time Transport Protocol : Padding/Bandwidth.
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Madam and Sirs,

I am responsible for Cipher/VoIP implementation at Mindspeed (ex Conexant).

I would like to draw your attention about padding:
draft-ietf-avt-srtp-05.txt - 4.1.1 :
<<Each of the three terms in the XOR-sum above is padded with as many
leading zeros as needed to make the operation well-defined..>>

By implementing block cipher (as AES), as you experts know already,
we have been facing the padding method consequence (RFC1423, NIST,..)
leading to increase bandwidth.

Example:
G.711 at 5ms generates a payload of 40 bytes.
AES blocks are made of 128 bits = 16 bytes.
40 / 16 = 2.5: AES will require 3 blocks increasing the encrypted payload
up to 16 x 3 =48 bytes.
Hence, AES encryption has impacted the original bandwidth consumption by
+20%.
This drawback applies to others codec G.729,...which have been designed to
save bandwidth.
This bandwidth increase is not acceptable for many Gtw manufacturers.

Mindspeed would like to submit to you a new scheme that allows encrypting
packets which size is not a multiple of AES block size without impacting
bandwidth
complying with existing modes ECB, CBC,...

Please, let me know if this new scheme could contribute to SRTP,
so I will send you related documents.

Dr Norbert Rossello.
Group Leader MindSpeed Cipher Development
norbert.rossello@mindspeed.com

Mindspeed Technologies
BP 161
06903 Sophia Antipolis Cedex - France
www.mindspeed.com




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



From exim@www1.ietf.org  Tue Jun 24 12:43:29 2003
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From: "Waterhouse, Richard" <Richard.Waterhouse@GDC4S.Com>
Subject: RE: [AVT] The Secure Real-time Transport Protocol : Padding/Bandw	idth.
To: norbert.rossello@mindspeed.com
Cc: avt@ietf.org
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We don't use SRTP, but we do use AES. We use Counter Mode. This mode doesn't
have "error extension". Thus after we have encrypted the 48 bytes we can
proceed to throw away 8 of them. When it comes time to decrypt the receiver
can just add 8 bytes of padding on the black side, decrypt, and then throw
away the padding bits on the red side. The problem you are pointing out is
not inherent to AES, it's inherent to the mode you are using with AES.

> ----------
> From:
> norbert.rossello@mindspeed.com[SMTP:norbert.rossello@mindspeed.com]
> Sent: 	Tuesday, June 24, 2003 12:08 PM
> To: 	mbaugher@cisco.com; rolf.blom@era.ericsson.se;
> elisabetta.carrara@era.ericsson.se; mcgrew@cisco.com;
> mats.naslund@era.ericsson.se; karl.norrman@era.ericsson.se;
> oran@cisco.com; avt@ietf.org
> Subject: 	[AVT] The Secure Real-time Transport Protocol :
> Padding/Bandwidth.
> 
> 
> Madam and Sirs,
> 
> I am responsible for Cipher/VoIP implementation at Mindspeed (ex
> Conexant).
> 
> I would like to draw your attention about padding:
> draft-ietf-avt-srtp-05.txt - 4.1.1 :
> <<Each of the three terms in the XOR-sum above is padded with as many
> leading zeros as needed to make the operation well-defined..>>
> 
> By implementing block cipher (as AES), as you experts know already,
> we have been facing the padding method consequence (RFC1423, NIST,..)
> leading to increase bandwidth.
> 
> Example:
> G.711 at 5ms generates a payload of 40 bytes.
> AES blocks are made of 128 bits = 16 bytes.
> 40 / 16 = 2.5: AES will require 3 blocks increasing the encrypted payload
> up to 16 x 3 =48 bytes.
> Hence, AES encryption has impacted the original bandwidth consumption by
> +20%.
> This drawback applies to others codec G.729,...which have been designed to
> save bandwidth.
> This bandwidth increase is not acceptable for many Gtw manufacturers.
> 
> Mindspeed would like to submit to you a new scheme that allows encrypting
> packets which size is not a multiple of AES block size without impacting
> bandwidth
> complying with existing modes ECB, CBC,...
> 
> Please, let me know if this new scheme could contribute to SRTP,
> so I will send you related documents.
> 
> Dr Norbert Rossello.
> Group Leader MindSpeed Cipher Development
> norbert.rossello@mindspeed.com
> 
> Mindspeed Technologies
> BP 161
> 06903 Sophia Antipolis Cedex - France
> www.mindspeed.com
> 
> 
> 
> 
> _______________________________________________
> 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: [AVT] Re: The Secure Real-time Transport Protocol : Padding/Bandwidth.
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Dear Dr Rosello,

