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Date: Fri, 01 Jun 2012 09:37:26 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: Jim Barnett <Jim.Barnett@genesyslab.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Regarding the use-case document
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I'm starting to think that we need to encapsulate the identity work somehow.

It's clear that there needs to be a basic set of tools exposed from the 
lower layer RTCWEB work (chiefly DTLS, and possibly EKT) to support 
identity assertions about communications partners.

But it's clear that our current use cases document doesn't adequately 
cover requirements on what kind of identity can exist, and what 
information they need to be able to verify about each other - and that's 
a rather big undertaking to get right.

draft-ietf-rtcweb-identity-use-cases, anyone?

                   Harald

On 05/31/2012 03:04 PM, Jim Barnett wrote:
> Yes, let's set aside some time to discuss this at the interim.  When I brought up call center use cases a while ago, we reached the conclusion that the webRTC endpoint was the corporate gateway/ACD and that anything behind it (including the agent) was out of scope (which is another way of saying that it could work any way the corporation wanted.)  The result was that call centers fell under legacy inter-op and no changes were needed to the use cases.
>
> In general corporate/call center uses of webRTC will be important, so it's worth going over this once more to make sure we agree on how they fit into the framework.
>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
> Sent: Thursday, May 31, 2012 3:28 AM
> To: Ravindran, Parthasarathi
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Regarding the use-case document
>
> On 05/30/2012 11:31 AM, Ravindran, Parthasarathi wrote:
>> Stefan,
>>
>>
>> 1) I have mentioned in Call Center [2] usecase calls for "Anonymous"
>> identity for agents
>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04295.html)
>> and also Andrew Hutton mentioned the same in the another mail
>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04221.html)
>>
>> 2) Also in another mail thread, "Anonymous" identity for customer is
>> discussed  as part of
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04254.html
>>
>> In case any more information required for clarity, please let me know.
>> Also, I'm fine with discussion about these usecase in interim meeting.
> Hi Partha,
>
> I would prefer if we discussed this at the interim before adding. The main reason is that the use-case document, and requirements, currently focus on media and data streams. When doing the first draft we deliberately left things like identity handling, find-and-connect and so on out of scope.
>
> This was partly inspired by the charter (that talks about enabling innovation on top of a set of basic components - those being peer-to-peer audio, video and data) and partly from that there are already a lot of web services used every day which has solved this - they have implemented presence, text chat, messaging, ... already and in theory it would be simple to add video etc. if we just got those "basic components" in place.
>
> If the use-case document is to be updated to cover things like identity, user management and so on that would be a big change, and I would prefer that to be discussed before starting actual work.
>
> Br,
> Stefan
>
>> Thanks Partha
>>
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
>>> Sent: Wednesday, May 30, 2012 2:24 PM To: rtcweb@ietf.org Subject:
>>> [rtcweb] Regarding the use-case document
>>>
>>> The rtcweb chairs asked for input on the use-case document on April
>>> 27th [1].
>>>
>>> Browsing the emails that followed, I note:
>>>
>>> 1. Proposal for a call center use case (made by Jim Barnett) [2]
>>>
>>> 2. Proposal to add peer-to-peer file sharing to the simple video comm
>>> service (made by Tim Terriberri) [3]
>>>
>>> 3. Comments related to security, SRTP and key exchange, F20 (by Dan
>>> Wing) [4]
>>>
>>> 4. Request for clarifying "eavesdropping", Andrew Hutton [5]
>>>
>>> 5. Enterprise policies use-cases proposed, Roman Shpount [6]
>>>
>>> 6. Multiparty with low complex central node, Stefan Håkansson [7]
>>>
>>> 7. Multiparty with central node that can not decipher media Oscar
>>> Ohlsson [8]
>>>
>>> 8. Four new reqs proposed (to be mapped to either new or existing
>>> use- cases), one additional use-case proposed, by Cullen Jennings
>>> [9]. Reqs: 8a. Req: When A calls B, B does not reveal their IP
>>> address to A 8b. Req: B can tell that the encrypted media is from A
>>> and not from a MITM 8c. Req: Be able to switch off Voice activity
>>> detection (if present) 8d. Req: Be able to prioritize between streams
>>> New use-case: "Collaboration between companies with node [in media
>>> path] that does/can not decipher media"
>>>
>>> 9. Proposal from Magnus Westerlund to clarify the language a bit [10]
>>> regarding 4. in this list
>>>
>>> Is anything important missing from the list above?
>>>
>>> Going through this list (and the discussion that followed on each
>>> item) it seems quite clear that:
>>>
>>> A. The language should be clarified (as Magnus proposed [10]), e.g.
>>> to clarify that also the data channel should be encrypted (this
>>> relates to 3. and 9. in the previous list)
>>>
>>> B. One more derivative of the "Simple Video Communication Service"
>>> use-case should be added to add peer-to-peer file sharing (this
>>> proposal got some support, no objection, and was also mentioned many
>>> times during the MV interim)
>>>
>>> C. There is no need to clarify "eavesdropping" (item 4. in the
>>> previous list); EKR proposed to do that as part of the security
>>> document instead
>>>
>>> D. For the new requirements proposed by Cullen (8a - d in the
>>> previous list), 8a and 8b can be incorporated in existing use cases,
>>> 8c is already covered in the "Distributed Music Band"
>>> use-case, 8d can be added to the "Hockey Game Viewer" use case.
>>> Regarding 8d it can be noted that there is already a requirement on
>>> "being able to use priority functions in network nodes" (F24); 8d
>>> would be different and relate more to how the browser reduces send
>>> rate for different flows as a response to e.g. network congestion.
>>> The reqs seem  uncontroversial to me.
>>>
>>> Unless there are objections to this, the editors plan to update the
>>> document according to A - D above.
>>>
>>>
>>>
>>> It is less clear on what to do regarding the call-center use case
>>> [2], the enterprise policies use-cases [6] and the two use-cases
>>> where a central node (in the media path) must not be able to access
>>> deciphered media [8] and [9]; I think this should be discussed at the
>>> interim meeting.
>>>
>>> Comments?
>>>
>>> Stefan
>>>
>>> [1]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04202.html
>>> [2]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04203.html
>>> [3]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04204.html
>>> [4]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04205.html
>>> [5]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04241.html
>>> [6]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04271.html
>>> [7]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04430.html
>>> [8]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html
>>> [9]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html
>>> [10]
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04214.html
>>> _______________________________________________ rtcweb mailing list
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: den 1 juni 2012 09:37
> To: Jim Barnett
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Regarding the use-case document
>=20
> I'm starting to think that we need to encapsulate the=20
> identity work somehow.
>=20
> It's clear that there needs to be a basic set of tools=20
> exposed from the lower layer RTCWEB work (chiefly DTLS, and=20
> possibly EKT) to support identity assertions about=20
> communications partners.
>=20
> But it's clear that our current use cases document doesn't=20
> adequately cover requirements on what kind of identity can=20
> exist, and what information they need to be able to verify=20
> about each other - and that's a rather big undertaking to get right.
>=20
> draft-ietf-rtcweb-identity-use-cases, anyone?

This could easily become a Pandoras box, right? We need the hooks, but at t=
he same time
not solve the whole identity area.

Do You have any idea as to how to limit the scope of this? A proposal that =
we can discuss in the
interim meeting could perhaps be useful.

>=20
>                    Harald
>=20
> On 05/31/2012 03:04 PM, Jim Barnett wrote:
> > Yes, let's set aside some time to discuss this at the=20
> interim.  When I brought up call center use cases a while=20
> ago, we reached the conclusion that the webRTC endpoint was=20
> the corporate gateway/ACD and that anything behind it=20
> (including the agent) was out of scope (which is another way=20
> of saying that it could work any way the corporation wanted.)=20
>  The result was that call centers fell under legacy inter-op=20
> and no changes were needed to the use cases.
> >
> > In general corporate/call center uses of webRTC will be=20
> important, so it's worth going over this once more to make=20
> sure we agree on how they fit into the framework.
> >
> > - Jim
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> > Behalf Of Stefan Hakansson LK
> > Sent: Thursday, May 31, 2012 3:28 AM
> > To: Ravindran, Parthasarathi
> > Cc: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Regarding the use-case document
> >
> > On 05/30/2012 11:31 AM, Ravindran, Parthasarathi wrote:
> >> Stefan,
> >>
> >>
> >> 1) I have mentioned in Call Center [2] usecase calls for=20
> "Anonymous"
> >> identity for agents
> >> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04295.html)
> >> and also Andrew Hutton mentioned the same in the another mail
> >> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04221.html)
> >>
> >> 2) Also in another mail thread, "Anonymous" identity for=20
> customer is=20
> >> discussed  as part of=20
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04254.html
> >>
> >> In case any more information required for clarity, please=20
> let me know.
> >> Also, I'm fine with discussion about these usecase in=20
> interim meeting.
> > Hi Partha,
> >
> > I would prefer if we discussed this at the interim before=20
> adding. The main reason is that the use-case document, and=20
> requirements, currently focus on media and data streams. When=20
> doing the first draft we deliberately left things like=20
> identity handling, find-and-connect and so on out of scope.
> >
> > This was partly inspired by the charter (that talks about=20
> enabling innovation on top of a set of basic components -=20
> those being peer-to-peer audio, video and data) and partly=20
> from that there are already a lot of web services used every=20
> day which has solved this - they have implemented presence,=20
> text chat, messaging, ... already and in theory it would be=20
> simple to add video etc. if we just got those "basic=20
> components" in place.
> >
> > If the use-case document is to be updated to cover things=20
> like identity, user management and so on that would be a big=20
> change, and I would prefer that to be discussed before=20
> starting actual work.
> >
> > Br,
> > Stefan
> >
> >> Thanks Partha
> >>
> >>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
> >>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
> >>> Sent: Wednesday, May 30, 2012 2:24 PM To: rtcweb@ietf.org Subject:
> >>> [rtcweb] Regarding the use-case document
> >>>
> >>> The rtcweb chairs asked for input on the use-case=20
> document on April=20
> >>> 27th [1].
> >>>
> >>> Browsing the emails that followed, I note:
> >>>
> >>> 1. Proposal for a call center use case (made by Jim Barnett) [2]
> >>>
> >>> 2. Proposal to add peer-to-peer file sharing to the simple video=20
> >>> comm service (made by Tim Terriberri) [3]
> >>>
> >>> 3. Comments related to security, SRTP and key exchange,=20
> F20 (by Dan
> >>> Wing) [4]
> >>>
> >>> 4. Request for clarifying "eavesdropping", Andrew Hutton [5]
> >>>
> >>> 5. Enterprise policies use-cases proposed, Roman Shpount [6]
> >>>
> >>> 6. Multiparty with low complex central node, Stefan H=E5kansson [7]
> >>>
> >>> 7. Multiparty with central node that can not decipher media Oscar=20
> >>> Ohlsson [8]
> >>>
> >>> 8. Four new reqs proposed (to be mapped to either new or existing
> >>> use- cases), one additional use-case proposed, by Cullen Jennings=20
> >>> [9]. Reqs: 8a. Req: When A calls B, B does not reveal their IP=20
> >>> address to A 8b. Req: B can tell that the encrypted media=20
> is from A=20
> >>> and not from a MITM 8c. Req: Be able to switch off Voice activity=20
> >>> detection (if present) 8d. Req: Be able to prioritize between=20
> >>> streams New use-case: "Collaboration between companies=20
> with node [in=20
> >>> media path] that does/can not decipher media"
> >>>
> >>> 9. Proposal from Magnus Westerlund to clarify the language a bit=20
> >>> [10] regarding 4. in this list
> >>>
> >>> Is anything important missing from the list above?
> >>>
> >>> Going through this list (and the discussion that followed on each
> >>> item) it seems quite clear that:
> >>>
> >>> A. The language should be clarified (as Magnus proposed=20
> [10]), e.g.
> >>> to clarify that also the data channel should be encrypted (this=20
> >>> relates to 3. and 9. in the previous list)
> >>>
> >>> B. One more derivative of the "Simple Video Communication Service"
> >>> use-case should be added to add peer-to-peer file sharing (this=20
> >>> proposal got some support, no objection, and was also=20
> mentioned many=20
> >>> times during the MV interim)
> >>>
> >>> C. There is no need to clarify "eavesdropping" (item 4. in the=20
> >>> previous list); EKR proposed to do that as part of the security=20
> >>> document instead
> >>>
> >>> D. For the new requirements proposed by Cullen (8a - d in the=20
> >>> previous list), 8a and 8b can be incorporated in existing=20
> use cases,=20
> >>> 8c is already covered in the "Distributed Music Band"
> >>> use-case, 8d can be added to the "Hockey Game Viewer" use case.
> >>> Regarding 8d it can be noted that there is already a=20
> requirement on=20
> >>> "being able to use priority functions in network nodes" (F24); 8d=20
> >>> would be different and relate more to how the browser=20
> reduces send=20
> >>> rate for different flows as a response to e.g. network congestion.
> >>> The reqs seem  uncontroversial to me.
> >>>
> >>> Unless there are objections to this, the editors plan to=20
> update the=20
> >>> document according to A - D above.
> >>>
> >>>
> >>>
> >>> It is less clear on what to do regarding the call-center use case=20
> >>> [2], the enterprise policies use-cases [6] and the two use-cases=20
> >>> where a central node (in the media path) must not be able=20
> to access=20
> >>> deciphered media [8] and [9]; I think this should be discussed at=20
> >>> the interim meeting.
> >>>
> >>> Comments?
> >>>
> >>> Stefan
> >>>
> >>> [1]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04202.html
> >>> [2]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04203.html
> >>> [3]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04204.html
> >>> [4]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04205.html
> >>> [5]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04241.html
> >>> [6]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04271.html
> >>> [7]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04430.html
> >>> [8]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html
> >>> [9]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html
> >>> [10]
> >>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04214.html
> >>> _______________________________________________ rtcweb=20
> mailing list=20
> >>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From pravindran@sonusnet.com  Fri Jun  1 01:52:08 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Harald Alvestrand <harald@alvestrand.no>, Jim Barnett <Jim.Barnett@genesyslab.com>
Thread-Topic: [rtcweb] Regarding the use-case document
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Harald,

Thanks for providing support for adding identity related usecases. As Stefa=
n mentioned in the below mail, the authors deliberately excluded identity u=
secases initially. The single RTCWeb usecase document will give better pict=
ure over multiple usecase document as the solution of RTCWeb identity, secu=
rity, media has the linkage. I'll wait for usecase document Editors opinion=
 and interim discussion here.

Having said that in case it is decided not to add RTCWeb identity usecases =
within the current RTCWeb usecases document as it is big change, it will be=
 better to work for draft-ietf-rtcweb-identity-use-cases and I volunteer to=
 be one of the document reviewer & provide inputs.

Thanks
Partha



>-----Original Message-----
>From: Harald Alvestrand [mailto:harald@alvestrand.no]
>Sent: Friday, June 01, 2012 1:07 PM
>To: Jim Barnett
>Cc: Stefan Hakansson LK; Ravindran, Parthasarathi; rtcweb@ietf.org
>Subject: Re: [rtcweb] Regarding the use-case document
>
>I'm starting to think that we need to encapsulate the identity work
>somehow.
>
>It's clear that there needs to be a basic set of tools exposed from the
>lower layer RTCWEB work (chiefly DTLS, and possibly EKT) to support
>identity assertions about communications partners.
>
>But it's clear that our current use cases document doesn't adequately
>cover requirements on what kind of identity can exist, and what
>information they need to be able to verify about each other - and that's
>a rather big undertaking to get right.
>
>draft-ietf-rtcweb-identity-use-cases, anyone?
>
>                   Harald
>
>On 05/31/2012 03:04 PM, Jim Barnett wrote:
>> Yes, let's set aside some time to discuss this at the interim.  When I
>brought up call center use cases a while ago, we reached the conclusion
>that the webRTC endpoint was the corporate gateway/ACD and that anything
>behind it (including the agent) was out of scope (which is another way
>of saying that it could work any way the corporation wanted.)  The
>result was that call centers fell under legacy inter-op and no changes
>were needed to the use cases.
>>
>> In general corporate/call center uses of webRTC will be important, so
>it's worth going over this once more to make sure we agree on how they
>fit into the framework.
>>
>> - Jim
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Stefan Hakansson LK
>> Sent: Thursday, May 31, 2012 3:28 AM
>> To: Ravindran, Parthasarathi
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Regarding the use-case document
>>
>> On 05/30/2012 11:31 AM, Ravindran, Parthasarathi wrote:
>>> Stefan,
>>>
>>>
>>> 1) I have mentioned in Call Center [2] usecase calls for "Anonymous"
>>> identity for agents
>>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04295.html)
>>> and also Andrew Hutton mentioned the same in the another mail
>>> (http://www.ietf.org/mail-archive/web/rtcweb/current/msg04221.html)
>>>
>>> 2) Also in another mail thread, "Anonymous" identity for customer is
>>> discussed  as part of
>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04254.html
>>>
>>> In case any more information required for clarity, please let me
>know.
>>> Also, I'm fine with discussion about these usecase in interim
>meeting.
>> Hi Partha,
>>
>> I would prefer if we discussed this at the interim before adding. The
>main reason is that the use-case document, and requirements, currently
>focus on media and data streams. When doing the first draft we
>deliberately left things like identity handling, find-and-connect and so
>on out of scope.
>>
>> This was partly inspired by the charter (that talks about enabling
>innovation on top of a set of basic components - those being peer-to-
>peer audio, video and data) and partly from that there are already a lot
>of web services used every day which has solved this - they have
>implemented presence, text chat, messaging, ... already and in theory it
>would be simple to add video etc. if we just got those "basic
>components" in place.
>>
>> If the use-case document is to be updated to cover things like
>identity, user management and so on that would be a big change, and I
>would prefer that to be discussed before starting actual work.
>>
>> Br,
>> Stefan
>>
>>> Thanks Partha
>>>
>>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
>>>> Sent: Wednesday, May 30, 2012 2:24 PM To: rtcweb@ietf.org Subject:
>>>> [rtcweb] Regarding the use-case document
>>>>
>>>> The rtcweb chairs asked for input on the use-case document on April
>>>> 27th [1].
>>>>
>>>> Browsing the emails that followed, I note:
>>>>
>>>> 1. Proposal for a call center use case (made by Jim Barnett) [2]
>>>>
>>>> 2. Proposal to add peer-to-peer file sharing to the simple video
>>>> comm service (made by Tim Terriberri) [3]
>>>>
>>>> 3. Comments related to security, SRTP and key exchange, F20 (by Dan
>>>> Wing) [4]
>>>>
>>>> 4. Request for clarifying "eavesdropping", Andrew Hutton [5]
>>>>
>>>> 5. Enterprise policies use-cases proposed, Roman Shpount [6]
>>>>
>>>> 6. Multiparty with low complex central node, Stefan H=E5kansson [7]
>>>>
>>>> 7. Multiparty with central node that can not decipher media Oscar
>>>> Ohlsson [8]
>>>>
>>>> 8. Four new reqs proposed (to be mapped to either new or existing
>>>> use- cases), one additional use-case proposed, by Cullen Jennings
>>>> [9]. Reqs: 8a. Req: When A calls B, B does not reveal their IP
>>>> address to A 8b. Req: B can tell that the encrypted media is from A
>>>> and not from a MITM 8c. Req: Be able to switch off Voice activity
>>>> detection (if present) 8d. Req: Be able to prioritize between
>>>> streams New use-case: "Collaboration between companies with node [in
>>>> media path] that does/can not decipher media"
>>>>
>>>> 9. Proposal from Magnus Westerlund to clarify the language a bit
>>>> [10] regarding 4. in this list
>>>>
>>>> Is anything important missing from the list above?
>>>>
>>>> Going through this list (and the discussion that followed on each
>>>> item) it seems quite clear that:
>>>>
>>>> A. The language should be clarified (as Magnus proposed [10]), e.g.
>>>> to clarify that also the data channel should be encrypted (this
>>>> relates to 3. and 9. in the previous list)
>>>>
>>>> B. One more derivative of the "Simple Video Communication Service"
>>>> use-case should be added to add peer-to-peer file sharing (this
>>>> proposal got some support, no objection, and was also mentioned many
>>>> times during the MV interim)
>>>>
>>>> C. There is no need to clarify "eavesdropping" (item 4. in the
>>>> previous list); EKR proposed to do that as part of the security
>>>> document instead
>>>>
>>>> D. For the new requirements proposed by Cullen (8a - d in the
>>>> previous list), 8a and 8b can be incorporated in existing use cases,
>>>> 8c is already covered in the "Distributed Music Band"
>>>> use-case, 8d can be added to the "Hockey Game Viewer" use case.
>>>> Regarding 8d it can be noted that there is already a requirement on
>>>> "being able to use priority functions in network nodes" (F24); 8d
>>>> would be different and relate more to how the browser reduces send
>>>> rate for different flows as a response to e.g. network congestion.
>>>> The reqs seem  uncontroversial to me.
>>>>
>>>> Unless there are objections to this, the editors plan to update the
>>>> document according to A - D above.
>>>>
>>>>
>>>>
>>>> It is less clear on what to do regarding the call-center use case
>>>> [2], the enterprise policies use-cases [6] and the two use-cases
>>>> where a central node (in the media path) must not be able to access
>>>> deciphered media [8] and [9]; I think this should be discussed at
>>>> the interim meeting.
>>>>
>>>> Comments?
>>>>
>>>> Stefan
>>>>
>>>> [1]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04202.html
>>>> [2]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04203.html
>>>> [3]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04204.html
>>>> [4]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04205.html
>>>> [5]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04241.html
>>>> [6]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04271.html
>>>> [7]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04430.html
>>>> [8]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html
>>>> [9]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04461.html
>>>> [10]
>>>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04214.html
>>>> _______________________________________________ rtcweb mailing list
>>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>


From fluffy@iii.ca  Fri Jun  1 09:39:44 2012
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To: Cameron Byrne <cb.list6@gmail.com>
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I'm interested in this use case.=20

It seems like if the the use of the work happening in =
public-webcrypto@w3.org to access things like SIM cards, combined with =
the Identity Proxy work here, might be able to solve a use case like =
this without any private information leaving the SIIM card. I've got no =
concrete idea on how to do this but seems like it might be possible...