I suspect that you have misread the spec, possibly caused
by some unclarity on our behalf, in which case I apologize.
More below.

norbert.rossello@mindspeed.com wrote:
> Madam and Sirs,
>=20
> I am responsible for Cipher/VoIP implementation at Mindspeed (ex Conexa=
nt).
>=20
> I would like to draw your attention about padding:
> draft-ietf-avt-srtp-05.txt - 4.1.1 :
> <<Each of the three terms in the XOR-sum above is padded with as many
> leading zeros as needed to make the operation well-defined..>>

Note that this does not produce padding of the cipher *output*,
it is a padding of the cipher *input*. Hence, there is no impact
on bandwidth at all (more below).

> By implementing block cipher (as AES), as you experts know already,
> we have been facing the padding method consequence (RFC1423, NIST,..)
> leading to increase bandwidth.
>=20
> Example:
> G.711 at 5ms generates a payload of 40 bytes.
> AES blocks are made of 128 bits =3D 16 bytes.
> 40 / 16 =3D 2.5: AES will require 3 blocks increasing the encrypted pay=
load
> up to 16 x 3 =3D48 bytes.
> Hence, AES encryption has impacted the original bandwidth consumption b=
y
> +20%.
> This drawback applies to others codec G.729,...which have been designed=
 to
> save bandwidth.
> This bandwidth increase is not acceptable for many Gtw manufacturers.

If you use SRTP with the default counter mode transform, here is what
will happen:

Let the codec output be denoted m (40 bytes).

1. Form the input to the cipher according to the text you copied above
    from Sect 4.1.1. Denote this (possibly padded) value by IV
    (which will always be 128 bits/16 bytes).

2. Run AES on the session key and the above input, IV, producing
    a 128 bit output which I denote s0.

3. Run AES again on IV+1, IV+2,... producing s1, s2... until at
    least 40 bytes of output has been obtained, denote this total
    output s =3D s0 || s1 || ,,, (i.e. size(s) >=3D 40 bytes).

4. Take the XOR of m and the **40 first bytes** of s, producing the
    ciphered output y (which has the same size as m, 40 bytes).
    (Disgard excess bytes of s.)

5. Transmit y.

On the receiver side, excactly the same procedure takes place,
only that "y" takes the place of "m", resulting in retrieving m
back.

>=20
> Mindspeed would like to submit to you a new scheme that allows encrypti=
ng
> packets which size is not a multiple of AES block size without impactin=
g
> bandwidth
> complying with existing modes ECB, CBC,...
>=20
> Please, let me know if this new scheme could contribute to SRTP,
> so I will send you related documents.

Since SRTP is very close to passing IESG review (I hope...) I think
adding another mode at this late stage would be highly undesireable.

Moreover, I don't see the need for it as there is no extra bandwidth
consumption implied by the already pre-defined transforms.

Best,

/Mats

----------------------------------------------
Mats N=E4slund, PhD, Senior Specialist
Communications Security Lab, Ericsson Research
SE-16480 Stockholm, Sweden
Visiting adr: Torshamnsgatan 23, Kista
Phone/Fax: (+46 8) 58533739/4047020



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



From exim@www1.ietf.org  Tue Jun 24 13:34:23 2003
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From: Euchner Martin <martin.euchner@siemens.com>
To: "'norbert.rossello@mindspeed.com'" <norbert.rossello@mindspeed.com>,
        mbaugher@cisco.com, rolf.blom@era.ericsson.se,
        elisabetta.carrara@era.ericsson.se, mcgrew@cisco.com,
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        oran@cisco.com, avt@ietf.org,
        Euchner Martin <martin.euchner@siemens.com>
Subject: RE: [AVT] The Secure Real-time Transport Protocol : Padding/Bandw
	idth.
Date: Tue, 24 Jun 2003 18:39:38 +0200
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Dear all,

I'm very confused by this entire SRTP padding discussion.

My understanding of SRTP is that the document describes a stream-cipher using AES in counter and/or F8 mode. As such, SRTP in this byte-stream mode does not need any padding of the payload, as stream-ciphers do not pad typically. (Note: One can use SRTP with AES with the 40 byte G.711/G.729 payload without any message expansion).

Since the SRTP draft ID provides the possibility of using other crypto mechanisms as an extension, padding might be an issue. Since we have not seen any such extension so far, we can't tell, if padding would be an issue nor how padding should be done for any potential future SRTP encryption mechanism.

My recommendation thus, is that SRTP padding issue be deferred to the future and be solved when there is a need for it. For now it should be sufficient to state, that padding issues should be addressed by the extension documents.



With kind regards


Martin Euchner.
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| Martin Euchner                 Phone: +49 89 722 55790
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 -----Original Message-----
From: 	norbert.rossello@mindspeed.com [mailto:norbert.rossello@mindspeed.com] 
Sent:	Tuesday, June 24, 2003 6:08 PM
To:	mbaugher@cisco.com; rolf.blom@era.ericsson.se; elisabetta.carrara@era.ericsson.se; mcgrew@cisco.com; mats.naslund@era.ericsson.se; karl.norrman@era.ericsson.se; oran@cisco.com; avt@ietf.org
Subject:	[AVT] The Secure Real-time Transport Protocol : Padding/Bandwidth.


Madam and Sirs,

I am responsible for Cipher/VoIP implementation at Mindspeed (ex Conexant).

I would like to draw your attention about padding:
draft-ietf-avt-srtp-05.txt - 4.1.1 :
<<Each of the three terms in the XOR-sum above is padded with as many
leading zeros as needed to make the operation well-defined..>>

By implementing block cipher (as AES), as you experts know already,
we have been facing the padding method consequence (RFC1423, NIST,..)
leading to increase bandwidth.

Example:
G.711 at 5ms generates a payload of 40 bytes.
AES blocks are made of 128 bits = 16 bytes.
40 / 16 = 2.5: AES will require 3 blocks increasing the encrypted payload
up to 16 x 3 =48 bytes.
Hence, AES encryption has impacted the original bandwidth consumption by
+20%.
This drawback applies to others codec G.729,...which have been designed to
save bandwidth.
This bandwidth increase is not acceptable for many Gtw manufacturers.

Mindspeed would like to submit to you a new scheme that allows encrypting
packets which size is not a multiple of AES block size without impacting
bandwidth
complying with existing modes ECB, CBC,...

Please, let me know if this new scheme could contribute to SRTP,
so I will send you related documents.

Dr Norbert Rossello.
Group Leader MindSpeed Cipher Development
norbert.rossello@mindspeed.com

Mindspeed Technologies
BP 161
06903 Sophia Antipolis Cedex - France
www.mindspeed.com




_______________________________________________
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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Date: Thu, 26 Jun 2003 17:18:17 +0300
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Thread-Topic: Q about draft-ietf-avt-rtp-new
Thread-Index: AcM77c2Nx11qh5vyEdeSdQDQt7+w2w==
From: "Oren Peleg" <OrenP@audiocodes.com>
To: "Stephen Casner" <casner@acm.org>
Cc: <avt@ietf.org>
Subject: [AVT] Q about draft-ietf-avt-rtp-new
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Hi Stephen,

Is this draft a potential RFC? or its like the avt-profile & =
avt-profile-new which one is the update of the other that is subject to =
future updates as well & therefore will never become an RFC.

Thanks,

Oren Peleg

Work phone No. : 03 - 5394167
Home phone No. : 03 - 5335913
Cell phone No. : 055 - 669613
Email : Oren.Peleg@audiocodes.com


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<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Hi Stephen,</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Is this draft a potential =
RFC? or its like the avt-profile &amp; avt-profile-new which one is the =
update of the other that is subject to future updates as well &amp; =
therefore will never become an RFC.</FONT></P>

<P DIR=3DLTR><FONT SIZE=3D2 FACE=3D"Arial">Thanks,</FONT></P>

<P DIR=3DLTR><FONT COLOR=3D"#000080" SIZE=3D4 FACE=3D"Monotype =
Corsiva">Oren Peleg</FONT></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Work phone No. : 03 - 5394167</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Home phone No. : 03 - 5335913</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT COLOR=3D"#000000" SIZE=3D2 =
FACE=3D"Book Antiqua">Cell phone No. : 055 - 669613</FONT></SPAN></P>