Cullen


On May 29, 2012, at 11:02 AM, Cameron Byrne wrote:

> Is there yet a pointer on how a SIM / IMS based identity will work
> with the webRTC web server for authentication?
>=20
> I think the obvious approach is for the SIM secret key to be bound to
> a username / password on the web server, and the web server proxies
> the authentication and registration into IMS (if you give me the right
> username credentials via the web, i will register you into IMS with
> the secret key), but that requires the undesirable step of exposing
> the SIM secret key to the web server or someplace outside the HSS/HLR
> for binding with the username password.
>=20
> Thoughts?  Pointers?
>=20
> Thanks,
>=20
> CB
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From rbarnes@bbn.com  Fri Jun  1 10:40:45 2012
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+1=20

WebCrypto is accepting use cases now, and this would be a good one.

--Richard



On Jun 1, 2012, at 12:39 PM, Cullen Jennings wrote:

>=20
> I'm interested in this use case.=20
>=20
> It seems like if the the use of the work happening in =
public-webcrypto@w3.org to access things like SIM cards, combined with =
the Identity Proxy work here, might be able to solve a use case like =
this without any private information leaving the SIIM card. I've got no =
concrete idea on how to do this but seems like it might be possible...
>=20
> Cullen
>=20
>=20
> On May 29, 2012, at 11:02 AM, Cameron Byrne wrote:
>=20
>> Is there yet a pointer on how a SIM / IMS based identity will work
>> with the webRTC web server for authentication?
>>=20
>> I think the obvious approach is for the SIM secret key to be bound to
>> a username / password on the web server, and the web server proxies
>> the authentication and registration into IMS (if you give me the =
right
>> username credentials via the web, i will register you into IMS with
>> the secret key), but that requires the undesirable step of exposing
>> the SIM secret key to the web server or someplace outside the HSS/HLR
>> for binding with the username password.
>>=20
>> Thoughts?  Pointers?
>>=20
>> Thanks,
>>=20
>> CB
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Cary.Bran@plantronics.com  Fri Jun  1 16:00:33 2012
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From: "Bran, Cary" <Cary.Bran@plantronics.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Focused Review of RTP usage draft
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Hello RTC-Web WG,

Below are some comments/feedback that Charles Eckel and I had from our
focused review of the WebRTC RTP usage draft -
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-02

Regards,

Cary and Charles


Overall feedback:

The draft covers some very important topics and is very thorough and well
written.

A theme throughout the draft is supporting legacy devices. From our read
we were unsure what position this draft is is with regards to interworking
with legacy devices/systems, sections 3.1 (translator), 4.2, 4.5 do not
seem to readily support interworking, while section 4.4 does.  We would
ask the authors to please clarify their position.

Section: Abstract
Please consider revising the first sentence of the abstract and first
sentence of paragraph two of the introduction as it is a little unclear to
the reader.
A possible suggestion for revision would be (take it or leave it): The Web
Real-Time Communication (WebRTC) framework provides support for native
interactive voice and video communications between web browsers.

Section: 2
For the terminology section - why not just adopt the usual RFC 2119
language?

Assuming the decision moving forward is to define new terms, please note
the following comment.
Currently RECOMMENDED is defined as, "Should be included as its brings
significant benefit, but the solution can potentially work without it."
This definition is too vague.
A suggested rewording is, RECOMMENDED: "Should be included as it brings
significant benefit; however, the solution can work without it, though
potentially in a more limited manner."


Section: 3.1
Supported RTP Topologies

Overall: To educate the reader with some of the issues with the proposed
topologies, it may be worthwhile (but maybe not possible in ASCII art) to
show the number of sessions instead of a bidirectional <----> notation.
For example in Figure 4: RTP Translator (Relay) with Only Unicast Paths I
my mental model of the topology was this (I removed the "D" endpoint for
clarity).

+---+                          +---+







|   |----> +------------+ <----|   |
                    | A | <-C--|            | --A->| B |
+---+ <-B--|            | --C->+---+
           | Translator |
+---+----> |            |
|   |<-B-- |            |
                    | C |<-A-- |            |
                    +---+      +------------+






Multi-unicast:
It may be worth expanding this topology to include scenarios in which it
is not a full mesh, such as a sender sending to a subset of 2 or more
receivers, or a few senders sending to the complete set of receivers. The
basic idea being that you may have some participants that are receive
only, or send only.
An example scenario could be using a recording service to record a Google
hangout session.


Mixer: Another downside is that there is a typically a loss of information
(e.g. the loss of video or a participant that is not speaking, or a loss
of audio of speaker not selected as an active
speaker)

Transport Translator: Where the draft reads, "It simply forwards the media
from all other to all the other", it should instead read, "It simply
forwards the media from each endpoint to all other endpoints."The
scalability and bandwidth concerns are similar to the multi-unicast
topology, with merely the distribution burden shifted to a centralized
node.

Near the end of the section, it reads, "The intention is that the presence
of the translator is transparent to A, however it is not certain that is
possible." This is a good goal. Hopefully subsequent versions of this
draft can expand on it.  The very last sentence of the this section may
want to be reworded, a suggestion here would be: "The scenario above is
included as a reference and discussion point for future mechanisms using a
Translator to support legacy end-points interoperating within a WebRTC
environment."

Section: 3.2

Perhaps add a reference to section 5 here?


Section: 4.3
Choice of RTP Payload Formats
This section is not clear, particularly the exception related to
multiplexing. One thing that needs to be kept in mind, and perhaps should
be stated explicitly, is the handling outlined in RFC 3264:

   7. Offerer Processing of the Answer

   When the offerer receives the answer, it MAY send media on the
   accepted stream(s) (assuming it is listed as sendrecv or recvonly in
   the answer).  It MUST send using a media format listed in the answer,
   and it SHOULD use the first media format listed in the answer when it
   does send.
   :
   :
   The offerer MAY immediately cease listening for media formats that
   were listed in the initial offer, but not present in the answer.

Section: 5
Overall feedback.  This is a great section and very informative.  As I
understand RTP, both sides have to implement the extensions in order for
the extensions to be used.  This currently this section is not requiring
any of the extensions to be implemented, my question to the authors is
what is the future purpose for this section?  Are some of these extensions
going to become mandatory in a future revision of the draft?

Section: 5.2
Header Extensions
The last two paragraphs of this section include introductory text for
sections 5.2.1-5.2.3.  The information contained within these paragraphs
is redundant to and could fall out of sync with sections 5.2.1-5.2.3. We
would recommend that the authors chop out introductory text to avoid
duplication and potential contradictions in future revisions of the draft.

Section: 6.1
Retransmission
Expand the last paragraph to include avoiding overly aggressive
retransmission schemes at the expense of being a bad internet citizen.

Section: 6.2.3
Recommendations for FEC
The recommendation for FEC is not clearly stated in this section.  It
would be valuable to the reader to have a recommendation based upon
results from experiments/analysis of FIR/PLI vs. retransmission/FEC, plus
some guidance as to when either rmethod is appropriate.

Section: 7.2
Some expansion for the reader is needed here.  A big help would be to
clarify the problem via example and then to propose (if there is) a
solution to the problem.





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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard


-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt



http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt



In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.



Richard



-----Original Message-----

From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]

Sent: Sunday, June 03, 2012 5:16 PM

To: Ejzak, Richard P (Richard)

Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt



A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.



Filename:   draft-ejzak-avtcore-rtp-subsessions

Revision:   00

Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions

Creation date:    2012-06-04

WG ID:            Individual Submission

Number of pages: 9



Abstract:

   This document describes a means of multiplexing RTP streams having

   different media types within a single transport connection and how to

   represent this multiplexing option in SDP.  This mechanism is an

   alternative to the BUNDLE and SHIM proposals currently under active

   consideration in AVTCORE.  Instead of adding an extra multiplexing

   header as in SHIM to allow multiple RTP sessions within a single

   transport connection, or using the payload type field to separate

   different media streams within a single RTP session, this document

   describes how to partition the existing SSRC space to create RTP

   subsessions from a single RTP session.  A filter can be used to

   identify each RTP subsession for different QoS handling as necessary.

   RTP subsessions can be treated like RTP sessions with a few

   restrictions.  In particular, SSRC mapping may be needed when

   forwarding RTP streams into an RTP subsession to avoid SSRC

   conflicts, but there are few use cases in which this limitation is a

   concern and RTP subsessions can be disabled if necessary.









The IETF Secretariat




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<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">Please see the mail attached below that I just sent to a=
vtcore announcing a personal ID I just submitted on the topic of media mult=
iplexing. &nbsp;I am sending this
 notice to rtcweb since this is the group that triggered recent work on RTP=
 multiplexing.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">I also request the rtcweb chairs consider giving me some=
 time to present this work at next week&#8217;s interim meeting if there is=
 time on the agenda. &nbsp;It might be
 useful to discuss the ideas in the ID before avtcore meets in <st1:place w=
:st=3D"on">
<st1:City w:st=3D"on">Vancouver</st1:City></st1:place>.<o:p></o:p></span></=
font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">Richard<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">-----Original Message-----<br>
From: Ejzak, Richard P (Richard) <br>
Sent: Sunday, June 03, 2012 6:08 PM<br>
To: 'avt@ietf.org'<br>
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt=
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><a href=3D"http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessio=
ns-00.txt">http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.tx=
t</a><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">In response to the chairs' request for additional input on the mult=
i session issue, I have submitted this draft for your consideration.&nbsp; =
There are superficial similarities
 with an expired draft from last year in http://tools.ietf.org/id/draft-ros=
enberg-rtcweb-rtpmux-00.txt, but my draft has a different take on the use o=
f SSRC as the basis for multi session multiplexing that I think is superior=
 and worthy of consideration as
 a potential replacement for BUNDLE and/or SHIM.&nbsp; Please comment.<o:p>=
</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Richard<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">-----Original Message-----<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Sent: Sunday, June 03, 2012 5:16 PM<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">To: Ejzak, Richard P (Richard)<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Subject: New Version Notification for draft-ejzak-avtcore-rtp-subse=
ssions-00.txt<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt ha=
s been successfully submitted by Richard Ejzak and posted to the IETF repos=
itory.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Filename:&nbsp;&nbsp; draft-ejzak-avtcore-rtp-subsessions<o:p></o:p=
></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Revision:&nbsp;&nbsp; 00<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; Media multiplexing with Real-time Transport Protocol (RTP) subsession=
s<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Creation date:&nbsp;&nbsp;&nbsp; 2012-06-04<o:p></o:p></span></font=
></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; Individual Submission<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Number of pages: 9<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Abstract:<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; This document describes a means of multiplexing RTP st=
reams having<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; different media types within a single transport connec=
tion and how to<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; represent this multiplexing option in SDP.&nbsp; This =
mechanism is an<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; alternative to the BUNDLE and SHIM proposals currently=
 under active<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; consideration in AVTCORE.&nbsp; Instead of adding an e=
xtra multiplexing<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; header as in SHIM to allow multiple RTP sessions withi=
n a single<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; transport connection, or using the payload type field =
to separate<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; different media streams within a single RTP session, t=
his document<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; describes how to partition the existing SSRC space to =
create RTP<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; subsessions from a single RTP session.&nbsp; A filter =
can be used to<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; identify each RTP subsession for different QoS handlin=
g as necessary.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; RTP subsessions can be treated like RTP sessions with =
a few<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; restrictions.&nbsp; In particular, SSRC mapping may be=
 needed when<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; forwarding RTP streams into an RTP subsession to avoid=
 SSRC<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; conflicts, but there are few use cases in which this l=
imitation is a<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; concern and RTP subsessions can be disabled if necessa=
ry.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">The IETF Secretariat<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">&nbsp;<o:p></o:p></span></font></p>
</div>
</body>
</html>

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From christer.holmberg@ericsson.com  Mon Jun  4 00:22:41 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 4 Jun 2012 09:22:34 +0200
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism  to indicate/negotiate such usage (in your case the ssrc-prefix could p=
erhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard


-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt



http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt



In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.



Richard



-----Original Message-----

From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org]<mailto:[mailto:internet-drafts@ietf.org]>

Sent: Sunday, June 03, 2012 5:16 PM

To: Ejzak, Richard P (Richard)

Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt



A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.



Filename:   draft-ejzak-avtcore-rtp-subsessions

Revision:   00

Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions

Creation date:    2012-06-04

WG ID:            Individual Submission

Number of pages: 9



Abstract:

   This document describes a means of multiplexing RTP streams having

   different media types within a single transport connection and how to

   represent this multiplexing option in SDP.  This mechanism is an

   alternative to the BUNDLE and SHIM proposals currently under active

   consideration in AVTCORE.  Instead of adding an extra multiplexing

   header as in SHIM to allow multiple RTP sessions within a single

   transport connection, or using the payload type field to separate

   different media streams within a single RTP session, this document

   describes how to partition the existing SSRC space to create RTP

   subsessions from a single RTP session.  A filter can be used to

   identify each RTP subsession for different QoS handling as necessary.

   RTP subsessions can be treated like RTP sessions with a few

   restrictions.  In particular, SSRC mapping may be needed when

   forwarding RTP streams into an RTP subsession to avoid SSRC

   conflicts, but there are few use cases in which this limitation is a

   concern and RTP subsessions can be disabled if necessary.









The IETF Secretariat




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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3D"#606420"><div class=3DWordSection1><p class=3DMsoNormal><span style=
=3D'font-family:"Calibri","sans-serif";color:#1F497D'>Hi Richard,<o:p></o:p=
></span></p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sans=
-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><sp=
an style=3D'font-family:"Calibri","sans-serif";color:#1F497D'>I will not co=
mment on the RTP aspects of your proposal, but I don&#8217;t understand you=
r comments regarding BUNDLE.<o:p></o:p></span></p><p class=3DMsoNormal><spa=
n style=3D'font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o=
:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sa=
ns-serif";color:#1F497D'>The core purpose of BUNDLE is to allow the usage o=
f identical IP address:port information for multiple m- lines.<o:p></o:p></=
span></p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-se=
rif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-family:"Calibri","sans-serif";color:#1F497D'>It is true that =
BUNDLE defines that, by default, all media belong to the same RTP session. =
But, nothing prevents you from using BUNDLE also with multiple RTP sessions=
, RTP &#8220;subsession&#8221; etc. You simply need an extension mechanism =
=A0to indicate/negotiate such usage (in your case the ssrc-prefix could per=
haps be used for that).<o:p></o:p></span></p><p class=3DMsoNormal><span sty=
le=3D'font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></=
span></p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-se=
rif";color:#1F497D'>Regards,<o:p></o:p></span></p><p class=3DMsoNormal><spa=
n style=3D'font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o=
:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sa=
ns-serif";color:#1F497D'>Christer<o:p></o:p></span></p><p class=3DMsoNormal=
><span style=3D'font-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbs=
p;</o:p></span></p><div><div style=3D'border:none;border-top:solid #B5C4DF =
1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span style=3D'fon=
t-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span styl=
e=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb-bounces@ie=
tf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Ejzak, Richard =
P (Richard)<br><b>Sent:</b> 4. kes=E4kuuta 2012 2:10<br><b>To:</b> rtcweb@i=
etf.org<br><b>Subject:</b> [rtcweb] draft on media multiplexing submitted t=
o AVTCORE<o:p></o:p></span></p></div></div><p class=3DMsoNormal><o:p>&nbsp;=
</o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:=
"Arial","sans-serif"'>Please see the mail attached below that I just sent t=
o avtcore announcing a personal ID I just submitted on the topic of media m=
ultiplexing. &nbsp;I am sending this notice to rtcweb since this is the gro=
up that triggered recent work on RTP multiplexing.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans=
-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'fo=
nt-size:10.0pt;font-family:"Arial","sans-serif"'>I also request the rtcweb =
chairs consider giving me some time to present this work at next week&#8217=
;s interim meeting if there is time on the agenda. &nbsp;It might be useful=
 to discuss the ideas in the ID before avtcore meets in Vancouver.<o:p></o:=
p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-fami=
ly:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><=
span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Richard<o:=
p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;fon=
t-family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoPl=
ainText>-----Original Message-----<br>From: Ejzak, Richard P (Richard) <br>=
Sent: Sunday, June 03, 2012 6:08 PM<br>To: 'avt@ietf.org'<br>Subject: [AVTC=
ORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt<o:p></o:p></p=
><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText><a hr=
ef=3D"http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt">ht=
tp://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt</a><o:p></o=
:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText=
>In response to the chairs' request for additional input on the multi sessi=
on issue, I have submitted this draft for your consideration.&nbsp; There a=
re superficial similarities with an expired draft from last year in <a href=
=3D"http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt">http://t=
ools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt</a>, but my draft has=
 a different take on the use of SSRC as the basis for multi session multipl=
exing that I think is superior and worthy of consideration as a potential r=
eplacement for BUNDLE and/or SHIM.&nbsp; Please comment.<o:p></o:p></p><p c=
lass=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Richard<o:=
p></o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlai=
nText>-----Original Message-----<o:p></o:p></p><p class=3DMsoPlainText>From=
: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> =
<a href=3D"mailto:[mailto:internet-drafts@ietf.org]">[mailto:internet-draft=
s@ietf.org]</a> <o:p></o:p></p><p class=3DMsoPlainText>Sent: Sunday, June 0=
3, 2012 5:16 PM<o:p></o:p></p><p class=3DMsoPlainText>To: Ejzak, Richard P =
(Richard)<o:p></o:p></p><p class=3DMsoPlainText>Subject: New Version Notifi=
cation for draft-ejzak-avtcore-rtp-subsessions-00.txt<o:p></o:p></p><p clas=
s=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>A new version=
 of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been successfully s=
ubmitted by Richard Ejzak and posted to the IETF repository.<o:p></o:p></p>=
<p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Filena=
me:&nbsp;&nbsp; draft-ejzak-avtcore-rtp-subsessions<o:p></o:p></p><p class=
=3DMsoPlainText>Revision:&nbsp;&nbsp; 00<o:p></o:p></p><p class=3DMsoPlainT=
ext>Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
; Media multiplexing with Real-time Transport Protocol (RTP) subsessions<o:=
p></o:p></p><p class=3DMsoPlainText>Creation date:&nbsp;&nbsp;&nbsp; 2012-0=
6-04<o:p></o:p></p><p class=3DMsoPlainText>WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Individual Submission<o:p></o:p></=
p><p class=3DMsoPlainText>Number of pages: 9<o:p></o:p></p><p class=3DMsoPl=
ainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText>Abstract:<o:p></o:p></=
p><p class=3DMsoPlainText>&nbsp;&nbsp; This document describes a means of m=
ultiplexing RTP streams having<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;=
&nbsp; different media types within a single transport connection and how t=
o<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;&nbsp; represent this multipl=
exing option in SDP.&nbsp; This mechanism is an<o:p></o:p></p><p class=3DMs=
oPlainText>&nbsp;&nbsp; alternative to the BUNDLE and SHIM proposals curren=
tly under active<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;&nbsp; conside=
ration in AVTCORE.&nbsp; Instead of adding an extra multiplexing<o:p></o:p>=
</p><p class=3DMsoPlainText>&nbsp;&nbsp; header as in SHIM to allow multipl=
e RTP sessions within a single<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;=
&nbsp; transport connection, or using the payload type field to separate<o:=
p></o:p></p><p class=3DMsoPlainText>&nbsp;&nbsp; different media streams wi=
thin a single RTP session, this document<o:p></o:p></p><p class=3DMsoPlainT=
ext>&nbsp;&nbsp; describes how to partition the existing SSRC space to crea=
te RTP<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;&nbsp; subsessions from =
a single RTP session.&nbsp; A filter can be used to<o:p></o:p></p><p class=
=3DMsoPlainText>&nbsp;&nbsp; identify each RTP subsession for different QoS=
 handling as necessary.<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;&nbsp; =
RTP subsessions can be treated like RTP sessions with a few<o:p></o:p></p><=
p class=3DMsoPlainText>&nbsp;&nbsp; restrictions.&nbsp; In particular, SSRC=
 mapping may be needed when<o:p></o:p></p><p class=3DMsoPlainText>&nbsp;&nb=
sp; forwarding RTP streams into an RTP subsession to avoid SSRC<o:p></o:p><=
/p><p class=3DMsoPlainText>&nbsp;&nbsp; conflicts, but there are few use ca=
ses in which this limitation is a<o:p></o:p></p><p class=3DMsoPlainText>&nb=
sp;&nbsp; concern and RTP subsessions can be disabled if necessary.<o:p></o=
:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText=
>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></p><p class=3DMsoPlainTex=
t><o:p>&nbsp;</o:p></p><p class=3DMsoPlainText><o:p>&nbsp;</o:p></p><p clas=
s=3DMsoPlainText>The IETF Secretariat<o:p></o:p></p><p class=3DMsoNormal><s=
pan style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'><o:p>&nbsp;=
</o:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-=
family:"Arial","sans-serif"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNorm=
al><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;=
<o:p></o:p></span></p></div></body></html>=