<P DIR=3DLTR><SPAN LANG=3D"en-us"><FONT SIZE=3D2 FACE=3D"Book =
Antiqua">Email :<U> </U></FONT><U><FONT COLOR=3D"#0000FF" SIZE=3D2 =
FACE=3D"Book =
Antiqua">Oren.Peleg@audiocodes.com</FONT></U></SPAN><U><SPAN =
LANG=3D"he"></SPAN></U><SPAN LANG=3D"he"></SPAN></P>

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



From exim@www1.ietf.org  Fri Jun 27 18:31:08 2003
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		 Fri, 27 Jun 2003 11:12:03 +0200
Subject: [AVT] CfP:  Frist European Workshop on Wireless Sensor Networks
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Dear Colleague,

I would like to draw your attention to the " Frist European Workshop
on Wireless Sensor Networks" Berlin, January 2004 (see Attachement).
Submission deadline is July 1st, 2003: submission details are available
under      http://www.ewsn.org

The submission deadline for EWSN has been extended to July 14th, 18:00 
middle European daylight savings time (corresponding to noon Eastern 
Standard Time or 9 am Pacific time). 

Best regards
Adam Wolisz 


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<div class="moz-text-html" lang="x-western">             
<div class="moz-text-html" lang="x-western">              
<div class="moz-text-html" lang="x-western">               
<p>Dear Colleague, <br>
       <br>
      I would like to draw your attention to the " Frist European Workshop
 <br>
      on Wireless Sensor Networks" Berlin, January 2004 (see Attachement).
 <br>
      Submission deadline is July 1st, 2003: submission details are available 
  <br>
      under&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <a class="moz-txt-link-freetext"
 href="http://www.ewsn.org">http://www.ewsn.org</a><br>
      <br>
        The submission deadline for EWSN has been <b> extended to July 14th,
  18:00  middle European daylight savings time </b>  (corresponding to noon
  Eastern  Standard Time or 9 am Pacific time).&nbsp;</p>
           
<p>Best regards <br>
      Adam Wolisz&nbsp;  </p>
      </div>
     </div>
    </div>
   </div>
  </div>
 </div>
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--------------070605040404060009040902--



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



From exim@www1.ietf.org  Fri Jun 27 18:31:07 2003
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To: jan.vandermeer@philips.com, dmackie@apple.com,
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Cc: casner@acm.org, magnus.westerlund@ericsson.com, csp@csperkins.org,
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Reply-To: mankin@psg.com
Date: Thu, 26 Jun 2003 21:19:00 -0700
From: Allison Mankin <mankin@psg.com>
Message-Id: <E19VkhR-000OBP-Dj@psg.com>
Subject: [AVT] IESG Review of draft-ietf-avt-mpeg4-simple-07.txt - Discuss Comments
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The IESG reviewed draft-ietf-avt-mpeg4-simple-07.txt and had a few concerns
that should be addressed before the draft can advance.  Editorially:

   Comment:  It is strange to have more than one section labeled
  "Introduction."  Please pick a new label for section 2.1.

Steve Bellovin and Ned Freed both request that a reference be given in
the Security Considerations for security model for ECMAscript.  Here
is Steve's Discuss comment:

   Where is the security model defined for ECMAScript in this context?  
   (Problems with the model have been part of the Javascript security 
   problem for Web browsers.)

Steve also asked:

   2.4:  Why is this form of application-level fragmentation better than
   IP fragmentation?

Please discuss the issues in email and we'll see if the fixes to the draft
can be done as notes rather than a revised i-d, to be quicker.

Allison








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



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Hi everyone,

	Just sent in an updated version of the RTP MIDI normative I-D
off to internet-drafts@ietf.org. You can download a copy now from:

  http://www.cs.berkeley.edu/~lazzaro/sa/pubs/txt/current-rtp-midi.txt

	In a hopeful sign, the Change Log is sufficiently short to
reproduce in full below:

  Chapter M (Appendix A.9) has been redesigned, to follow the semantic
  design of Chapters C and E.  Several definitions in Appendix A.1 have
  been changed to reflect this change, as have the chapter inclusion
  semantics for Chapter M in Appendix C.1.3.

  Many small editorial changes throughout the document, to correct
  grammatical errors and improve phrasing.

I won't be present in Vienna (unlike Billy Joel, Vienna does not wait
for me :-), but I'll be submitting a few PDF slides that update RTP
MIDI status, and will be sending a longer report based on the slides
to the WG group, as the meeting date approaches.

-------------------------------------------------------------------------
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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Subject: [AVT] IRTF p2prg ...
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Hi everyone,

	The IRTF Peer-to-Peer Research Group was announced a few
days ago:

	http://www.irtf.org/charters/p2prg.html

From the charter:

  To this end, the research group also emphasizes the following 
  near-term goals: [...] exploring interfaces to IETF protocols
  to realize the models; and offering input to the IETF as a starting
  place for possible groups standardizing new protocols that are useful
  in building P2P applications. 

	From the perspective of RTP, one way to look at this charter
is that RTP knows the abstract model it prefers for a population of
hosts -- a multicast group -- and so if a goal is to interface P2P to
RTP, p2prg should offer a multicast abstraction of some flavor.

	However, one could imagine other ways to look at this charter
with respect to RTP that are quite different ...

	In any event, the odds of an outcome useful to AVT go up if we
communicate with them early in their process, so I thought I'd post to
point interested folks in their direction ... I'm monitoring the group
(nothing much has happened yet), but I really don't have the routing
experience to be an active participant in it.

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



From exim@www1.ietf.org  Sun Jun 29 02:01:23 2003
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Subject: [AVT] RFC 2190 : Clarification
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--0-19141782-1056802004=:97067
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Hello 
 
I am working on the implementation part of RFC 2190
for H.263 video packtization.
 
I have a clarification.
 
1) As per RFC 2190, In MODE A,   beginning of Packet should contain GOB with its Header. So, this means only first GOB (or part of it) in  each packet should contain GOB header, rest GOBs will not have GOB header in the same MODE-A packet.
 
2) But H.263 standard says ( this with respective to my discussion with H.263 developers) whether all GOB's should have headers and all should not have GOB headers in a paerticular frame.
 
3) So to acheive above point 1 with respective above point 2, we need to  calculate header for all GOB's and transmit GOB header only for first GOB of each MODE - A packet.
 
Please clarify above point 3 is correct or we can calculate GOB headers for only first GOB of each MODE -A packet. So that we can save time of GOB header computation at  rest of GOBs.
 
Thanking you in Advance
 
Muni Chanra Reddy




---------------------------------
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<DIV>
<DIV><FONT face=Arial size=2>Hello </FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>I am working on the implementation part of RFC 2190<BR>for H.263 video packtization.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>I have a clarification.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>1) As per RFC 2190, In MODE A,&nbsp;&nbsp; beginning of Packet should contain GOB with its Header. So, this means only first GOB (or part of it) in&nbsp; each packet should contain GOB header, rest GOBs will not have GOB header in the same MODE-A packet.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>2) But H.263 standard says ( this with respective to my discussion with H.263 developers) whether all GOB's should have headers and all should not have GOB headers in a paerticular frame.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>3) So to acheive above point 1 with respective above point 2, we need to&nbsp; calculate header for all GOB's and transmit GOB header only for first GOB of each MODE - A packet.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Please clarify above point 3 is correct or we can calculate GOB headers for only first GOB of each MODE -A packet. So that we can save time of GOB header computation&nbsp;at &nbsp;rest of GOBs.</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Thanking you in Advance</FONT></DIV>
<DIV>&nbsp;</DIV>
<DIV><FONT face=Arial size=2>Muni Chanra Reddy<BR></FONT></DIV></DIV><p><hr SIZE=1>
Do you Yahoo!?<br>
<a href="http://pa.yahoo.com/*http://rd.yahoo.com/evt=1207/*http://promo.yahoo.com/sbc/">SBC Yahoo! DSL</a> - Now only $29.95 per month!
--0-19141782-1056802004=:97067--

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



From exim@www1.ietf.org  Sun Jun 29 13:19:34 2003
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Subject: [AVT] Re: Question on MFR1 in draft-ietf-avt-rfc2833bis-02.txt
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Zhenfu Lin wrote:

> Hello editors.
> 
> I have a couple of quesions regard MFR1.