--_000_7F2072F1E0DE894DA4B517B93C6A05852C45A1C2DDESESSCMS0356e_--

From richard.ejzak@alcatel-lucent.com  Mon Jun  4 06:34:21 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Hi Christer,
I agree that BUNDLE in its current form would be consistent with the RTP su=
bsessions proposal, but it would also be technically redundant.  I'm not su=
re exactly which comments in the paper you object to.  I agree with you abo=
ut the purpose of BUNDLE and that it works (with limitations) for its inten=
ded purpose.  My understanding is that BUNDLE is primarily intended for the=
 case where a single RTP session is shared across m lines.  There is no nee=
d for BUNDLE (and specifically the constrained PT assignment) when you use =
another mechanism such as SHIM or RTP subsessions to segregate the flows as=
sociated with each m line.  Whether the grouping framework is applicable is=
 open for discussion.

So perhaps we should separately discuss the grouping framework and constrai=
ned PT assignment procedures associated with BUNDLE to be sure we understan=
d one another.  The source of our disagreement may be more what is meant by=
 "BUNDLE" than anything else.  I have no problem redefining it if there is =
agreement to do so!

I look forward to an in-depth discussion of these issues in Stockholm.

BR, Richard

________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism  to indicate/negotiate such usage (in your case the ssrc-prefix could p=
erhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard


-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt



http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt



In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.



Richard



-----Original Message-----

From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org]<mailto:%5bmailto:internet-drafts@ietf.org%5d>

Sent: Sunday, June 03, 2012 5:16 PM

To: Ejzak, Richard P (Richard)

Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt



A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.



Filename:   draft-ejzak-avtcore-rtp-subsessions

Revision:   00

Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions

Creation date:    2012-06-04

WG ID:            Individual Submission

Number of pages: 9



Abstract:

   This document describes a means of multiplexing RTP streams having

   different media types within a single transport connection and how to

   represent this multiplexing option in SDP.  This mechanism is an

   alternative to the BUNDLE and SHIM proposals currently under active

   consideration in AVTCORE.  Instead of adding an extra multiplexing

   header as in SHIM to allow multiple RTP sessions within a single

   transport connection, or using the payload type field to separate

   different media streams within a single RTP session, this document

   describes how to partition the existing SSRC space to create RTP

   subsessions from a single RTP session.  A filter can be used to

   identify each RTP subsession for different QoS handling as necessary.

   RTP subsessions can be treated like RTP sessions with a few

   restrictions.  In particular, SSRC mapping may be needed when

   forwarding RTP streams into an RTP subsession to avoid SSRC

   conflicts, but there are few use cases in which this limitation is a

   concern and RTP subsessions can be disabled if necessary.









The IETF Secretariat




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<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Hi Christer,<o:p></o:p></span></font><=
/p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">I agree that BUNDLE in its current for=
m would be consistent with the RTP subsessions proposal, but it would also =
be technically redundant.
 &nbsp;I&#8217;m not sure exactly which comments in the paper you object to=
. &nbsp;I agree with you about the purpose of BUNDLE and that it works (wit=
h limitations) for its intended purpose. &nbsp;My understanding is that BUN=
DLE is primarily intended for the case where a single
 RTP session is shared across m lines. &nbsp;There is no need for BUNDLE (a=
nd specifically the constrained PT assignment) when you use another mechani=
sm such as SHIM or RTP subsessions to segregate the flows associated with e=
ach m line. &nbsp;Whether the grouping framework
 is applicable is open for discussion.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">So perhaps we should separately discus=
s the grouping framework and constrained PT assignment procedures associate=
d with BUNDLE to be
 sure we understand one another.&nbsp; The source of our disagreement may b=
e more what is meant by &#8220;BUNDLE&#8221; than anything else. &nbsp;I ha=
ve no problem redefining it if there is agreement to do so!<o:p></o:p></spa=
n></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">I look forward to an in-depth discussi=
on of these issues in
<st1:City w:st=3D"on"><st1:place w:st=3D"on">Stockholm</st1:place></st1:Cit=
y>.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">BR, Richard<o:p></o:p></span></font></=
p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" face=3D"Times New Roman"><span style=3D"font-size:12.0pt;font-f=
amily:&quot;Times New Roman&quot;">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> Chri=
ster Holmberg [mailto:christer.holmberg@ericsson.com]
<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Monday, June 04, 2012 =
2:23 AM<br>
<b><span style=3D"font-weight:bold">To:</span></b> Ejzak, Richard P (Richar=
d); rtcweb@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> RE: [rtcweb] draft =
on media multiplexing submitted to AVTCORE</span></font><font size=3D"3" fa=
ce=3D"Times New Roman"><span style=3D"font-size:12.0pt;font-family:&quot;Ti=
mes New Roman&quot;"><o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"FuturaA Bk BT"><span style=
=3D"font-size:
11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">Hi Richa=
rd,<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">I will n=
ot comment on the RTP aspects of your proposal, but I don&#8217;t understan=
d your comments regarding BUNDLE.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">The core=
 purpose of BUNDLE is to allow the usage of identical IP address:port infor=
mation for multiple m- lines.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">It is tr=
ue that BUNDLE defines that, by default, all media belong to the same RTP s=
ession. But, nothing prevents you from using
 BUNDLE also with multiple RTP sessions, RTP &#8220;subsession&#8221; etc. =
You simply need an extension mechanism &nbsp;to indicate/negotiate such usa=
ge (in your case the ssrc-prefix could perhaps be used for that).<o:p></o:p=
></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">Regards,=
<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D">Christer=
<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"#1f497d" face=3D"Calibri">=
<span style=3D"font-size:11.0pt;font-family:Calibri;color:#1F497D"><o:p>&nb=
sp;</o:p></span></font></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><font size=3D"2" face=3D"Tahoma"><span style=3D"f=
ont-size:10.0pt;
font-family:Tahoma;font-weight:bold">From:</span></font></b><font size=3D"2=
" face=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma"> rtcw=
eb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b><span style=3D"font-weight:bold">On Behalf Of </span></b>Ejzak, Richard =
P (Richard)<br>
<b><span style=3D"font-weight:bold">Sent:</span></b> 4. kes=E4kuuta 2012 2:=
10<br>
<b><span style=3D"font-weight:bold">To:</span></b> rtcweb@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> [rtcweb] draft on m=
edia multiplexing submitted to AVTCORE<o:p></o:p></span></font></p>
</div>
</div>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"FuturaA Bk BT"><span style=
=3D"font-size:
11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">Please see the mail attached below that I just sent to a=
vtcore announcing a personal ID I just submitted on the topic of media mult=
iplexing. &nbsp;I am sending this
 notice to rtcweb since this is the group that triggered recent work on RTP=
 multiplexing.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">I also request the rtcweb chairs consider giving me some=
 time to present this work at next week&#8217;s interim meeting if there is=
 time on the agenda. &nbsp;It might be
 useful to discuss the ideas in the ID before avtcore meets in <st1:City w:=
st=3D"on">
<st1:place w:st=3D"on">Vancouver</st1:place></st1:City>.<o:p></o:p></span><=
/font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial">Richard<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" face=3D"Arial"><span style=3D"font-=
size:10.0pt;
font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">-----Original Message-----<br>
From: Ejzak, Richard P (Richard) <br>
Sent: Sunday, June 03, 2012 6:08 PM<br>
To: 'avt@ietf.org'<br>
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt=
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><a href=3D"http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessio=
ns-00.txt">http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.tx=
t</a><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">In response to the chairs' request for additional input on the mult=
i session issue, I have submitted this draft for your consideration.&nbsp; =
There are superficial similarities
 with an expired draft from last year in <a href=3D"http://tools.ietf.org/i=
d/draft-rosenberg-rtcweb-rtpmux-00.txt">
http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt</a>, but my d=
raft has a different take on the use of SSRC as the basis for multi session=
 multiplexing that I think is superior and worthy of consideration as a pot=
ential replacement for BUNDLE and/or
 SHIM.&nbsp; Please comment.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Richard<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">-----Original Message-----<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">From:
<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> <a=
 href=3D"mailto:%5bmailto:internet-drafts@ietf.org%5d">
[mailto:internet-drafts@ietf.org]</a> <o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Sent: Sunday, June 03, 2012 5:16 PM<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">To: Ejzak, Richard P (Richard)<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Subject: New Version Notification for draft-ejzak-avtcore-rtp-subse=
ssions-00.txt<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt ha=
s been successfully submitted by Richard Ejzak and posted to the IETF repos=
itory.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Filename:&nbsp;&nbsp; draft-ejzak-avtcore-rtp-subsessions<o:p></o:p=
></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Revision:&nbsp;&nbsp; 00<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; Media multiplexing with Real-time Transport Protocol (RTP) subsession=
s<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Creation date:&nbsp;&nbsp;&nbsp; 2012-06-04<o:p></o:p></span></font=
></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp; Individual Submission<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Number of pages: 9<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">Abstract:<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; This document describes a means of multiplexing RTP st=
reams having<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; different media types within a single transport connec=
tion and how to<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; represent this multiplexing option in SDP.&nbsp; This =
mechanism is an<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; alternative to the BUNDLE and SHIM proposals currently=
 under active<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; consideration in AVTCORE.&nbsp; Instead of adding an e=
xtra multiplexing<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; header as in SHIM to allow multiple RTP sessions withi=
n a single<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; transport connection, or using the payload type field =
to separate<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; different media streams within a single RTP session, t=
his document<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; describes how to partition the existing SSRC space to =
create RTP<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; subsessions from a single RTP session.&nbsp; A filter =
can be used to<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; identify each RTP subsession for different QoS handlin=
g as necessary.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; RTP subsessions can be treated like RTP sessions with =
a few<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; restrictions.&nbsp; In particular, SSRC mapping may be=
 needed when<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; forwarding RTP streams into an RTP subsession to avoid=
 SSRC<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; conflicts, but there are few use cases in which this l=
imitation is a<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp; concern and RTP subsessions can be disabled if necessa=
ry.<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
=3D"font-size:
10.0pt"><o:p>&nbsp;</o:p></span></font></p>
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<p class=3D"MsoPlainText"><font size=3D"2" face=3D"Courier New"><span style=
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From christer.holmberg@ericsson.com  Mon Jun  4 06:54:04 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 4 Jun 2012 15:54:00 +0200
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi,

> I agree that BUNDLE in its current form would be consistent with the RTP =
subsessions proposal, but it would also be technically redundant. =A0I'm no=
t sure exactly which comments in the paper you object to. =A0I agree with y=
ou about the purpose of BUNDLE=20
> and that it works (with limitations) for its intended purpose. =A0My unde=
rstanding is that BUNDLE is primarily intended for the case where a single =
RTP session is shared across m lines.

That is not true. That is only the "default" that we chose, and the limitat=
ions in the draft are related to the case when you use a single RTP session=
 - they are not related to the BUNDLE mechanism as such.

>=A0There is no need for BUNDLE (and specifically the constrained PT assign=
ment) when you use another mechanism such as SHIM or RTP subsessions to seg=
regate the=20
> flows associated with each m line. =A0Whether the grouping framework is a=
pplicable is open for discussion.

You still need something in order to use identical IP address:port values f=
or multiple m- lines.

> So perhaps we should separately discuss the grouping framework and constr=
ained PT assignment procedures associated with BUNDLE to be sure we underst=
and one another.=A0
> The source of our disagreement may be more what is meant by "BUNDLE" than=
 anything else. =A0I have no problem redefining it if there is agreement to=
 do so!

We should separate the discussions on how the multiplexing is done on the R=
TP level (which I think your draft mostly focuses on) and how it is signale=
d in SDP.

Regards,

Christer



________________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism =A0to indicate/negotiate such usage (in your case the ssrc-prefix could=
 perhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing. =A0I am s=
ending this notice to rtcweb since this is the group that triggered recent =
work on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda. =A0I=
t might be useful to discuss the ideas in the ID before avtcore meets in Va=
ncouver.

Richard

-----Original Message-----
From: Ejzak, Richard P (Richard)=20
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt

http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt

In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.=A0 There are s=
uperficial similarities with an expired draft from last year in http://tool=
s.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a diff=
erent take on the use of SSRC as the basis for multi session multiplexing t=
hat I think is superior and worthy of consideration as a potential replacem=
ent for BUNDLE and/or SHIM.=A0 Please comment.

Richard

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: Sunday, June 03, 2012 5:16 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt

A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.

Filename:=A0=A0 draft-ejzak-avtcore-rtp-subsessions
Revision:=A0=A0 00
Title:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Media multiplexing with Real-time T=
ransport Protocol (RTP) subsessions
Creation date:=A0=A0=A0 2012-06-04
WG ID:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Individual Submission
Number of pages: 9

Abstract:
=A0=A0 This document describes a means of multiplexing RTP streams having
=A0=A0 different media types within a single transport connection and how t=
o
=A0=A0 represent this multiplexing option in SDP.=A0 This mechanism is an
=A0=A0 alternative to the BUNDLE and SHIM proposals currently under active
=A0=A0 consideration in AVTCORE.=A0 Instead of adding an extra multiplexing
=A0=A0 header as in SHIM to allow multiple RTP sessions within a single
=A0=A0 transport connection, or using the payload type field to separate
=A0=A0 different media streams within a single RTP session, this document
=A0=A0 describes how to partition the existing SSRC space to create RTP
=A0=A0 subsessions from a single RTP session.=A0 A filter can be used to
=A0=A0 identify each RTP subsession for different QoS handling as necessary=
.
=A0=A0 RTP subsessions can be treated like RTP sessions with a few
=A0=A0 restrictions.=A0 In particular, SSRC mapping may be needed when
=A0=A0 forwarding RTP streams into an RTP subsession to avoid SSRC
=A0=A0 conflicts, but there are few use cases in which this limitation is a
=A0=A0 concern and RTP subsessions can be disabled if necessary.

=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=20


The IETF Secretariat


=A0

From richard.ejzak@alcatel-lucent.com  Mon Jun  4 07:55:07 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Date: Mon, 4 Jun 2012 14:54:55 +0000
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi Christer,
I agree with you that we should separate the discussion of how multiplexing=
 is done at the RTP level from how it is signaled in SDP.  My draft address=
es both aspects by describing how to partition the SSRC space between m lin=
es at the RTP level and how to signal that use in SDP using a new "ssrc-pre=
fix" attribute.

With the use of the ssrc-prefix attribute to identify RTP subsessions it is=
 not absolutely necessary to use the grouping framework to indicate bundlin=
g, since the ssrc-prefix attribute implicitly indicates the bundling.  Thes=
e two mechanisms are compatible but redundant.

BR, Richard=20

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Monday, June 04, 2012 8:54 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi,

> I agree that BUNDLE in its current form would be consistent with the RTP =
subsessions proposal, but it would also be technically redundant. =A0I'm no=
t sure exactly which comments in the paper you object to. =A0I agree with y=
ou about the purpose of BUNDLE=20
> and that it works (with limitations) for its intended purpose. =A0My unde=
rstanding is that BUNDLE is primarily intended for the case where a single =
RTP session is shared across m lines.

That is not true. That is only the "default" that we chose, and the limitat=
ions in the draft are related to the case when you use a single RTP session=
 - they are not related to the BUNDLE mechanism as such.

>=A0There is no need for BUNDLE (and specifically the constrained PT assign=
ment) when you use another mechanism such as SHIM or RTP subsessions to seg=
regate the=20
> flows associated with each m line. =A0Whether the grouping framework is a=
pplicable is open for discussion.

You still need something in order to use identical IP address:port values f=
or multiple m- lines.

> So perhaps we should separately discuss the grouping framework and constr=
ained PT assignment procedures associated with BUNDLE to be sure we underst=
and one another.=A0
> The source of our disagreement may be more what is meant by "BUNDLE" than=
 anything else. =A0I have no problem redefining it if there is agreement to=
 do so!

We should separate the discussions on how the multiplexing is done on the R=
TP level (which I think your draft mostly focuses on) and how it is signale=
d in SDP.

Regards,

Christer



________________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism =A0to indicate/negotiate such usage (in your case the ssrc-prefix could=
 perhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing. =A0I am s=
ending this notice to rtcweb since this is the group that triggered recent =
work on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda. =A0I=
t might be useful to discuss the ideas in the ID before avtcore meets in Va=
ncouver.

Richard

-----Original Message-----
From: Ejzak, Richard P (Richard)=20
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt

http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt

In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.=A0 There are s=
uperficial similarities with an expired draft from last year in http://tool=
s.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a diff=
erent take on the use of SSRC as the basis for multi session multiplexing t=
hat I think is superior and worthy of consideration as a potential replacem=
ent for BUNDLE and/or SHIM.=A0 Please comment.

Richard

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]=20
Sent: Sunday, June 03, 2012 5:16 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt

A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.