My apologies for the delay in responding.

> 
> 1) What is the frequency pair for MF R1 S3 in Table 6 of 
> draft-ietf-avt-rfc2833bis-02.txt? (I am guessing that it is frequency
> pair 700/1700 Hz.)

This was a mistake, related to the item below.

> 2) Both KP and KP1 refer to the same frequency pair, namely 1100/1700
> Hz. In Table 6, KP is mapped to 138 and KP1 is mapped to 139. So,
> frequency pair 1100/1700 Hz maps to two codes (138 and 139). - a:
> Does this mean that one of these codes is used when a RFC 2833 packet
>  is generated? Which one should be used? - b: Does this mean that the
> receiver is prepared to receive both codes, and map both to the
> frequency pair 1100/1700 Hz?

This was a bug caused by confusion in the description and labeling of 
these tones. I've now cleaned this up and removed duplicates, so that 
each frequency pair has a single label:

MF 0\ldots 9                                & 128\ldots 137 & & yes\\
MF 700/1700  (Code 11, KP3P, ST3P)          & 138           & & yes\\
MF 1100/1700 (KP, KP1)                      & 139           & & yes\\
MF 1300/1700 (KP2, ST2P)                    & 140           & & yes\\
MF 1500/1700 (ST)                           & 141           & & yes\\
MF 900/1700 (Code 12, STP)                  & 142           & & yes\\

I've looked at a variety of sources (ITU Q.140/Q.151, Freeman, GR-506) 
and the labeling seems now self-consistent.

-03 should appear early next week.

> 
> Thank you.
> 
> Zhenfu





_______________________________________________
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A few comments in response:
=20
- RFC 2190 should only be used for H.263 that pre-date 1998 in syntax
design (i.e., no use of PLUSPTYPE header format).
=20
- RFC 2429 is a better design than 2190, and 2429 can be used for both
old and newer versions of H.263 syntax.
=20
Regarding the questions:
=20
1) I don't think RFC 2190 any prohibition against having additional GOB
headers within Mode A packets.  The RFC says it "allows fragmentation at
GOB boundaries", but I don't see anywhere where it says it *requires*
such fragmentation or prohibits GOB headers within packets.
=20
2) As far as the H.263 standard (and within RFC 2190, I believe) is
concerned, it is up to the discretion of the encoder when to send GOB
headers and when not to.  Even within a single coded picture, some GOBs
can have headers while others do not.  In some particular system
environments (e.g. in some H.32x environments), there are ways to
indicate that GOB headers should be sent or do not need to be sent for
all GOBs - but this is outside the scope of H.263 itself.  There is
nothing in the H.263 video spec itself that requires any particular
behavior for whether GOB headers are present or not.
=20
3) Given the above statements about item 1, I don't think item 3 is
especially relevant anymore.
=20
Best Regards,
=20
Gary Sullivan
=20
 -----Original Message-----
From: avt-admin@ietf.org [mailto:avt-admin@ietf.org] On Behalf Of Muni
Chandra
Sent: Saturday, June 28, 2003 5:07 AM
To: avt@ietf.org
Subject: [AVT] RFC 2190 : Clarification



Hello=20
=20
I am working on the implementation part of RFC 2190
for H.263 video packtization.
=20
I have a clarification.
=20
1) As per RFC 2190, In MODE A,   beginning of Packet should contain GOB
with its Header. So, this means only first GOB (or part of it) in  each
packet should contain GOB header, rest GOBs will not have GOB header in
the same MODE-A packet.
=20
2) But H.263 standard says ( this with respective to my discussion with
H.263 developers) whether all GOB's should have headers and all should
not have GOB headers in a paerticular frame.
=20
3) So to acheive above point 1 with respective above point 2, we need to
calculate header for all GOB's and transmit GOB header only for first
GOB of each MODE - A packet.
=20
Please clarify above point 3 is correct or we can calculate GOB headers
for only first GOB of each MODE -A packet. So that we can save time of
GOB header computation at  rest of GOBs.
=20
Thanking you in Advance
=20
Muni Chanra Reddy




  _____ =20

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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 6.00.2800.1170" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>A few=20
comments in response:</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>- RFC=20
2190 should only be used for H.263 that pre-date 1998 in syntax design =
(i.e., no=20
use of PLUSPTYPE header format).</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>- RFC=20
2429 is a better design than 2190, and 2429 can be used for both old and =
newer=20
versions of H.263 syntax.</FONT></SPAN></DIV>
<DIV><SPAN =
class=3D745265317-30062003></SPAN>&nbsp;</DIV></FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff>Regarding the questions:</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>1) I=20
don't&nbsp;think&nbsp;RFC 2190&nbsp;any prohibition&nbsp;against having=20
additional GOB headers within Mode A packets.&nbsp; The RFC says it =
"allows=20
fragmentation at GOB boundaries", but I don't see anywhere where it says =
it=20
*requires* such fragmentation or prohibits GOB headers within=20
packets.</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>2) As=20
far as the H.263 standard (and within RFC 2190, I believe) is concerned, =
it is=20
up to the discretion of the encoder when to send GOB headers and when =
not=20
to.&nbsp;&nbsp;Even within&nbsp;a single coded picture, some GOBs can =
have=20
headers while others do not.&nbsp;&nbsp;In some particular system =
environments=20
(e.g. in some H.32x environments), there are ways to indicate that GOB =
headers=20
should be sent or do not need&nbsp;to be sent for all GOBs - but this is =
outside=20
the scope of H.263 itself.&nbsp; There is nothing in the H.263 video =
spec itself=20
that requires any particular behavior for whether GOB headers are =
present or=20
not.</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>3)=20
Given the above statements about item 1, I don't think item 3 is =
especially=20
relevant anymore.</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>Best=20
Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New"=20
color=3D#0000ff></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D745265317-30062003><FONT face=3D"Courier New" =
color=3D#0000ff>Gary=20
Sullivan</FONT></SPAN></DIV>
<DIV><SPAN class=3D745265317-30062003></SPAN><FONT face=3DTahoma =
size=3D2><SPAN=20
class=3D745265317-30062003><FONT face=3D"Courier New" color=3D#0000ff=20
size=3D3>&nbsp;</FONT></SPAN></FONT></DIV>
<DIV><FONT face=3DTahoma size=3D2><SPAN=20
class=3D745265317-30062003>&nbsp;</SPAN>-----Original =
Message-----<BR><B>From:</B>=20
avt-admin@ietf.org [mailto:avt-admin@ietf.org] <B>On Behalf Of </B>Muni=20
Chandra<BR><B>Sent:</B> Saturday, June 28, 2003 5:07 AM<BR><B>To:</B>=20
avt@ietf.org<BR><B>Subject:</B> [AVT] RFC 2190 :=20
Clarification<BR><BR></DIV></FONT>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV>
  <DIV><FONT face=3DArial size=3D2>Hello </FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>I am working on the implementation =
part of RFC=20
  2190<BR>for H.263 video packtization.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>I have a clarification.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>1) As per RFC 2190, In MODE =
A,&nbsp;&nbsp;=20
  beginning of Packet should contain GOB with its Header. So, this means =
only=20
  first GOB (or part of it) in&nbsp; each packet should contain GOB =
header, rest=20
  GOBs will not have GOB header in the same MODE-A packet.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>2) But H.263 standard says ( this =
with respective=20
  to my discussion with H.263 developers) whether all GOB's should have =
headers=20
  and all should not have GOB headers in a paerticular =
frame.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>3) So to acheive above point 1 with =
respective=20
  above point 2, we need to&nbsp; calculate header for all GOB's and =
transmit=20
  GOB header only for first GOB of each MODE - A packet.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Please clarify above point 3 is =
correct or we can=20
  calculate GOB headers for only first GOB of each MODE -A packet. So =
that we=20
  can save time of GOB header computation&nbsp;at &nbsp;rest of=20
  GOBs.</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Thanking you in Advance</FONT></DIV>
  <DIV>&nbsp;</DIV>
  <DIV><FONT face=3DArial size=3D2>Muni Chanra =
Reddy<BR></FONT></DIV></DIV>
  <P>
  <HR SIZE=3D1>
  Do you Yahoo!?<BR><A=20
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href=3D"http://pa.yahoo.com/*http://rd.yahoo.com/evt=3D1207/*http://promo=
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