Filename:=A0=A0 draft-ejzak-avtcore-rtp-subsessions
Revision:=A0=A0 00
Title:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Media multiplexing with Real-time T=
ransport Protocol (RTP) subsessions
Creation date:=A0=A0=A0 2012-06-04
WG ID:=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Individual Submission
Number of pages: 9

Abstract:
=A0=A0 This document describes a means of multiplexing RTP streams having
=A0=A0 different media types within a single transport connection and how t=
o
=A0=A0 represent this multiplexing option in SDP.=A0 This mechanism is an
=A0=A0 alternative to the BUNDLE and SHIM proposals currently under active
=A0=A0 consideration in AVTCORE.=A0 Instead of adding an extra multiplexing
=A0=A0 header as in SHIM to allow multiple RTP sessions within a single
=A0=A0 transport connection, or using the payload type field to separate
=A0=A0 different media streams within a single RTP session, this document
=A0=A0 describes how to partition the existing SSRC space to create RTP
=A0=A0 subsessions from a single RTP session.=A0 A filter can be used to
=A0=A0 identify each RTP subsession for different QoS handling as necessary=
.
=A0=A0 RTP subsessions can be treated like RTP sessions with a few
=A0=A0 restrictions.=A0 In particular, SSRC mapping may be needed when
=A0=A0 forwarding RTP streams into an RTP subsession to avoid SSRC
=A0=A0 conflicts, but there are few use cases in which this limitation is a
=A0=A0 concern and RTP subsessions can be disabled if necessary.

=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=20


The IETF Secretariat


=A0

From internet-drafts@ietf.org  Mon Jun  4 08:42:40 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-08.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Web Real-Time Communication Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-08.txt
	Pages           : 27
	Date            : 2012-06-04

   This document describes web based real-time communication use-cases.
   Based on the use-cases, the document also derives requirements
   related to the browser, and the API used by web applications to
   request and control media stream and data exchange services provided
   by the browser.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-08.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-08.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requiremen=
ts/


From christer.holmberg@ericsson.com  Mon Jun  4 09:10:05 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 4 Jun 2012 18:10:02 +0200
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi Richard,

> I agree with you that we should separate the discussion of how multiplexi=
ng is done at the RTP level from=20
> how it is signaled in SDP.  My draft addresses both aspects by describing=
 how to partition the SSRC space=20
> between m lines at the RTP level and how to signal that use in SDP using =
a new "ssrc-prefix" attribute.

Sure. But as far as the IP address:port re-use is concerned, I think you sh=
ould use BUNDLE.

> With the use of the ssrc-prefix attribute to identify RTP subsessions it =
is not absolutely necessary to use the=20
> grouping framework to indicate bundling, since the ssrc-prefix attribute =
implicitly indicates the bundling. =20
> These two mechanisms are compatible but redundant.

I see no reason for having two mechanisms to do the same thing, and in gene=
ral I also think we should try to avoid implicit indications. If often caus=
es troubles later on.

Using BUNDLE would also make it easier to define fallback behavior. For exa=
mple, you could say that if the SDP answerer does not support the ssrc-pref=
ix, but it does support BUNDLE, the fallback would be default BUNDLE behavi=
or.

Regards,

Christer




-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 8:54 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi,

> I agree that BUNDLE in its current form would be consistent with the RTP =
subsessions proposal, but it would also be technically redundant.  I'm not =
sure exactly which comments in the paper you object to.  I agree with you a=
bout the purpose of BUNDLE
> and that it works (with limitations) for its intended purpose.  My unders=
tanding is that BUNDLE is primarily intended for the case where a single RT=
P session is shared across m lines.

That is not true. That is only the "default" that we chose, and the limitat=
ions in the draft are related to the case when you use a single RTP session=
 - they are not related to the BUNDLE mechanism as such.

> There is no need for BUNDLE (and specifically the constrained PT assignme=
nt) when you use another mechanism such as SHIM or RTP subsessions to segre=
gate the
> flows associated with each m line.  Whether the grouping framework is app=
licable is open for discussion.

You still need something in order to use identical IP address:port values f=
or multiple m- lines.

> So perhaps we should separately discuss the grouping framework and constr=
ained PT assignment procedures associated with BUNDLE to be sure we underst=
and one another.
> The source of our disagreement may be more what is meant by "BUNDLE" than=
 anything else.  I have no problem redefining it if there is agreement to d=
o so!

We should separate the discussions on how the multiplexing is done on the R=
TP level (which I think your draft mostly focuses on) and how it is signale=
d in SDP.

Regards,

Christer



________________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism  to indicate/negotiate such usage (in your case the ssrc-prefix could p=
erhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard

-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt

http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt

In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.

Richard

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
Sent: Sunday, June 03, 2012 5:16 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt

A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.

Filename:   draft-ejzak-avtcore-rtp-subsessions
Revision:   00
Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions
Creation date:    2012-06-04
WG ID:            Individual Submission
Number of pages: 9

Abstract:
   This document describes a means of multiplexing RTP streams having
   different media types within a single transport connection and how to
   represent this multiplexing option in SDP.  This mechanism is an
   alternative to the BUNDLE and SHIM proposals currently under active
   consideration in AVTCORE.  Instead of adding an extra multiplexing
   header as in SHIM to allow multiple RTP sessions within a single
   transport connection, or using the payload type field to separate
   different media streams within a single RTP session, this document
   describes how to partition the existing SSRC space to create RTP
   subsessions from a single RTP session.  A filter can be used to
   identify each RTP subsession for different QoS handling as necessary.
   RTP subsessions can be treated like RTP sessions with a few
   restrictions.  In particular, SSRC mapping may be needed when
   forwarding RTP streams into an RTP subsession to avoid SSRC
   conflicts, but there are few use cases in which this limitation is a
   concern and RTP subsessions can be disabled if necessary.




The IETF Secretariat=

From richard.ejzak@alcatel-lucent.com  Mon Jun  4 09:35:49 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi Christer,
I just want to be clear what you mean by "BUNDLE".  I assume here that you =
refer to the use of the grouping mechanism and NOT necessarily the use of d=
isjoint payload type values?

Correct me if I am wrong:
So you propose that the SDP grouping mechanism would be used regardless of =
the RTP multiplexing scheme employed to indicate that some form of multiple=
xing is to be used and to identify which lines are grouped together.  If no=
 other mechanism is agreed, then multiplexing based on payload type is assu=
med.  If ssrc-prefix values are successfully negotiated then RTP subsession=
s are used.  If SHIM is successfully negotiated then SHIM is used.  (This a=
ll depends on which schemes are ultimately supported.)

I have no objection to this approach in principle but am not convinced that=
 we need to support all of these multiplexing schemes.

BR, Richard



-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Monday, June 04, 2012 11:10 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

> I agree with you that we should separate the discussion of how multiplexi=
ng is done at the RTP level from=20
> how it is signaled in SDP.  My draft addresses both aspects by describing=
 how to partition the SSRC space=20
> between m lines at the RTP level and how to signal that use in SDP using =
a new "ssrc-prefix" attribute.

Sure. But as far as the IP address:port re-use is concerned, I think you sh=
ould use BUNDLE.

> With the use of the ssrc-prefix attribute to identify RTP subsessions it =
is not absolutely necessary to use the=20
> grouping framework to indicate bundling, since the ssrc-prefix attribute =
implicitly indicates the bundling. =20
> These two mechanisms are compatible but redundant.

I see no reason for having two mechanisms to do the same thing, and in gene=
ral I also think we should try to avoid implicit indications. If often caus=
es troubles later on.

Using BUNDLE would also make it easier to define fallback behavior. For exa=
mple, you could say that if the SDP answerer does not support the ssrc-pref=
ix, but it does support BUNDLE, the fallback would be default BUNDLE behavi=
or.

Regards,

Christer




-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 8:54 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi,

> I agree that BUNDLE in its current form would be consistent with the RTP =
subsessions proposal, but it would also be technically redundant.  I'm not =
sure exactly which comments in the paper you object to.  I agree with you a=
bout the purpose of BUNDLE
> and that it works (with limitations) for its intended purpose.  My unders=
tanding is that BUNDLE is primarily intended for the case where a single RT=
P session is shared across m lines.

That is not true. That is only the "default" that we chose, and the limitat=
ions in the draft are related to the case when you use a single RTP session=
 - they are not related to the BUNDLE mechanism as such.

> There is no need for BUNDLE (and specifically the constrained PT assignme=
nt) when you use another mechanism such as SHIM or RTP subsessions to segre=
gate the
> flows associated with each m line.  Whether the grouping framework is app=
licable is open for discussion.

You still need something in order to use identical IP address:port values f=
or multiple m- lines.

> So perhaps we should separately discuss the grouping framework and constr=
ained PT assignment procedures associated with BUNDLE to be sure we underst=
and one another.
> The source of our disagreement may be more what is meant by "BUNDLE" than=
 anything else.  I have no problem redefining it if there is agreement to d=
o so!

We should separate the discussions on how the multiplexing is done on the R=
TP level (which I think your draft mostly focuses on) and how it is signale=
d in SDP.

Regards,

Christer



________________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism  to indicate/negotiate such usage (in your case the ssrc-prefix could p=
erhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard

-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt

http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt

In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.

Richard

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
Sent: Sunday, June 03, 2012 5:16 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt

A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.

Filename:   draft-ejzak-avtcore-rtp-subsessions
Revision:   00
Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions
Creation date:    2012-06-04
WG ID:            Individual Submission
Number of pages: 9

Abstract:
   This document describes a means of multiplexing RTP streams having
   different media types within a single transport connection and how to
   represent this multiplexing option in SDP.  This mechanism is an
   alternative to the BUNDLE and SHIM proposals currently under active
   consideration in AVTCORE.  Instead of adding an extra multiplexing
   header as in SHIM to allow multiple RTP sessions within a single
   transport connection, or using the payload type field to separate
   different media streams within a single RTP session, this document
   describes how to partition the existing SSRC space to create RTP
   subsessions from a single RTP session.  A filter can be used to
   identify each RTP subsession for different QoS handling as necessary.
   RTP subsessions can be treated like RTP sessions with a few
   restrictions.  In particular, SSRC mapping may be needed when
   forwarding RTP streams into an RTP subsession to avoid SSRC
   conflicts, but there are few use cases in which this limitation is a
   concern and RTP subsessions can be disabled if necessary.




The IETF Secretariat

From christer.holmberg@ericsson.com  Mon Jun  4 10:26:41 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 4 Jun 2012 19:26:37 +0200
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi,

> I just want to be clear what you mean by "BUNDLE".  I assume here that yo=
u refer to the use of the grouping mechanism and NOT necessarily the use of=
 disjoint payload type values?

Correct.

> Correct me if I am wrong:
> So you propose that the SDP grouping mechanism would be used regardless o=
f the RTP multiplexing scheme employed to indicate that some form of multip=
lexing is to be used and to identify which lines are grouped together.

Correct.

> If no other mechanism is agreed, then multiplexing based on payload type =
is assumed.

Yes. The details of that, however, is something that the community will hav=
e to discuss and agree upon. My point was that it would be easier to define=
 a fallback if we use a single mechanism for bundling m- lines together.

> If ssrc-prefix values are successfully negotiated then RTP subsessions ar=
e used.  If SHIM is successfully negotiated then SHIM is used.  (This all d=
epends on which schemes are ultimately supported.)

Correct.

> I have no objection to this approach in principle but am not convinced th=
at we need to support all of these multiplexing schemes.

That is a separate discussion :)

Regards,

Christer




-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 11:10 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

> I agree with you that we should separate the discussion of how multiplexi=
ng is done at the RTP level from
> how it is signaled in SDP.  My draft addresses both aspects by describing=
 how to partition the SSRC space
> between m lines at the RTP level and how to signal that use in SDP using =
a new "ssrc-prefix" attribute.

Sure. But as far as the IP address:port re-use is concerned, I think you sh=
ould use BUNDLE.

> With the use of the ssrc-prefix attribute to identify RTP subsessions it =
is not absolutely necessary to use the
> grouping framework to indicate bundling, since the ssrc-prefix attribute =
implicitly indicates the bundling.
> These two mechanisms are compatible but redundant.

I see no reason for having two mechanisms to do the same thing, and in gene=
ral I also think we should try to avoid implicit indications. If often caus=
es troubles later on.

Using BUNDLE would also make it easier to define fallback behavior. For exa=
mple, you could say that if the SDP answerer does not support the ssrc-pref=
ix, but it does support BUNDLE, the fallback would be default BUNDLE behavi=
or.

Regards,

Christer




-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 8:54 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi,

> I agree that BUNDLE in its current form would be consistent with the RTP =
subsessions proposal, but it would also be technically redundant.  I'm not =
sure exactly which comments in the paper you object to.  I agree with you a=
bout the purpose of BUNDLE
> and that it works (with limitations) for its intended purpose.  My unders=
tanding is that BUNDLE is primarily intended for the case where a single RT=
P session is shared across m lines.

That is not true. That is only the "default" that we chose, and the limitat=
ions in the draft are related to the case when you use a single RTP session=
 - they are not related to the BUNDLE mechanism as such.

> There is no need for BUNDLE (and specifically the constrained PT assignme=
nt) when you use another mechanism such as SHIM or RTP subsessions to segre=
gate the
> flows associated with each m line.  Whether the grouping framework is app=
licable is open for discussion.

You still need something in order to use identical IP address:port values f=
or multiple m- lines.

> So perhaps we should separately discuss the grouping framework and constr=
ained PT assignment procedures associated with BUNDLE to be sure we underst=
and one another.
> The source of our disagreement may be more what is meant by "BUNDLE" than=
 anything else.  I have no problem redefining it if there is agreement to d=
o so!

We should separate the discussions on how the multiplexing is done on the R=
TP level (which I think your draft mostly focuses on) and how it is signale=
d in SDP.

Regards,

Christer



________________________________________
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
Sent: Monday, June 04, 2012 2:23 AM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] draft on media multiplexing submitted to AVTCORE

Hi Richard,

I will not comment on the RTP aspects of your proposal, but I don't underst=
and your comments regarding BUNDLE.

The core purpose of BUNDLE is to allow the usage of identical IP address:po=
rt information for multiple m- lines.

It is true that BUNDLE defines that, by default, all media belong to the sa=
me RTP session. But, nothing prevents you from using BUNDLE also with multi=
ple RTP sessions, RTP "subsession" etc. You simply need an extension mechan=
ism  to indicate/negotiate such usage (in your case the ssrc-prefix could p=
erhaps be used for that).

Regards,

Christer

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ejzak, Richard P (Richard)
Sent: 4. kes=E4kuuta 2012 2:10
To: rtcweb@ietf.org
Subject: [rtcweb] draft on media multiplexing submitted to AVTCORE

Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard

-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt

http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt

In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.

Richard

-----Original Message-----
From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
Sent: Sunday, June 03, 2012 5:16 PM
To: Ejzak, Richard P (Richard)
Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt

A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.

Filename:   draft-ejzak-avtcore-rtp-subsessions
Revision:   00
Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions
Creation date:    2012-06-04
WG ID:            Individual Submission
Number of pages: 9

Abstract:
   This document describes a means of multiplexing RTP streams having
   different media types within a single transport connection and how to
   represent this multiplexing option in SDP.  This mechanism is an
   alternative to the BUNDLE and SHIM proposals currently under active
   consideration in AVTCORE.  Instead of adding an extra multiplexing
   header as in SHIM to allow multiple RTP sessions within a single
   transport connection, or using the payload type field to separate
   different media streams within a single RTP session, this document
   describes how to partition the existing SSRC space to create RTP
   subsessions from a single RTP session.  A filter can be used to
   identify each RTP subsession for different QoS handling as necessary.
   RTP subsessions can be treated like RTP sessions with a few
   restrictions.  In particular, SSRC mapping may be needed when
   forwarding RTP streams into an RTP subsession to avoid SSRC
   conflicts, but there are few use cases in which this limitation is a
   concern and RTP subsessions can be disabled if necessary.




The IETF Secretariat=

From harald@alvestrand.no  Mon Jun  4 11:42:41 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Richard,
having scanned your draft:

- I still fundamentally think that the partition of media types into 
separate RTP sessions was a design mistake that we should seek to 
rectify, not promulgate, and additional complexity like what you propose 
for perpetuating the use of "separated" sessions is not helpful. See 
draft-alvestrand-rtp-sess-neutral for details.

- When we have parts of the ecosystem that have code that is written on 
the assumption that the SSRC space is a 32-bit flat space, I think the 
partitioning of that space into subspaces as you propose is likely to 
lead to "interesting" bugs. We should reinforce the randomness and 
flatness of the SSRC space, not partition it.

- You say "The biggest problem with BUNDLE is that it is
    difficult to partition the RTP streams associated with different
    media lines since this requires filtering on sets of payload type
    values.  RTP subsessions is superior for this purpose since it is
    only necessary to screen for a single value in the first 16 bits".
This assumes that the partitioning associated with different media lines 
is a good thing.
I claim that it is not.

If there are no bigger problems than that with BUNDLE, I would prefer to 
stick with that approach.





On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote:
>
> Please see the mail attached below that I just sent to avtcore 
> announcing a personal ID I just submitted on the topic of media 
> multiplexing.  I am sending this notice to rtcweb since this is the 
> group that triggered recent work on RTP multiplexing.
>
> I also request the rtcweb chairs consider giving me some time to 
> present this work at next week's interim meeting if there is time on 
> the agenda.  It might be useful to discuss the ideas in the ID before 
> avtcore meets in Vancouver.
>
> Richard
>
> -----Original Message-----
> From: Ejzak, Richard P (Richard)
> Sent: Sunday, June 03, 2012 6:08 PM
> To: 'avt@ietf.org'
> Subject: [AVTCORE] multi session 
> draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> In response to the chairs' request for additional input on the multi 
> session issue, I have submitted this draft for your consideration.  
> There are superficial similarities with an expired draft from last 
> year in http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, 
> but my draft has a different take on the use of SSRC as the basis for 
> multi session multiplexing that I think is superior and worthy of 
> consideration as a potential replacement for BUNDLE and/or SHIM.  
> Please comment.
>
> Richard
>
> -----Original Message-----
>
> From: internet-drafts@ietf.org [mailto:internet-drafts@ietf.org]
>
> Sent: Sunday, June 03, 2012 5:16 PM
>
> To: Ejzak, Richard P (Richard)
>
> Subject: New Version Notification for 
> draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has 
> been successfully submitted by Richard Ejzak and posted to the IETF 
> repository.
>
> Filename:   draft-ejzak-avtcore-rtp-subsessions
>
> Revision:   00
>
> Title:            Media multiplexing with Real-time Transport Protocol 
> (RTP) subsessions
>
> Creation date:    2012-06-04
>
> WG ID:            Individual Submission
>
> Number of pages: 9
>
> Abstract:
>
>    This document describes a means of multiplexing RTP streams having
>
>    different media types within a single transport connection and how to
>
>    represent this multiplexing option in SDP.  This mechanism is an
>
>    alternative to the BUNDLE and SHIM proposals currently under active
>
>    consideration in AVTCORE.  Instead of adding an extra multiplexing
>
>    header as in SHIM to allow multiple RTP sessions within a single
>
>    transport connection, or using the payload type field to separate
>
>    different media streams within a single RTP session, this document
>
>    describes how to partition the existing SSRC space to create RTP
>
>    subsessions from a single RTP session.  A filter can be used to
>
>    identify each RTP subsession for different QoS handling as necessary.
>
>    RTP subsessions can be treated like RTP sessions with a few
>
>    restrictions.  In particular, SSRC mapping may be needed when
>
>    forwarding RTP streams into an RTP subsession to avoid SSRC
>
>    conflicts, but there are few use cases in which this limitation is a
>
>    concern and RTP subsessions can be disabled if necessary.
>
> The IETF Secretariat
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------020905030208080109020004
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    Richard,<br>
    having scanned your draft:<br>
    <br>
    - I still fundamentally think that the partition of media types into
    separate RTP sessions was a design mistake that we should seek to
    rectify, not promulgate, and additional complexity like what you
    propose for perpetuating the use of "separated" sessions is not
    helpful. See draft-alvestrand-rtp-sess-neutral for details.<br>
    <br>
    - When we have parts of the ecosystem that have code that is written
    on the assumption that the SSRC space is a 32-bit flat space, I
    think the partitioning of that space into subspaces as you propose
    is likely to lead to "interesting" bugs. We should reinforce the
    randomness and flatness of the SSRC space, not partition it.<br>
    <br>
    - You say "The biggest problem with BUNDLE is that it is<br>
    &nbsp;&nbsp; difficult to partition the RTP streams associated with different<br>
    &nbsp;&nbsp; media lines since this requires filtering on sets of payload type<br>
    &nbsp;&nbsp; values.&nbsp; RTP subsessions is superior for this purpose since it is<br>
    &nbsp;&nbsp; only necessary to screen for a single value in the first 16
    bits".<br>
    This assumes that the partitioning associated with different media
    lines is a good thing.<br>
    I claim that it is not.<br>
    <br>
    If there are no bigger problems than that with BUNDLE, I would
    prefer to stick with that approach.<br>
    <br>
    <br>
    <br>
    <br>
    <br>
    On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote:
    <blockquote
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          <div class="Section1">
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial">Please see the mail attached below
                  that I just sent to avtcore announcing a personal ID I
                  just submitted on the topic of media multiplexing. &nbsp;I
                  am sending this notice to rtcweb since this is the
                  group that triggered recent work on RTP multiplexing.<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial">I also request the rtcweb chairs
                  consider giving me some time to present this work at
                  next week&#8217;s interim meeting if there is time on the
                  agenda. &nbsp;It might be useful to discuss the ideas in
                  the ID before avtcore meets in <st1:place w:st="on">
                    <st1:city w:st="on">Vancouver</st1:city></st1:place>.<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial">Richard<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">-----Original Message-----<br>
                  From: Ejzak, Richard P (Richard) <br>
                  Sent: Sunday, June 03, 2012 6:08 PM<br>
                  To: '<a class="moz-txt-link-abbreviated" href="mailto:avt@ietf.org">avt@ietf.org</a>'<br>
                  Subject: [AVTCORE] multi session
                  draft-ejzak-avtcore-rtp-subsessions-00.txt<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><a moz-do-not-send="true"
                    href="http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt">http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt</a><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">In response to the chairs' request for
                  additional input on the multi session issue, I have
                  submitted this draft for your consideration.&nbsp; There
                  are superficial similarities with an expired draft
                  from last year in
                  <a class="moz-txt-link-freetext" href="http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt">http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt</a>,
                  but my draft has a different take on the use of SSRC
                  as the basis for multi session multiplexing that I
                  think is superior and worthy of consideration as a
                  potential replacement for BUNDLE and/or SHIM.&nbsp; Please
                  comment.<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Richard<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">-----Original Message-----<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">From: <a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:internet-drafts@ietf.org">mailto:internet-drafts@ietf.org</a>]
                  <o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Sent: Sunday, June 03, 2012 5:16 PM<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">To: Ejzak, Richard P (Richard)<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Subject: New Version Notification for
                  draft-ejzak-avtcore-rtp-subsessions-00.txt<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">A new version of I-D,
                  draft-ejzak-avtcore-rtp-subsessions-00.txt has been
                  successfully submitted by Richard Ejzak and posted to
                  the IETF repository.<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Filename:&nbsp;&nbsp;
                  draft-ejzak-avtcore-rtp-subsessions<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Revision:&nbsp;&nbsp; 00<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Media multiplexing with
                  Real-time Transport Protocol (RTP) subsessions<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Creation date:&nbsp;&nbsp;&nbsp; 2012-06-04<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Individual Submission<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Number of pages: 9<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">Abstract:<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; This document describes a means of
                  multiplexing RTP streams having<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; different media types within a single
                  transport connection and how to<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; represent this multiplexing option in SDP.&nbsp;
                  This mechanism is an<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; alternative to the BUNDLE and SHIM
                  proposals currently under active<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; consideration in AVTCORE.&nbsp; Instead of
                  adding an extra multiplexing<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; header as in SHIM to allow multiple RTP
                  sessions within a single<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; transport connection, or using the payload
                  type field to separate<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; different media streams within a single RTP
                  session, this document<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; describes how to partition the existing
                  SSRC space to create RTP<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; subsessions from a single RTP session.&nbsp; A
                  filter can be used to<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; identify each RTP subsession for different
                  QoS handling as necessary.<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; RTP subsessions can be treated like RTP
                  sessions with a few<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; restrictions.&nbsp; In particular, SSRC mapping
                  may be needed when<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; forwarding RTP streams into an RTP
                  subsession to avoid SSRC<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; conflicts, but there are few use cases in
                  which this limitation is a<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&nbsp;&nbsp; concern and RTP subsessions can be disabled
                  if necessary.<o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">&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;&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;
                  <o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoPlainText"><font face="Courier New" size="2"><span
                  style="font-size:
                  10.0pt">The IETF Secretariat<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font face="Arial" size="2"><span
                  style="font-size:10.0pt;
                  font-family:Arial">&nbsp;<o:p></o:p></span></font></p>
          </div>
          <br>
          <fieldset class="mimeAttachmentHeader"></fieldset>
          <br>
          <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
        </o:smarttagtype></o:smarttagtype></blockquote>
    <br>
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb]  draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Harald,
While ideally we should be able to maintain the entire SSRC space for use b=
y an RTP session, the current concept of BUNDLE breaks the model by mixing =
what are otherwise different RTP sessions together.  The scope of an RTP se=
ssion is beyond just the transport connection on which bundling is occurrin=
g (see the RTP session topology diagrams in RFC 3550).  Even though RTP pac=
kets with different payload types could potentially reuse the same SSRC val=
ue, SSRC values cannot be independently assigned per m line with the curren=
t BUNDLE scheme since RTCP packets can be identified only using SSRC values=
 (payload type is not a field in RTCP packets).  So the bundled m lines hav=
e to share a single SSRC space, thus increasing the likelihood of SSRC coll=
ision and preventing use of the same SSRC value across m lines and ports (a=
s some applications require).  To avoid these problems, translators impleme=
nting the current BUNDLE will generally need to do SSRC mapping anyway, so =
BUNDLE cannot take full advantage of the "randomness and flatness of the SS=
RC space".

I have read your draft and should have referenced it in my paper.  I agree =
that different media types should be able to share the same transport conne=
ction.  But I also see a need to be able to apply differential treatment of=
 media types within the network.

You disagree with the need to separate packets associated with different me=
dia lines for differential treatment in the network.  Nevertheless, this is=
sue has been raised in RFC 3550 and numerous drafts since then as a legitim=
ate issue in certain deployments.  While I agree with you that there are si=
gnificant use cases in which such differential treatment is not necessary, =
there are others in which it can be very beneficial.  It would be really un=
fortunate if there were no way to support such use cases in rtcweb.

Richard


________________________________
From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Monday, June 04, 2012 1:43 PM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE

Richard,
having scanned your draft:

- I still fundamentally think that the partition of media types into separa=
te RTP sessions was a design mistake that we should seek to rectify, not pr=
omulgate, and additional complexity like what you propose for perpetuating =
the use of "separated" sessions is not helpful. See draft-alvestrand-rtp-se=
ss-neutral for details.

- When we have parts of the ecosystem that have code that is written on the=
 assumption that the SSRC space is a 32-bit flat space, I think the partiti=
oning of that space into subspaces as you propose is likely to lead to "int=
eresting" bugs. We should reinforce the randomness and flatness of the SSRC=
 space, not partition it.

- You say "The biggest problem with BUNDLE is that it is
   difficult to partition the RTP streams associated with different
   media lines since this requires filtering on sets of payload type
   values.  RTP subsessions is superior for this purpose since it is
   only necessary to screen for a single value in the first 16 bits".
This assumes that the partitioning associated with different media lines is=
 a good thing.
I claim that it is not.

If there are no bigger problems than that with BUNDLE, I would prefer to st=
ick with that approach.





On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote:
Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard


-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org<mailto:avt@ietf.org>'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt



http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt



In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.



Richard



-----Original Message-----

From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org]

Sent: Sunday, June 03, 2012 5:16 PM

To: Ejzak, Richard P (Richard)

Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt



A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.



Filename:   draft-ejzak-avtcore-rtp-subsessions

Revision:   00

Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions

Creation date:    2012-06-04

WG ID:            Individual Submission

Number of pages: 9



Abstract:

   This document describes a means of multiplexing RTP streams having

   different media types within a single transport connection and how to

   represent this multiplexing option in SDP.  This mechanism is an

   alternative to the BUNDLE and SHIM proposals currently under active

   consideration in AVTCORE.  Instead of adding an extra multiplexing

   header as in SHIM to allow multiple RTP sessions within a single

   transport connection, or using the payload type field to separate

   different media streams within a single RTP session, this document

   describes how to partition the existing SSRC space to create RTP

   subsessions from a single RTP session.  A filter can be used to

   identify each RTP subsession for different QoS handling as necessary.

   RTP subsessions can be treated like RTP sessions with a few

   restrictions.  In particular, SSRC mapping may be needed when

   forwarding RTP streams into an RTP subsession to avoid SSRC

   conflicts, but there are few use cases in which this limitation is a

   concern and RTP subsessions can be disabled if necessary.









The IETF Secretariat







_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb


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<body bgcolor=3D"white" lang=3D"EN-US" link=3D"blue" vlink=3D"#606420">
<div class=3D"Section1">
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Harald,<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">While ideally we should be able to mai=
ntain the entire SSRC space for use by an RTP session, the current concept =
of BUNDLE breaks the
 model by mixing what are otherwise different RTP sessions together. &nbsp;=
The scope of an RTP session is beyond just the transport connection on whic=
h bundling is occurring (see the RTP session topology diagrams in RFC 3550)=
. &nbsp;Even though RTP packets with different
 payload types could potentially reuse the same SSRC value, SSRC values can=
not be independently assigned per m line with the current BUNDLE scheme sin=
ce RTCP packets can be identified only using SSRC values (payload type is n=
ot a field in RTCP packets).&nbsp; So
 the bundled m lines have to share a single SSRC space, thus increasing the=
 likelihood of SSRC collision and preventing use of the same SSRC value acr=
oss m lines and ports (as some applications require).&nbsp; To avoid these =
problems, translators implementing the
 current BUNDLE will generally need to do SSRC mapping anyway, so BUNDLE ca=
nnot take full advantage of the &#8220;randomness and flatness of the SSRC =
space&#8221;.<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">I have read your draft and should have=
 referenced it in my paper.&nbsp; I agree that different media types should=
 be able to share the same
 transport connection. &nbsp;But I also see a need to be able to apply diff=
erential treatment of media types within the network.<o:p></o:p></span></fo=
nt></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">You disagree with the need to separate=
 packets associated with different media lines for differential treatment i=
n the network. &nbsp;Nevertheless,
 this issue has been raised in RFC 3550 and numerous drafts since then as a=
 legitimate issue in certain deployments. &nbsp;While I agree with you that=
 there are significant use cases in which such differential treatment is no=
t necessary, there are others in which
 it can be very beneficial. &nbsp;It would be really unfortunate if there w=
ere no way to support such use cases in rtcweb.<o:p></o:p></span></font></p=
>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy">Richard<o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"navy" face=3D"Arial"><span=
 style=3D"font-size:
10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center"><font=
 size=3D"3" color=3D"black" face=3D"Times New Roman"><span style=3D"font-si=
ze:12.0pt;font-family:
&quot;Times New Roman&quot;;color:windowtext">
<hr size=3D"2" width=3D"100%" align=3D"center" tabindex=3D"-1">
</span></font></div>
<p class=3D"MsoNormal"><b><font size=3D"2" color=3D"black" face=3D"Tahoma">=
<span style=3D"font-size:10.0pt;font-family:Tahoma;color:windowtext;font-we=
ight:bold">From:</span></font></b><font size=3D"2" color=3D"black" face=3D"=
Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma;
color:windowtext">
 Harald Alvestrand [mailto:harald@alvestrand.no] <br>
<b><span style=3D"font-weight:bold">Sent:</span></b> Monday, June 04, 2012 =
1:43 PM<br>
<b><span style=3D"font-weight:bold">To:</span></b> Ejzak, Richard P (Richar=
d)<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> rtcweb@ietf.org<br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [rtcweb] draft =
on media multiplexing submitted to AVTCORE</span></font><font size=3D"3" co=
lor=3D"black" face=3D"Times New Roman"><span style=3D"font-size:12.0pt;font=
-family:&quot;Times New Roman&quot;;
color:windowtext"><o:p></o:p></span></font></p>
</div>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"FuturaA Bk =
BT"><span style=3D"font-size:11.0pt"><o:p>&nbsp;</o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"FuturaA Bk =
BT"><span style=3D"font-size:11.0pt">Richard,<br>
having scanned your draft:<br>
<br>
- I still fundamentally think that the partition of media types into separa=
te RTP sessions was a design mistake that we should seek to rectify, not pr=
omulgate, and additional complexity like what you propose for perpetuating =
the use of &quot;separated&quot; sessions
 is not helpful. See draft-alvestrand-rtp-sess-neutral for details.<br>
<br>
- When we have parts of the ecosystem that have code that is written on the=
 assumption that the SSRC space is a 32-bit flat space, I think the partiti=
oning of that space into subspaces as you propose is likely to lead to &quo=
t;interesting&quot; bugs. We should reinforce
 the randomness and flatness of the SSRC space, not partition it.<br>
<br>
- You say &quot;The biggest problem with BUNDLE is that it is<br>
&nbsp;&nbsp; difficult to partition the RTP streams associated with differe=
nt<br>
&nbsp;&nbsp; media lines since this requires filtering on sets of payload t=
ype<br>
&nbsp;&nbsp; values.&nbsp; RTP subsessions is superior for this purpose sin=
ce it is<br>
&nbsp;&nbsp; only necessary to screen for a single value in the first 16 bi=
ts&quot;.<br>
This assumes that the partitioning associated with different media lines is=
 a good thing.<br>
I claim that it is not.<br>
<br>
If there are no bigger problems than that with BUNDLE, I would prefer to st=
ick with that approach.<br>
<br>
<br>
<br>
<br>
<br>
On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote: </span></font><fo=
nt face=3D"Times New Roman"><span style=3D"font-family:&quot;Times New Roma=
n&quot;"><o:p></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:smarttagtype namespaceuri=3D"urn:schemas-micr=
osoft-com:office:smarttags" name=3D"City"><u1:smarttagtype namespaceuri=3D"=
urn:schemas-microsoft-com:office:smarttags" name=3D"place"><!--[if gte mso =
9]><xml>
   <u1:shapedefaults u2:ext=3D"edit" spidmax=3D"1026"/>
</xml><![endif]--><!--[if gte mso 9]><xml>
   <u1:shapelayout u3:ext=3D"edit">
    <u1:idmap u3:ext=3D"edit" data=3D"1"/>
   </u1:shapelayout>
</xml><![endif]-->Please
 see the mail attached below that I just sent to avtcore announcing a perso=
nal ID I just submitted on the topic of media multiplexing. &nbsp;I am send=
ing this notice to rtcweb since this is the group that triggered recent wor=
k on RTP multiplexing.<u1:p></u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial">I also request the rtcweb chairs consider giving =
me some time to present this work at next week&#8217;s interim meeting if t=
here is time on the agenda.
 &nbsp;It might be useful to discuss the ideas in the ID before avtcore mee=
ts in <st1:place u4:st=3D"on">
<st1:city u4:st=3D"on"><st1:City w:st=3D"on"><st1:place w:st=3D"on">Vancouv=
er</st1:city></st1:place></st1:place></st1:City>.<u1:p></u1:p></span></font=
><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial">Richard<u1:p></u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">-----Original Message-----<br>
From: Ejzak, Richard P (Richard) <br>
Sent: Sunday, June 03, 2012 6:08 PM<br>
To: '<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>'<br>
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt=
<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><a href=3D"http://www.ietf.org/id/dra=
ft-ejzak-avtcore-rtp-subsessions-00.txt" moz-do-not-send=3D"true">http://ww=
w.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt</a><u1:p></u1:p><o=
:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">In response to the chairs' request fo=
r additional input on the multi session issue, I have submitted this draft =
for your consideration.&nbsp; There are superficial
 similarities with an expired draft from last year in <a href=3D"http://too=
ls.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt">
http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt</a>, but my d=
raft has a different take on the use of SSRC as the basis for multi session=
 multiplexing that I think is superior and worthy of consideration as a pot=
ential replacement for BUNDLE and/or
 SHIM.&nbsp; Please comment.<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Richard<u1:p></u1:p><o:p></o:p></span=
></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">-----Original Message-----<u1:p></u1:=
p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">From:
<a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> [<=
a href=3D"mailto:internet-drafts@ietf.org">mailto:internet-drafts@ietf.org<=
/a>]
<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Sent: Sunday, June 03, 2012 5:16 PM<u=
1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">To: Ejzak, Richard P (Richard)<u1:p><=
/u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Subject: New Version Notification for=
 draft-ejzak-avtcore-rtp-subsessions-00.txt<u1:p></u1:p><o:p></o:p></span><=
/font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">A new version of I-D, draft-ejzak-avt=
core-rtp-subsessions-00.txt has been successfully submitted by Richard Ejza=
k and posted to the IETF repository.<u1:p></u1:p><o:p></o:p></span></font><=
/p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Filename:&nbsp;&nbsp; draft-ejzak-avt=
core-rtp-subsessions<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Revision:&nbsp;&nbsp; 00<u1:p></u1:p>=
<o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Media multiplexing with Real-time Trans=
port Protocol (RTP) subsessions<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Creation date:&nbsp;&nbsp;&nbsp; 2012=
-06-04<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Individual Submission<u1:p></u1:p><o:p>=
</o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Number of pages: 9<u1:p></u1:p><o:p><=
/o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">Abstract:<u1:p></u1:p><o:p></o:p></sp=
an></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; This document describes =
a means of multiplexing RTP streams having<u1:p></u1:p><o:p></o:p></span></=
font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; different media types wi=
thin a single transport connection and how to<u1:p></u1:p><o:p></o:p></span=
></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; represent this multiplex=
ing option in SDP.&nbsp; This mechanism is an<u1:p></u1:p><o:p></o:p></span=
></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; alternative to the BUNDL=
E and SHIM proposals currently under active<u1:p></u1:p><o:p></o:p></span><=
/font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; consideration in AVTCORE=
.&nbsp; Instead of adding an extra multiplexing<u1:p></u1:p><o:p></o:p></sp=
an></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; header as in SHIM to all=
ow multiple RTP sessions within a single<u1:p></u1:p><o:p></o:p></span></fo=
nt></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; transport connection, or=
 using the payload type field to separate<u1:p></u1:p><o:p></o:p></span></f=
ont></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; different media streams =
within a single RTP session, this document<u1:p></u1:p><o:p></o:p></span></=
font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; describes how to partiti=
on the existing SSRC space to create RTP<u1:p></u1:p><o:p></o:p></span></fo=
nt></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; subsessions from a singl=
e RTP session.&nbsp; A filter can be used to<u1:p></u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; identify each RTP subses=
sion for different QoS handling as necessary.<u1:p></u1:p><o:p></o:p></span=
></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; RTP subsessions can be t=
reated like RTP sessions with a few<u1:p></u1:p><o:p></o:p></span></font></=
p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; restrictions.&nbsp; In p=
articular, SSRC mapping may be needed when<u1:p></u1:p><o:p></o:p></span></=
font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; forwarding RTP streams i=
nto an RTP subsession to avoid SSRC<u1:p></u1:p><o:p></o:p></span></font></=
p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; conflicts, but there are=
 few use cases in which this limitation is a<u1:p></u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp; concern and RTP subsessi=
ons can be disabled if necessary.<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
<u1:p></u1:p><o:p></o:p></span></font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span>=
</font></p>
<p class=3D"MsoPlainText"><font size=3D"2" color=3D"black" face=3D"Courier =
New"><span style=3D"font-size:10.0pt">The IETF Secretariat<u1:p></u1:p><o:p=
></o:p></span></font></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"2" color=3D"black" face=3D"Arial"><spa=
n style=3D"font-size:
10.0pt;font-family:Arial">&nbsp;<u1:p></u1:p></span></font><o:p></o:p></p>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt;font-family:&quot;Times New Roman&quo=
t;"><br>
<br>
<br>
</p>
<fieldset class=3D"mimeAttachmentHeader"></fieldset><o:p></o:p></span></fon=
t>
<p></p>
<pre><font size=3D"2" color=3D"black" face=3D"Courier New"><span style=3D"f=
ont-size:10.0pt">_______________________________________________<o:p></o:p>=
</span></font></pre>
<pre><font size=3D"2" color=3D"black" face=3D"Courier New"><span style=3D"f=
ont-size:10.0pt">rtcweb mailing list<o:p></o:p></span></font></pre>
<pre><font size=3D"2" color=3D"black" face=3D"Courier New"><span style=3D"f=
ont-size:10.0pt"><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p=
></o:p></span></font></pre>
<pre><font size=3D"2" color=3D"black" face=3D"Courier New"><span style=3D"f=
ont-size:10.0pt"><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">h=
ttps://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></font></p=
re>
<p class=3D"MsoNormal"><font size=3D"3" color=3D"black" face=3D"Times New R=
oman"><span style=3D"font-size:12.0pt;font-family:&quot;Times New Roman&quo=
t;"><o:p>&nbsp;</o:p></span></font></p>
</u1:smarttagtype></u1:smarttagtype></div>
</body>
</html>

--_000_03FBA798AC24E3498B74F47FD082A92FC55AUS70UWXCHMBA05zamal_--

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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-rtp-usage-03.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Web Real-Time Communication (WebRTC): Media Transport an=
d Use of RTP
	Author(s)       : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-03.txt
	Pages           : 61
	Date            : 2012-06-04

   The Web Real-Time Communication (WebRTC) framework provides support
   for direct interactive rich communication using audio, video, text,
   collaboration, games, etc. between two peers' web-browsers.  This
   memo describes the media transport aspects of the WebRTC framework.
   It specifies how the Real-time Transport Protocol (RTP) is used in
   the WebRTC context, and gives requirements for which RTP features,
   profiles, and extensions need to be supported.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-rtp-usage-03.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-rtp-usage-03.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/


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On 06/04/2012 10:03 PM, Ejzak, Richard P (Richard) wrote:
>
> Harald,
>
> While ideally we should be able to maintain the entire SSRC space for 
> use by an RTP session, the current concept of BUNDLE breaks the model 
> by mixing what are otherwise different RTP sessions together.  The 
> scope of an RTP session is beyond just the transport connection on 
> which bundling is occurring (see the RTP session topology diagrams in 
> RFC 3550).
>
Yes, all parts of the RTP session need to agree on where the limits of 
the RTP session are. BUNDLE (the way it's currently written; Christer 
and I disagree on whether there is value in extending it to multiple 
sessions on the same address pairs) is about modifying the content of a 
single RTP session, it does not "mix different RTP sessions".
>
>  Even though RTP packets with different payload types could 
> potentially reuse the same SSRC value, SSRC values cannot be 
> independently assigned per m line with the current BUNDLE scheme since 
> RTCP packets can be identified only using SSRC values (payload type is 
> not a field in RTCP packets).  So the bundled m lines have to share a 
> single SSRC space, thus increasing the likelihood of SSRC collision
>
Yes. At 302 SSRCs in the same RTP session the chance of a collision is 
50%. Applications that don't handle SSRC collisions are broken.
>
> and preventing use of the same SSRC value across m lines and ports (as 
> some applications require).
>
These applications have a design mistake, and should be recommended 
against. While they are still in use, they force the use of multiple RTP 
sessions where it's otherwise superfluous. Luckily there are not that 
many of them.
>
>   To avoid these problems, translators implementing the current BUNDLE 
> will generally need to do SSRC mapping anyway, so BUNDLE cannot take 
> full advantage of the "randomness and flatness of the SSRC space".
>
RTP media translators in an RTP session where BUNDLE is used for setup 
need to set up their RTP sessions in such a way that either BUNDLE is 
used on all legs, or BUNDLE is used on no legs. This is because RTP 
media translators don't terminate RTP sessions.

Gateways that terminate RTP sessions have to remap SSRCs anyway. No news 
here.
>
> I have read your draft and should have referenced it in my paper.  I 
> agree that different media types should be able to share the same 
> transport connection.  But I also see a need to be able to apply 
> differential treatment of media types within the network.
>
Yes, some applications need that.
  In that case, the simplest way (allowing per-flow prioritization) is 
to use different RTP sessions on different transport addresses.
The next simplest way (using DiffServ) is to apply different DSCP 
markings on a per-packet basis. Both techniques are well established.

Breaking up the SSRC space and applying differential treatment based on 
that requires rewriting the traffic management components. That's new 
code that needs to be deployed. I don't see the need.
>
> You disagree with the need to separate packets associated with 
> different media lines for differential treatment in the network.
>
No, I don't disagree at all with that need, and have said so many times. 
I disagree that this is a need that is present *all the time*, and 
disagree that this need is so omnipresent that we need to shape our 
architectures around it.
>
>  Nevertheless, this issue has been raised in RFC 3550 and numerous 
> drafts since then as a legitimate issue in certain deployments.  While 
> I agree with you that there are significant use cases in which such 
> differential treatment is not necessary, there are others in which it 
> can be very beneficial.  It would be really unfortunate if there were 
> no way to support such use cases in rtcweb.
>
The point I was trying to make in rtp-sess-neutral is that the need for 
such differential treatment is orthogonal to the media types, and 
applications need to assign media streams to RTP sessions as the 
*applications*, not the *architecture*, think is adequate. That's really 
the focus of BUNDLE.

But really, this discussion belongs on AVTCORE.
>
> Richard
>
> ------------------------------------------------------------------------
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Monday, June 04, 2012 1:43 PM
> *To:* Ejzak, Richard P (Richard)
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
>
> Richard,
> having scanned your draft:
>
> - I still fundamentally think that the partition of media types into 
> separate RTP sessions was a design mistake that we should seek to 
> rectify, not promulgate, and additional complexity like what you 
> propose for perpetuating the use of "separated" sessions is not 
> helpful. See draft-alvestrand-rtp-sess-neutral for details.
>
> - When we have parts of the ecosystem that have code that is written 
> on the assumption that the SSRC space is a 32-bit flat space, I think 
> the partitioning of that space into subspaces as you propose is likely 
> to lead to "interesting" bugs. We should reinforce the randomness and 
> flatness of the SSRC space, not partition it.
>
> - You say "The biggest problem with BUNDLE is that it is
>    difficult to partition the RTP streams associated with different
>    media lines since this requires filtering on sets of payload type
>    values.  RTP subsessions is superior for this purpose since it is
>    only necessary to screen for a single value in the first 16 bits".
> This assumes that the partitioning associated with different media 
> lines is a good thing.
> I claim that it is not.
>
> If there are no bigger problems than that with BUNDLE, I would prefer 
> to stick with that approach.
>
>
>
>
>
> On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote:
>
> Please see the mail attached below that I just sent to avtcore 
> announcing a personal ID I just submitted on the topic of media 
> multiplexing.  I am sending this notice to rtcweb since this is the 
> group that triggered recent work on RTP multiplexing.
>
> I also request the rtcweb chairs consider giving me some time to 
> present this work at next week's interim meeting if there is time on 
> the agenda.  It might be useful to discuss the ideas in the ID before 
> avtcore meets in Vancouver.
>
> Richard
>
> -----Original Message-----
> From: Ejzak, Richard P (Richard)
> Sent: Sunday, June 03, 2012 6:08 PM
> To: 'avt@ietf.org <mailto:avt@ietf.org>'
> Subject: [AVTCORE] multi session 
> draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> In response to the chairs' request for additional input on the multi 
> session issue, I have submitted this draft for your consideration.  
> There are superficial similarities with an expired draft from last 
> year in http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, 
> but my draft has a different take on the use of SSRC as the basis for 
> multi session multiplexing that I think is superior and worthy of 
> consideration as a potential replacement for BUNDLE and/or SHIM.  
> Please comment.
>
> Richard
>
> -----Original Message-----
>
> From: internet-drafts@ietf.org <mailto:internet-drafts@ietf.org> 
> [mailto:internet-drafts@ietf.org]
>
> Sent: Sunday, June 03, 2012 5:16 PM
>
> To: Ejzak, Richard P (Richard)
>
> Subject: New Version Notification for 
> draft-ejzak-avtcore-rtp-subsessions-00.txt
>
> A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has 
> been successfully submitted by Richard Ejzak and posted to the IETF 
> repository.
>
> Filename:   draft-ejzak-avtcore-rtp-subsessions
>
> Revision:   00
>
> Title:            Media multiplexing with Real-time Transport Protocol 
> (RTP) subsessions
>
> Creation date:    2012-06-04
>
> WG ID:            Individual Submission
>
> Number of pages: 9
>
> Abstract:
>
>    This document describes a means of multiplexing RTP streams having
>
>    different media types within a single transport connection and how to
>
>    represent this multiplexing option in SDP.  This mechanism is an
>
>    alternative to the BUNDLE and SHIM proposals currently under active
>
>    consideration in AVTCORE.  Instead of adding an extra multiplexing
>
>    header as in SHIM to allow multiple RTP sessions within a single
>
>    transport connection, or using the payload type field to separate
>
>    different media streams within a single RTP session, this document
>
>    describes how to partition the existing SSRC space to create RTP
>
>    subsessions from a single RTP session.  A filter can be used to
>
>    identify each RTP subsession for different QoS handling as necessary.
>
>    RTP subsessions can be treated like RTP sessions with a few
>
>    restrictions.  In particular, SSRC mapping may be needed when
>
>    forwarding RTP streams into an RTP subsession to avoid SSRC
>
>    conflicts, but there are few use cases in which this limitation is a
>
>    concern and RTP subsessions can be disabled if necessary.
>
> The IETF Secretariat
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org  <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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    On 06/04/2012 10:03 PM, Ejzak, Richard P (Richard) wrote:
    <blockquote
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          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy">Harald,<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;">While ideally we should be able
                  to maintain the entire SSRC space for use by an RTP
                  session, the current concept of BUNDLE breaks the
                  model by mixing what are otherwise different RTP
                  sessions together. &nbsp;The scope of an RTP session is
                  beyond just the transport connection on which bundling
                  is occurring (see the RTP session topology diagrams in
                  RFC 3550).</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    Yes, all parts of the RTP session need to agree on where the limits
    of the RTP session are. BUNDLE (the way it's currently written;
    Christer and I disagree on whether there is value in extending it to
    multiple sessions on the same address pairs) is about modifying the
    content of a single RTP session, it does not "mix different RTP
    sessions".<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;"> &nbsp;Even though RTP packets with
                  different payload types could potentially reuse the
                  same SSRC value, SSRC values cannot be independently
                  assigned per m line with the current BUNDLE scheme
                  since RTCP packets can be identified only using SSRC
                  values (payload type is not a field in RTCP packets).&nbsp;
                  So the bundled m lines have to share a single SSRC
                  space, thus increasing the likelihood of SSRC
                  collision</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    Yes. At 302 SSRCs in the same RTP session the chance of a collision
    is 50%. Applications that don't handle SSRC collisions are broken.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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        name="City"><o:smarttagtype
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          name="place">
          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;"> and preventing use of the same
                  SSRC value across m lines and ports (as some
                  applications require).</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    These applications have a design mistake, and should be recommended
    against. While they are still in use, they force the use of multiple
    RTP sessions where it's otherwise superfluous. Luckily there are not
    that many of them.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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        name="City"><o:smarttagtype
          namespaceuri="urn:schemas-microsoft-com:office:smarttags"
          name="place">
          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;">&nbsp; To avoid these problems,
                  translators implementing the current BUNDLE will
                  generally need to do SSRC mapping anyway, so BUNDLE
                  cannot take full advantage of the &#8220;randomness and
                  flatness of the SSRC space&#8221;.</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    RTP media translators in an RTP session where BUNDLE is used for
    setup need to set up their RTP sessions in such a way that either
    BUNDLE is used on all legs, or BUNDLE is used on no legs. This is
    because RTP media translators don't terminate RTP sessions.<br>
    <br>
    Gateways that terminate RTP sessions have to remap SSRCs anyway. No
    news here.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
        namespaceuri="urn:schemas-microsoft-com:office:smarttags"
        name="City"><o:smarttagtype
          namespaceuri="urn:schemas-microsoft-com:office:smarttags"
          name="place">
          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;">I have read your draft and should
                  have referenced it in my paper.&nbsp; I agree that
                  different media types should be able to share the same
                  transport connection. &nbsp;But I also see a need to be
                  able to apply differential treatment of media types
                  within the network.</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    Yes, some applications need that.<br>
    &nbsp;In that case, the simplest way (allowing per-flow prioritization)
    is to use different RTP sessions on different transport addresses.<br>
    The next simplest way (using DiffServ) is to apply different DSCP
    markings on a per-packet basis. Both techniques are well
    established.<br>
    <br>
    Breaking up the SSRC space and applying differential treatment based
    on that requires rewriting the traffic management components. That's
    new code that needs to be deployed. I don't see the need.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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        name="City"><o:smarttagtype
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          name="place">
          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;">You disagree with the need to
                  separate packets associated with different media lines
                  for differential treatment in the network. </span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    No, I don't disagree at all with that need, and have said so many
    times. I disagree that this is a need that is present *all the
    time*, and disagree that this need is so omnipresent that we need to
    shape our architectures around it.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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        name="City"><o:smarttagtype
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          name="place">
          <div class="Section1">
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size: 10pt; font-family:
                  Arial; color: navy;">&nbsp;Nevertheless, this issue has
                  been raised in RFC 3550 and numerous drafts since then
                  as a legitimate issue in certain deployments. &nbsp;While I
                  agree with you that there are significant use cases in
                  which such differential treatment is not necessary,
                  there are others in which it can be very beneficial.
                  &nbsp;It would be really unfortunate if there were no way
                  to support such use cases in rtcweb.</span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    The point I was trying to make in rtp-sess-neutral is that the need
    for such differential treatment is orthogonal to the media types,
    and applications need to assign media streams to RTP sessions as the
    *applications*, not the *architecture*, think is adequate. That's
    really the focus of BUNDLE.<br>
    <br>
    But really, this discussion belongs on AVTCORE.<br>
    <blockquote
cite="mid:03FBA798AC24E3498B74F47FD082A92FC55A@US70UWXCHMBA05.zam.alcatel-lucent.com"
      type="cite"><o:smarttagtype
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        name="City"><o:smarttagtype
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          name="place">
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            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
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            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy">Richard<o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font color="navy" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial;color:navy"><o:p>&nbsp;</o:p></span></font></p>
            <div>
              <div class="MsoNormal" style="text-align:center"
                align="center"><font color="black" face="Times New
                  Roman" size="3"><span
                    style="font-size:12.0pt;font-family:
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                      width="100%">
                  </span></font></div>
              <p class="MsoNormal"><b><font color="black" face="Tahoma"
                    size="2"><span
style="font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:bold">From:</span></font></b><font
                  color="black" face="Tahoma" size="2"><span
                    style="font-size:10.0pt;font-family:Tahoma;
                    color:windowtext"> Harald Alvestrand
                    [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
                    <b><span style="font-weight:bold">Sent:</span></b>
                    Monday, June 04, 2012 1:43 PM<br>
                    <b><span style="font-weight:bold">To:</span></b>
                    Ejzak, Richard P (Richard)<br>
                    <b><span style="font-weight:bold">Cc:</span></b>
                    <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    <b><span style="font-weight:bold">Subject:</span></b>
                    Re: [rtcweb] draft on media multiplexing submitted
                    to AVTCORE</span></font><font color="black"
                  face="Times New Roman" size="3"><span
                    style="font-size:12.0pt;font-family:&quot;Times New
                    Roman&quot;;
                    color:windowtext"><o:p></o:p></span></font></p>
            </div>
            <p class="MsoNormal"><font color="black" face="FuturaA Bk
                BT" size="2"><span style="font-size:11.0pt"><o:p>&nbsp;</o:p></span></font></p>
            <p class="MsoNormal"><font color="black" face="FuturaA Bk
                BT" size="2"><span style="font-size:11.0pt">Richard,<br>
                  having scanned your draft:<br>
                  <br>
                  - I still fundamentally think that the partition of
                  media types into separate RTP sessions was a design
                  mistake that we should seek to rectify, not
                  promulgate, and additional complexity like what you
                  propose for perpetuating the use of "separated"
                  sessions is not helpful. See
                  draft-alvestrand-rtp-sess-neutral for details.<br>
                  <br>
                  - When we have parts of the ecosystem that have code
                  that is written on the assumption that the SSRC space
                  is a 32-bit flat space, I think the partitioning of
                  that space into subspaces as you propose is likely to
                  lead to "interesting" bugs. We should reinforce the
                  randomness and flatness of the SSRC space, not
                  partition it.<br>
                  <br>
                  - You say "The biggest problem with BUNDLE is that it
                  is<br>
                  &nbsp;&nbsp; difficult to partition the RTP streams associated
                  with different<br>
                  &nbsp;&nbsp; media lines since this requires filtering on sets
                  of payload type<br>
                  &nbsp;&nbsp; values.&nbsp; RTP subsessions is superior for this
                  purpose since it is<br>
                  &nbsp;&nbsp; only necessary to screen for a single value in the
                  first 16 bits".<br>
                  This assumes that the partitioning associated with
                  different media lines is a good thing.<br>
                  I claim that it is not.<br>
                  <br>
                  If there are no bigger problems than that with BUNDLE,
                  I would prefer to stick with that approach.<br>
                  <br>
                  <br>
                  <br>
                  <br>
                  <br>
                  On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard)
                  wrote: </span></font><font face="Times New Roman"><span
                  style="font-family:&quot;Times New Roman&quot;"><o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:smarttagtype
                    namespaceuri="urn:schemas-microsoft-com:office:smarttags"
                    name="City"><u1:smarttagtype
                      namespaceuri="urn:schemas-microsoft-com:office:smarttags"
                      name="place"><!--[if gte mso 9]><xml>
   <u1:shapedefaults u2:ext="edit" spidmax="1026"/>
</xml><![endif]--><!--[if gte mso 9]><xml>
   <u1:shapelayout u3:ext="edit">
    <u1:idmap u3:ext="edit" data="1"/>
   </u1:shapelayout>
</xml><![endif]-->Please see the mail attached below that I just sent to
                      avtcore announcing a personal ID I just submitted
                      on the topic of media multiplexing. &nbsp;I am sending
                      this notice to rtcweb since this is the group that
                      triggered recent work on RTP multiplexing.<u1:p></u1:p></u1:smarttagtype></u1:smarttagtype></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial">I also request the rtcweb
                  chairs consider giving me some time to present this
                  work at next week&#8217;s interim meeting if there is time
                  on the agenda. &nbsp;It might be useful to discuss the
                  ideas in the ID before avtcore meets in <st1:place
                    u4:st="on">
                    <st1:city u4:st="on"><st1:city w:st="on"><st1:place
                          w:st="on">Vancouver</st1:place></st1:city></st1:city></st1:place>.<u1:p></u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial">Richard<u1:p></u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">-----Original
                  Message-----<br>
                  From: Ejzak, Richard P (Richard) <br>
                  Sent: Sunday, June 03, 2012 6:08 PM<br>
                  To: '<a moz-do-not-send="true"
                    href="mailto:avt@ietf.org">avt@ietf.org</a>'<br>
                  Subject: [AVTCORE] multi session
                  draft-ejzak-avtcore-rtp-subsessions-00.txt<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><a
                    href="http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt"
                    moz-do-not-send="true">http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt</a><u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">In response
                  to the chairs' request for additional input on the
                  multi session issue, I have submitted this draft for
                  your consideration.&nbsp; There are superficial
                  similarities with an expired draft from last year in <a
                    moz-do-not-send="true"
                    href="http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt">
http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt</a>, but
                  my draft has a different take on the use of SSRC as
                  the basis for multi session multiplexing that I think
                  is superior and worthy of consideration as a potential
                  replacement for BUNDLE and/or SHIM.&nbsp; Please comment.<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Richard<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">-----Original
                  Message-----<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">From:
                  <a moz-do-not-send="true"
                    href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>
                  [<a moz-do-not-send="true"
                    href="mailto:internet-drafts@ietf.org">mailto:internet-drafts@ietf.org</a>]
                  <u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Sent:
                  Sunday, June 03, 2012 5:16 PM<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">To: Ejzak,
                  Richard P (Richard)<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Subject:
                  New Version Notification for
                  draft-ejzak-avtcore-rtp-subsessions-00.txt<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">A new
                  version of I-D,
                  draft-ejzak-avtcore-rtp-subsessions-00.txt has been
                  successfully submitted by Richard Ejzak and posted to
                  the IETF repository.<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Filename:&nbsp;&nbsp;
                  draft-ejzak-avtcore-rtp-subsessions<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Revision:&nbsp;&nbsp;
                  00<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Title:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                  Media multiplexing with Real-time Transport Protocol
                  (RTP) subsessions<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Creation
                  date:&nbsp;&nbsp;&nbsp; 2012-06-04<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">WG
                  ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Individual Submission<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Number of
                  pages: 9<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">Abstract:<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp; This
                  document describes a means of multiplexing RTP streams
                  having<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  different media types within a single transport
                  connection and how to<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  represent this multiplexing option in SDP.&nbsp; This
                  mechanism is an<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  alternative to the BUNDLE and SHIM proposals currently
                  under active<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  consideration in AVTCORE.&nbsp; Instead of adding an extra
                  multiplexing<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp; header
                  as in SHIM to allow multiple RTP sessions within a
                  single<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  transport connection, or using the payload type field
                  to separate<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  different media streams within a single RTP session,
                  this document<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  describes how to partition the existing SSRC space to
                  create RTP<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  subsessions from a single RTP session.&nbsp; A filter can
                  be used to<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp; identify
                  each RTP subsession for different QoS handling as
                  necessary.<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp; RTP
                  subsessions can be treated like RTP sessions with a
                  few<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  restrictions.&nbsp; In particular, SSRC mapping may be
                  needed when<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  forwarding RTP streams into an RTP subsession to avoid
                  SSRC<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp;
                  conflicts, but there are few use cases in which this
                  limitation is a<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&nbsp;&nbsp; concern
                  and RTP subsessions can be disabled if necessary.<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">&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;&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;
                  <u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt"><u1:p>&nbsp;</u1:p><o:p></o:p></span></font></p>
            <p class="MsoPlainText"><font color="black" face="Courier
                New" size="2"><span style="font-size:10.0pt">The IETF
                  Secretariat<u1:p></u1:p><o:p></o:p></span></font></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial"><u1:p>&nbsp;</u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Arial"
                size="2"><span style="font-size:
                  10.0pt;font-family:Arial">&nbsp;<u1:p></u1:p></span></font><o:p></o:p></p>
            <p class="MsoNormal"><font color="black" face="Times New
                Roman" size="3"><span
                  style="font-size:12.0pt;font-family:&quot;Times New
                  Roman&quot;"><br>
                  <br>
                  <br>
                </span></font></p>
            <font color="black" face="Times New Roman" size="3">
              <fieldset class="mimeAttachmentHeader"></fieldset>
              <o:p></o:p></font>
            <pre><font color="black" face="Courier New" size="2"><span style="font-size:10.0pt">_______________________________________________<o:p></o:p></span></font></pre>
            <pre><font color="black" face="Courier New" size="2"><span style="font-size:10.0pt">rtcweb mailing list<o:p></o:p></span></font></pre>
            <pre><font color="black" face="Courier New" size="2"><span style="font-size:10.0pt"><a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></span></font></pre>
            <pre><font color="black" face="Courier New" size="2"><span style="font-size:10.0pt"><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></font></pre>
            <p class="MsoNormal"><font color="black" face="Times New
                Roman" size="3"><span
                  style="font-size:12.0pt;font-family:&quot;Times New
                  Roman&quot;"><o:p>&nbsp;</o:p></span></font></p>
          </div>
        </o:smarttagtype></o:smarttagtype></blockquote>
    <br>
  </body>
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
Date: Tue, 5 Jun 2012 09:11:53 +0200
Thread-Topic: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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Hi,

Harald,
While ideally we should be able to maintain the entire SSRC space for use b=
y an RTP session, the current concept of BUNDLE breaks the model by mixing =
what are otherwise different RTP sessions together.  The scope of an RTP se=
ssion is beyond just the transport connection on which bundling is occurrin=
g (see the RTP session topology diagrams in RFC 3550).
Yes, all parts of the RTP session need to agree on where the limits of the =
RTP session are. BUNDLE (the way it's currently written; Christer and I dis=
agree on whether there is value in extending it to multiple sessions on the=
 same address pairs) is about modifying the content of a single RTP session=
, it does not "mix different RTP sessions".

I am not sure whether I disagree with you. Yes, the BUNDLE mechanism can be=
 extended (which I DO think is good) to be used also for multiple sessions,=
 but I'll leave it to the RTP experts to discuss/decide what (if any) value=
 there is in using multiple sessions :)

And, I even strongly agree with you that the RTP aspects of multiplexing sh=
ould be discussed in AVTCORE :)

Regards,

Christer



 Even though RTP packets with different payload types could potentially reu=
se the same SSRC value, SSRC values cannot be independently assigned per m =
line with the current BUNDLE scheme since RTCP packets can be identified on=
ly using SSRC values (payload type is not a field in RTCP packets).  So the=
 bundled m lines have to share a single SSRC space, thus increasing the lik=
elihood of SSRC collision
Yes. At 302 SSRCs in the same RTP session the chance of a collision is 50%.=
 Applications that don't handle SSRC collisions are broken.

and preventing use of the same SSRC value across m lines and ports (as some=
 applications require).
These applications have a design mistake, and should be recommended against=
. While they are still in use, they force the use of multiple RTP sessions =
where it's otherwise superfluous. Luckily there are not that many of them.

  To avoid these problems, translators implementing the current BUNDLE will=
 generally need to do SSRC mapping anyway, so BUNDLE cannot take full advan=
tage of the "randomness and flatness of the SSRC space".
RTP media translators in an RTP session where BUNDLE is used for setup need=
 to set up their RTP sessions in such a way that either BUNDLE is used on a=
ll legs, or BUNDLE is used on no legs. This is because RTP media translator=
s don't terminate RTP sessions.

Gateways that terminate RTP sessions have to remap SSRCs anyway. No news he=
re.


I have read your draft and should have referenced it in my paper.  I agree =
that different media types should be able to share the same transport conne=
ction.  But I also see a need to be able to apply differential treatment of=
 media types within the network.
Yes, some applications need that.
 In that case, the simplest way (allowing per-flow prioritization) is to us=
e different RTP sessions on different transport addresses.
The next simplest way (using DiffServ) is to apply different DSCP markings =
on a per-packet basis. Both techniques are well established.

Breaking up the SSRC space and applying differential treatment based on tha=
t requires rewriting the traffic management components. That's new code tha=
t needs to be deployed. I don't see the need.


You disagree with the need to separate packets associated with different me=
dia lines for differential treatment in the network.
No, I don't disagree at all with that need, and have said so many times. I =
disagree that this is a need that is present *all the time*, and disagree t=
hat this need is so omnipresent that we need to shape our architectures aro=
und it.

 Nevertheless, this issue has been raised in RFC 3550 and numerous drafts s=
ince then as a legitimate issue in certain deployments.  While I agree with=
 you that there are significant use cases in which such differential treatm=
ent is not necessary, there are others in which it can be very beneficial. =
 It would be really unfortunate if there were no way to support such use ca=
ses in rtcweb.
The point I was trying to make in rtp-sess-neutral is that the need for suc=
h differential treatment is orthogonal to the media types, and applications=
 need to assign media streams to RTP sessions as the *applications*, not th=
e *architecture*, think is adequate. That's really the focus of BUNDLE.

But really, this discussion belongs on AVTCORE.


Richard


________________________________
From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: Monday, June 04, 2012 1:43 PM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE

Richard,
having scanned your draft:

- I still fundamentally think that the partition of media types into separa=
te RTP sessions was a design mistake that we should seek to rectify, not pr=
omulgate, and additional complexity like what you propose for perpetuating =
the use of "separated" sessions is not helpful. See draft-alvestrand-rtp-se=
ss-neutral for details.

- When we have parts of the ecosystem that have code that is written on the=
 assumption that the SSRC space is a 32-bit flat space, I think the partiti=
oning of that space into subspaces as you propose is likely to lead to "int=
eresting" bugs. We should reinforce the randomness and flatness of the SSRC=
 space, not partition it.

- You say "The biggest problem with BUNDLE is that it is
   difficult to partition the RTP streams associated with different
   media lines since this requires filtering on sets of payload type
   values.  RTP subsessions is superior for this purpose since it is
   only necessary to screen for a single value in the first 16 bits".
This assumes that the partitioning associated with different media lines is=
 a good thing.
I claim that it is not.

If there are no bigger problems than that with BUNDLE, I would prefer to st=
ick with that approach.





On 06/04/2012 01:09 AM, Ejzak, Richard P (Richard) wrote:
Please see the mail attached below that I just sent to avtcore announcing a=
 personal ID I just submitted on the topic of media multiplexing.  I am sen=
ding this notice to rtcweb since this is the group that triggered recent wo=
rk on RTP multiplexing.

I also request the rtcweb chairs consider giving me some time to present th=
is work at next week's interim meeting if there is time on the agenda.  It =
might be useful to discuss the ideas in the ID before avtcore meets in Vanc=
ouver.

Richard


-----Original Message-----
From: Ejzak, Richard P (Richard)
Sent: Sunday, June 03, 2012 6:08 PM
To: 'avt@ietf.org<mailto:avt@ietf.org>'
Subject: [AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt



http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-00.txt



In response to the chairs' request for additional input on the multi sessio=
n issue, I have submitted this draft for your consideration.  There are sup=
erficial similarities with an expired draft from last year in http://tools.=
ietf.org/id/draft-rosenberg-rtcweb-rtpmux-00.txt, but my draft has a differ=
ent take on the use of SSRC as the basis for multi session multiplexing tha=
t I think is superior and worthy of consideration as a potential replacemen=
t for BUNDLE and/or SHIM.  Please comment.



Richard



-----Original Message-----

From: internet-drafts@ietf.org<mailto:internet-drafts@ietf.org> [mailto:int=
ernet-drafts@ietf.org]

Sent: Sunday, June 03, 2012 5:16 PM

To: Ejzak, Richard P (Richard)

Subject: New Version Notification for draft-ejzak-avtcore-rtp-subsessions-0=
0.txt



A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has been s=
uccessfully submitted by Richard Ejzak and posted to the IETF repository.



Filename:   draft-ejzak-avtcore-rtp-subsessions

Revision:   00

Title:            Media multiplexing with Real-time Transport Protocol (RTP=
) subsessions

Creation date:    2012-06-04

WG ID:            Individual Submission

Number of pages: 9



Abstract:

   This document describes a means of multiplexing RTP streams having

   different media types within a single transport connection and how to

   represent this multiplexing option in SDP.  This mechanism is an

   alternative to the BUNDLE and SHIM proposals currently under active

   consideration in AVTCORE.  Instead of adding an extra multiplexing

   header as in SHIM to allow multiple RTP sessions within a single

   transport connection, or using the payload type field to separate

   different media streams within a single RTP session, this document

   describes how to partition the existing SSRC space to create RTP

   subsessions from a single RTP session.  A filter can be used to

   identify each RTP subsession for different QoS handling as necessary.

   RTP subsessions can be treated like RTP sessions with a few

   restrictions.  In particular, SSRC mapping may be needed when

   forwarding RTP streams into an RTP subsession to avoid SSRC

   conflicts, but there are few use cases in which this limitation is a

   concern and RTP subsessions can be disabled if necessary.









The IETF Secretariat






_______________________________________________

rtcweb mailing list

rtcweb@ietf.org<mailto:rtcweb@ietf.org>

https://www.ietf.org/mailman/listinfo/rtcweb



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</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite lang=3DEN-US=
 link=3Dblue vlink=3D"#606420"><div class=3DWordSection1><p class=3DMsoNorm=
al><span style=3D'font-family:"Calibri","sans-serif";color:red'>Hi,<o:p></o=
:p></span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-fam=
ily:"Arial","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p clas=
s=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-ser=
if";color:navy'>Harald,</span><o:p></o:p></p><p class=3DMsoNormal><span sty=
le=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>While i=
deally we should be able to maintain the entire SSRC space for use by an RT=
P session, the current concept of BUNDLE breaks the model by mixing what ar=
e otherwise different RTP sessions together. &nbsp;The scope of an RTP sess=
ion is beyond just the transport connection on which bundling is occurring =
(see the RTP session topology diagrams in RFC 3550).</span><o:p></o:p></p><=
p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'>Yes, all parts of the RTP session need to agree on where th=
e limits of the RTP session are. BUNDLE (the way it's currently written; Ch=
rister and I disagree on whether there is value in extending it to multiple=
 sessions on the same address pairs) is about modifying the content of a si=
ngle RTP session, it does not &quot;mix different RTP sessions&quot;.<br><b=
r><o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Ca=
libri","sans-serif";color:red'>I am not sure whether I disagree with you. Y=
es, the BUNDLE mechanism can be extended (which I DO think is good) to be u=
sed also for multiple sessions, but I&#8217;ll leave it to the RTP experts =
to discuss/decide what (if any) value there is in using multiple sessions :=
)<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'font-family:"Cal=
ibri","sans-serif";color:red'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNor=
mal><span style=3D'font-family:"Calibri","sans-serif";color:red'>And, I eve=
n strongly agree with you that the RTP aspects of multiplexing should be di=
scussed in AVTCORE :)<o:p></o:p></span></p><p class=3DMsoNormal><span style=
=3D'font-family:"Calibri","sans-serif";color:red'><o:p>&nbsp;</o:p></span><=
/p><p class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-serif";c=
olor:red'>Regards,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D=
'font-family:"Calibri","sans-serif";color:red'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-serif";colo=
r:red'>Christer<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'fo=
nt-family:"Calibri","sans-serif";color:red'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-family:"Calibri","sans-serif";color:#=
1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span style=3D'fon=
t-family:"Calibri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";color:navy'>&nbsp;Even though RTP packets with different payload=
 types could potentially reuse the same SSRC value, SSRC values cannot be i=
ndependently assigned per m line with the current BUNDLE scheme since RTCP =
packets can be identified only using SSRC values (payload type is not a fie=
ld in RTCP packets).&nbsp; So the bundled m lines have to share a single SS=
RC space, thus increasing the likelihood of SSRC collision</span><o:p></o:p=
></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"Time=
s New Roman","serif"'>Yes. At 302 SSRCs in the same RTP session the chance =
of a collision is 50%. Applications that don't handle SSRC collisions are b=
roken.<br><br><o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'fon=
t-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>and preventing u=
se of the same SSRC value across m lines and ports (as some applications re=
quire).</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:=
12.0pt;font-family:"Times New Roman","serif"'>These applications have a des=
ign mistake, and should be recommended against. While they are still in use=
, they force the use of multiple RTP sessions where it's otherwise superflu=
ous. Luckily there are not that many of them.<br><br><o:p></o:p></span></p>=
<p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","s=
ans-serif";color:navy'>&nbsp; To avoid these problems, translators implemen=
ting the current BUNDLE will generally need to do SSRC mapping anyway, so B=
UNDLE cannot take full advantage of the &#8220;randomness and flatness of t=
he SSRC space&#8221;.</span><o:p></o:p></p><p class=3DMsoNormal><span style=
=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>RTP media trans=
lators in an RTP session where BUNDLE is used for setup need to set up thei=
r RTP sessions in such a way that either BUNDLE is used on all legs, or BUN=
DLE is used on no legs. This is because RTP media translators don't termina=
te RTP sessions.<br><br>Gateways that terminate RTP sessions have to remap =
SSRCs anyway. No news here.<br><br><o:p></o:p></span></p><p class=3DMsoNorm=
al><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:n=
avy'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-s=
ize:10.0pt;font-family:"Arial","sans-serif";color:navy'>I have read your dr=
aft and should have referenced it in my paper.&nbsp; I agree that different=
 media types should be able to share the same transport connection. &nbsp;B=
ut I also see a need to be able to apply differential treatment of media ty=
pes within the network.</span><o:p></o:p></p><p class=3DMsoNormal><span sty=
le=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>Yes, some app=
lications need that.<br>&nbsp;In that case, the simplest way (allowing per-=
flow prioritization) is to use different RTP sessions on different transpor=
t addresses.<br>The next simplest way (using DiffServ) is to apply differen=
t DSCP markings on a per-packet basis. Both techniques are well established=
.<br><br>Breaking up the SSRC space and applying differential treatment bas=
ed on that requires rewriting the traffic management components. That's new=
 code that needs to be deployed. I don't see the need.<br><br><o:p></o:p></=
span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"=
Arial","sans-serif";color:navy'>&nbsp;</span><o:p></o:p></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";colo=
r:navy'>You disagree with the need to separate packets associated with diff=
erent media lines for differential treatment in the network. </span><o:p></=
o:p></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt;font-family:"T=
imes New Roman","serif"'>No, I don't disagree at all with that need, and ha=
ve said so many times. I disagree that this is a need that is present *all =
the time*, and disagree that this need is so omnipresent that we need to sh=
ape our architectures around it.<br><br><o:p></o:p></span></p><p class=3DMs=
oNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";co=
lor:navy'>&nbsp;Nevertheless, this issue has been raised in RFC 3550 and nu=
merous drafts since then as a legitimate issue in certain deployments. &nbs=
p;While I agree with you that there are significant use cases in which such=
 differential treatment is not necessary, there are others in which it can =
be very beneficial. &nbsp;It would be really unfortunate if there were no w=
ay to support such use cases in rtcweb.</span><o:p></o:p></p><p class=3DMso=
Normal><span style=3D'font-size:12.0pt;font-family:"Times New Roman","serif=
"'>The point I was trying to make in rtp-sess-neutral is that the need for =
such differential treatment is orthogonal to the media types, and applicati=
ons need to assign media streams to RTP sessions as the *applications*, not=
 the *architecture*, think is adequate. That's really the focus of BUNDLE.<=
br><br>But really, this discussion belongs on AVTCORE.<br><br><o:p></o:p></=
span></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"=
Arial","sans-serif";color:navy'>&nbsp;</span><o:p></o:p></p><p class=3DMsoN=
ormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";colo=
r:navy'>Richard</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'fo=
nt-size:10.0pt;font-family:"Arial","sans-serif";color:navy'>&nbsp;</span><o=
:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-fami=
ly:"Arial","sans-serif";color:navy'>&nbsp;</span><o:p></o:p></p><div><div c=
lass=3DMsoNormal align=3Dcenter style=3D'text-align:center'><span style=3D'=
font-size:12.0pt;font-family:"Times New Roman","serif";color:windowtext'><h=
r size=3D2 width=3D"100%" align=3Dcenter></span></div><p class=3DMsoNormal>=
<b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:=
windowtext'>From:</span></b><span style=3D'font-size:10.0pt;font-family:"Ta=
homa","sans-serif";color:windowtext'> Harald Alvestrand [<a href=3D"mailto:=
harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br><b>Sent:</b> Mon=
day, June 04, 2012 1:43 PM<br><b>To:</b> Ejzak, Richard P (Richard)<br><b>C=
c:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject=
:</b> Re: [rtcweb] draft on media multiplexing submitted to AVTCORE</span><=
o:p></o:p></p></div><p class=3DMsoNormal><span style=3D'font-family:"Futura=
A Bk&#13;&#10;                BT","serif"'>&nbsp;</span><o:p></o:p></p><p c=
lass=3DMsoNormal><span style=3D'font-family:"FuturaA Bk&#13;&#10;          =
      BT","serif"'>Richard,<br>having scanned your draft:<br><br>- I still =
fundamentally think that the partition of media types into separate RTP ses=
sions was a design mistake that we should seek to rectify, not promulgate, =
and additional complexity like what you propose for perpetuating the use of=
 &quot;separated&quot; sessions is not helpful. See draft-alvestrand-rtp-se=
ss-neutral for details.<br><br>- When we have parts of the ecosystem that h=
ave code that is written on the assumption that the SSRC space is a 32-bit =
flat space, I think the partitioning of that space into subspaces as you pr=
opose is likely to lead to &quot;interesting&quot; bugs. We should reinforc=
e the randomness and flatness of the SSRC space, not partition it.<br><br>-=
 You say &quot;The biggest problem with BUNDLE is that it is<br>&nbsp;&nbsp=
; difficult to partition the RTP streams associated with different<br>&nbsp=
;&nbsp; media lines since this requires filtering on sets of payload type<b=
r>&nbsp;&nbsp; values.&nbsp; RTP subsessions is superior for this purpose s=
ince it is<br>&nbsp;&nbsp; only necessary to screen for a single value in t=
he first 16 bits&quot;.<br>This assumes that the partitioning associated wi=
th different media lines is a good thing.<br>I claim that it is not.<br><br=
>If there are no bigger problems than that with BUNDLE, I would prefer to s=
tick with that approach.<br><br><br><br><br><br>On 06/04/2012 01:09 AM, Ejz=
ak, Richard P (Richard) wrote: </span><o:p></o:p></p><p class=3DMsoNormal><=
span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>Please see=
 the mail attached below that I just sent to avtcore announcing a personal =
ID I just submitted on the topic of media multiplexing. &nbsp;I am sending =
this notice to rtcweb since this is the group that triggered recent work on=
 RTP multiplexing.</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D=
'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span><o:p></o:p=
></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Aria=
l","sans-serif"'>I also request the rtcweb chairs consider giving me some t=
ime to present this work at next week&#8217;s interim meeting if there is t=
ime on the agenda. &nbsp;It might be useful to discuss the ideas in the ID =
before avtcore meets in Vancouver.</span><o:p></o:p></p><p class=3DMsoNorma=
l><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&nbsp;<=
/span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:10.0pt;f=
ont-family:"Arial","sans-serif"'>Richard</span><o:p></o:p></p><p class=3DMs=
oNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>&=
nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-fami=
ly:"Courier&#13;&#10;                New","serif"'>-----Original Message---=
--<br>From: Ejzak, Richard P (Richard) <br>Sent: Sunday, June 03, 2012 6:08=
 PM<br>To: '<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a>'<br>Subject: [=
AVTCORE] multi session draft-ejzak-avtcore-rtp-subsessions-00.txt</span><o:=
p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13=
;&#10;                New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3D=
MsoPlainText><span style=3D'font-family:"Courier&#13;&#10;                N=
ew","serif"'><a href=3D"http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subs=
essions-00.txt">http://www.ietf.org/id/draft-ejzak-avtcore-rtp-subsessions-=
00.txt</a></span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font=
-family:"Courier&#13;&#10;                New","serif"'>&nbsp;</span><o:p><=
/o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#=
10;                New","serif"'>In response to the chairs' request for add=
itional input on the multi session issue, I have submitted this draft for y=
our consideration.&nbsp; There are superficial similarities with an expired=
 draft from last year in <a href=3D"http://tools.ietf.org/id/draft-rosenber=
g-rtcweb-rtpmux-00.txt">http://tools.ietf.org/id/draft-rosenberg-rtcweb-rtp=
mux-00.txt</a>, but my draft has a different take on the use of SSRC as the=
 basis for multi session multiplexing that I think is superior and worthy o=
f consideration as a potential replacement for BUNDLE and/or SHIM.&nbsp; Pl=
ease comment.</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'f=
ont-family:"Courier&#13;&#10;                New","serif"'>&nbsp;</span><o:=
p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13=
;&#10;                New","serif"'>Richard</span><o:p></o:p></p><p class=
=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;              =
  New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span s=
tyle=3D'font-family:"Courier&#13;&#10;                New","serif"'>-----Or=
iginal Message-----</span><o:p></o:p></p><p class=3DMsoPlainText><span styl=
e=3D'font-family:"Courier&#13;&#10;                New","serif"'>From: <a h=
ref=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a> [<a hr=
ef=3D"mailto:internet-drafts@ietf.org">mailto:internet-drafts@ietf.org</a>]=
 </span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"=
Courier&#13;&#10;                New","serif"'>Sent: Sunday, June 03, 2012 =
5:16 PM</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-fa=
mily:"Courier&#13;&#10;                New","serif"'>To: Ejzak, Richard P (=
Richard)</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-f=
amily:"Courier&#13;&#10;                New","serif"'>Subject: New Version =
Notification for draft-ejzak-avtcore-rtp-subsessions-00.txt</span><o:p></o:=
p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;=
                New","serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoPla=
inText><span style=3D'font-family:"Courier&#13;&#10;                New","s=
erif"'>A new version of I-D, draft-ejzak-avtcore-rtp-subsessions-00.txt has=
 been successfully submitted by Richard Ejzak and posted to the IETF reposi=
tory.</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-fami=
ly:"Courier&#13;&#10;                New","serif"'>&nbsp;</span><o:p></o:p>=
</p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;  =
              New","serif"'>Filename:&nbsp;&nbsp; draft-ejzak-avtcore-rtp-s=
ubsessions</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font=
-family:"Courier&#13;&#10;                New","serif"'>Revision:&nbsp;&nbs=
p; 00</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-fami=
ly:"Courier&#13;&#10;                New","serif"'>Title:&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Media multiplexing with Re=
al-time Transport Protocol (RTP) subsessions</span><o:p></o:p></p><p class=
=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;              =
  New","serif"'>Creation date:&nbsp;&nbsp;&nbsp; 2012-06-04</span><o:p></o:=
p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;=
                New","serif"'>WG ID:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp; Individual Submission</span><o:p></o:p></p><p c=
lass=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;          =
      New","serif"'>Number of pages: 9</span><o:p></o:p></p><p class=3DMsoP=
lainText><span style=3D'font-family:"Courier&#13;&#10;                New",=
"serif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D=
'font-family:"Courier&#13;&#10;                New","serif"'>Abstract:</spa=
n><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courie=
r&#13;&#10;                New","serif"'>&nbsp;&nbsp; This document describ=
es a means of multiplexing RTP streams having</span><o:p></o:p></p><p class=
=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;              =
  New","serif"'>&nbsp;&nbsp; different media types within a single transpor=
t connection and how to</span><o:p></o:p></p><p class=3DMsoPlainText><span =
style=3D'font-family:"Courier&#13;&#10;                New","serif"'>&nbsp;=
&nbsp; represent this multiplexing option in SDP.&nbsp; This mechanism is a=
n</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"=
Courier&#13;&#10;                New","serif"'>&nbsp;&nbsp; alternative to =
the BUNDLE and SHIM proposals currently under active</span><o:p></o:p></p><=
p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;       =
         New","serif"'>&nbsp;&nbsp; consideration in AVTCORE.&nbsp; Instead=
 of adding an extra multiplexing</span><o:p></o:p></p><p class=3DMsoPlainTe=
xt><span style=3D'font-family:"Courier&#13;&#10;                New","serif=
"'>&nbsp;&nbsp; header as in SHIM to allow multiple RTP sessions within a s=
ingle</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-fami=
ly:"Courier&#13;&#10;                New","serif"'>&nbsp;&nbsp; transport c=
onnection, or using the payload type field to separate</span><o:p></o:p></p=
><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;     =
           New","serif"'>&nbsp;&nbsp; different media streams within a sing=
le RTP session, this document</span><o:p></o:p></p><p class=3DMsoPlainText>=
<span style=3D'font-family:"Courier&#13;&#10;                New","serif"'>=
&nbsp;&nbsp; describes how to partition the existing SSRC space to create R=
TP</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:=
"Courier&#13;&#10;                New","serif"'>&nbsp;&nbsp; subsessions fr=
om a single RTP session.&nbsp; A filter can be used to</span><o:p></o:p></p=
><p class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;     =
           New","serif"'>&nbsp;&nbsp; identify each RTP subsession for diff=
erent QoS handling as necessary.</span><o:p></o:p></p><p class=3DMsoPlainTe=
xt><span style=3D'font-family:"Courier&#13;&#10;                New","serif=
"'>&nbsp;&nbsp; RTP subsessions can be treated like RTP sessions with a few=
</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"C=
ourier&#13;&#10;                New","serif"'>&nbsp;&nbsp; restrictions.&nb=
sp; In particular, SSRC mapping may be needed when</span><o:p></o:p></p><p =
class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;         =
       New","serif"'>&nbsp;&nbsp; forwarding RTP streams into an RTP subses=
sion to avoid SSRC</span><o:p></o:p></p><p class=3DMsoPlainText><span style=
=3D'font-family:"Courier&#13;&#10;                New","serif"'>&nbsp;&nbsp=
; conflicts, but there are few use cases in which this limitation is a</spa=
n><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Courie=
r&#13;&#10;                New","serif"'>&nbsp;&nbsp; concern and RTP subse=
ssions can be disabled if necessary.</span><o:p></o:p></p><p class=3DMsoPla=
inText><span style=3D'font-family:"Courier&#13;&#10;                New","s=
erif"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'f=
ont-family:"Courier&#13;&#10;                New","serif"'>&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; </span><o:p></o:p></p><p class=3DMsoPlainText><span st=
yle=3D'font-family:"Courier&#13;&#10;                New","serif"'>&nbsp;</=
span><o:p></o:p></p><p class=3DMsoPlainText><span style=3D'font-family:"Cou=
rier&#13;&#10;                New","serif"'>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoPlainText><span style=3D'font-family:"Courier&#13;&#10;         =
       New","serif"'>The IETF Secretariat</span><o:p></o:p></p><p class=3DM=
soNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-serif"'>=
&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:1=
0.0pt;font-family:"Arial","sans-serif"'>&nbsp;</span><o:p></o:p></p><p clas=
s=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Arial","sans-ser=
if"'>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal style=3D'margin-botto=
m:12.0pt'><span style=3D'font-size:12.0pt;font-family:"Times New Roman","se=
rif"'><br><br></span><o:p></o:p></p><pre>__________________________________=
_____________<o:p></o:p></pre><pre>rtcweb mailing list<o:p></o:p></pre><pre=
><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pr=
e><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf=
.org/mailman/listinfo/rtcweb</a><o:p></o:p></pre><p class=3DMsoNormal><span=
 style=3D'font-size:12.0pt;font-family:"Times New Roman","serif"'>&nbsp;</s=
pan><o:p></o:p></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt;fon=
t-family:"Times New Roman","serif"'><o:p>&nbsp;</o:p></span></p></div></bod=
y></html>=

--_000_7F2072F1E0DE894DA4B517B93C6A05852C45A1C906ESESSCMS0356e_--

From csp@csperkins.org  Tue Jun  5 06:18:23 2012
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] draft on media multiplexing submitted to AVTCORE
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On 5 Jun 2012, at 07:49, Harald Alvestrand wrote:
> On 06/04/2012 10:03 PM, Ejzak, Richard P (Richard) wrote:
...
>>  Even though RTP packets with different payload types could =
potentially reuse the same SSRC value, SSRC values cannot be =
independently assigned per m line with the current BUNDLE scheme since =
RTCP packets can be identified only using SSRC values (payload type is =
not a field in RTCP packets).  So the bundled m lines have to share a =
single SSRC space, thus increasing the likelihood of SSRC collision
>=20
> Yes. At 302 SSRCs in the same RTP session the chance of a collision is =
50%. Applications that don't handle SSRC collisions are broken.

I agree that applications that don't handle SSRC collisions are broken, =
but RFC 3550 Section 8.1 gives a vastly lower probability of collision. =
Is there a mistake in the calculation in the RFC?

--=20
Colin Perkins
http://csperkins.org/




From internet-drafts@ietf.org  Tue Jun  5 09:32:32 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Security Considerations for RTC-Web
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-03.txt
	Pages           : 22
	Date            : 2012-06-05

   The Real-Time Communications on the Web (RTC-Web) working group is
   tasked with standardizing protocols for real-time communications
   between Web browsers.  The major use cases for RTC-Web technology are
   real-time audio and/or video calls, Web conferencing, and direct data
   transfer.  Unlike most conventional real-time systems (e.g., SIP-
   based soft phones) RTC-Web communications are directly controlled by
   some Web server, which poses new security challenges.  For instance,
   a Web browser might expose a JavaScript API which allows a server to
   place a video call.  Unrestricted access to such an API would allow
   any site which a user visited to "bug" a user's computer, capturing
   any activity which passed in front of their camera.  This document
   defines the RTC-Web threat model and defines an architecture which
   provides security within that threat model.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-security-03.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-security-03.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/


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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-security-arch-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : RTCWEB Security Architecture
	Author(s)       : Eric Rescorla
	Filename        : draft-ietf-rtcweb-security-arch-02.txt
	Pages           : 20
	Date            : 2012-06-05

   The Real-Time Communications on the Web (RTCWEB) working group is
   tasked with standardizing protocols for enabling real-time
   communications within user-agents using web technologies (e.g
   JavaScript).  The major use cases for RTCWEB technology are real-time
   audio and/or video calls, Web conferencing, and direct data transfer.
   Unlike most conventional real-time systems (e.g., SIP-based soft
   phones) RTCWEB communications are directly controlled by some Web
   server, which poses new security challenges.  For instance, a Web
   browser might expose a JavaScript API which allows a server to place
   a video call.  Unrestricted access to such an API would allow any
   site which a user visited to "bug" a user's computer, capturing any
   activity which passed in front of their camera.  [I-D.ietf-rtcweb-
   security] defines the RTCWEB threat model.  This document defines an
   architecture which provides security within that threat model.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-security-arch-02.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-security-arch-02.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/


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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, Jun 5, 2012 at 9:36 AM
Subject: Re: [rtcweb] Review of security documents
To: Martin Thomson <martin.thomson@gmail.com>


> =A0THOMSON:
Martin,

Thanks for your detailed review.

Comments/responses below. Let me know if you think I have
gotten something wrong.

> HTTP/HTTPS
>
> S3, last paragraph before S3.1:
>
> =A0 [...], but realistically many sites do not run
> =A0 HTTPS [RFC2818] and so our ability to defend against network
> =A0 attackers is necessarily somewhat limited.
>
> This isn't especially relevant.
>
> Obviously, the standard class of problems with unsecured HTTP exist, but
> within the context of this application, there aren't that many more that
> this enables. =A0The example in S4.1.3 is not unique to this application.
> It applies to any user consent that is tied to a particular web origin.
>
> In comparison to possibly visiting and _using_ a site operated by a web
> attacker, this is not substantially worse, or requiring significantly
> more effort to analyze.
>
> Of course, the only safe assumption is that you are talking to a web
> attacker when using unsecured HTTP.

> ((Aside: I'd be interested in learning how this might turn into an
> attack on user consent. =A0Otherwise, this is a bit of a scary statement
> to just drop in without support:
> =A0 Note: =A0this issue is not restricted to PAGES
> =A0 which contain mixed content. =A0If a page from a given origin ever
> =A0 loads mixed content then it is possible for a network attacker to
> =A0 infect the browser's notion of that origin semi-permanently.))
>
> NETWORK ATTACKERS
>
> Same paragraph:
>
> =A0 [...], with the assumption that
> =A0 protection against network attackers is provided by running HTTPS.
>
> Thankfully, the draft does not make this assumption. =A0Attacks on HTTP
> are out of scope, but we still need to deal with network attackers.

Thanks for catching this inconsistency. I'm taking the sum of these
comments as "stop acting like you're not addressing HTTP." So, I
rewrote this graf as:

=A0 =A0 =A0 =A0Conventionally, we refer to either WEB ATTACKERS, who are ab=
le
=A0 =A0 =A0 =A0to induce you to visit their sites but do not control the
=A0 =A0 =A0 =A0network, and NETWORK ATTACKERS, who are able to control your
=A0 =A0 =A0 =A0network. Network attackers correspond to the <xref
=A0 =A0 =A0 =A0target=3D"RFC3552"/> "Internet Threat Model". Note that for =
HTTP
=A0 =A0 =A0 =A0traffic, a network attacker is also a Web attacker, since it
=A0 =A0 =A0 =A0can inject traffic as if it were any non-HTTPS Web site. Thu=
s,
=A0 =A0 =A0 =A0when analyzing HTTP connections, we must assume that traffic
=A0 =A0 =A0 =A0is going to the attacker.

> TYPES OF CONSENT
>
> The document talks about consent in general as being important but it
> doesn't do anything to address what specific consent is needed.
>
> I think that it has been well-established that consent is required for
> access to input devices (e.g., camera/microphone). =A0The implication fro=
m
> S4.1 is that this is sufficient as well as necessary. =A0There is one
> crucial piece of the argument that is absent:
>
> =A0 A site with access to camera or microphone could send media either to
> =A0 itself or any site that indicates consent (see CORS). =A0Sending medi=
a
> =A0 over HTTP or thewebsocketprotocol is likely to perform less well than
> =A0 is ideal, but it would work.
>
> Therefore, it's easy to draw the implicit conclusions of the draft,
> namely:
> =A01. a receipt consent mechanism like CORS is necessary, and
> =A02. user consent for access to input devices is necessary _and_
> =A0 =A0 sufficient...for this mode.

I've tried to address this some more in Section 4.1.


> PRIVATE COMMUNICATIONS AND CONSENT
>
> The above assumes that the site has access to media. =A0That is,
> permission for input devices is being granted to the site. =A0However, it
> is possible to imagine a mode of communications where the site mediates
> the creation of a secure channel, but does not have access to that
> channel.

IMO there are actually two issues here:

(1) consent
(2) peer authentication

To give an example, I might be happy to have Gmail be able to
access my camera and microphone without a big consent experience,
but I still want to be able to have calls that I know Gmail
can't listen in on in. That way I can have easy calls, but
when I want security I can have it. I have tried to make this
clear in the new text.


> This changes the assumptions - and the nature of the consent -
> dramatically. =A0S4.1.2 and security-arch@S5.2 covers this case, but
> doesn't really properly emphasize just how different this is.
>
> Just as for the entirety of S4.1, the problem then becomes one of
> unambiguous identification. =A0And UI. =A0From S4.1.2:
>
> =A0 Naturally, it is somewhat challenging to design UI
> =A0 primitives which express this sort of policy.
>
> Complications include group calling. =A0How does the site ask for
> permissions to talk to "a@b.com" and "x@y.net"? =A0How does such a
> privilege persist? =A0Does it even make sense to persist at all?
> Obviously, a conference of any significant size tends toward having a
> bridge. =A0At that point, input devices are most likely granted to
> "host.example.com".

I don't actually think this is an issue at the WebRTC level. ISTM
that there are two ways to do a multi-user call in WebRTC:

(a) a set of individual calls, which the browser handles separately.
(b) a call to a bridge which manifests as a call to someone other
than a@b.com or x@y.net

So in either case, not our problem.


> TRULY PRIVATE COMMUNICATIONS
>
> Keeping the site out of the loop requires that the browser lock down
> access to media recording. =A0The trick is to ensure that the media is
> unmodified all the way from source to sink. =A0It's relatively easy to
> ensure that media coming off the network is unmodified.
>
> It is harder to ensure that it did not get modified by a web attacker
> prior to being put on the network. =A0That requires a verifiable assertio=
n
> from the remote user that it did not allow a web attacker access to see
> or modify the stream prior to it being placed in the pipe.
>
> The distinction between media stream visibility and modifiability might
> be worth discussing a little. =A0My initial thought is that it was not
> especially useful in this context. =A0I can imagine work-arounds that
> would enable features that depend on visibility such as recording where
> authenticity is also desirable.

Good point. I have added a new section to make this point.


> IDENTIFICATION
>
> S4.1.1.1.1.1.1.1.1 asks the obvious question:
>
> =A0 [...] there is a question about whether the
> =A0 user can know who they are talking to.
>
> When a site has access to your media, then you are talking to the site.
> ...and anyone the site chooses to forward your media to. =A0This is
> exactly what you get when you use SRTP security descriptions (and also
> EKT, if you allow the site to insert keys).

See above.


> USER INTERACTION CONSIDERATIONS
>
> S4.1.1.2 hides two important UI considerations:
>
> =A0 [...] great care must be taken in the design of this interface
> =A0 to avoid the users just clicking through
>
> and
>
> =A0 [...] the user
> =A0 interface chrome must clearly display elements showing that the call
> =A0 is continuing in order to avoid attacks where the calling site just
> =A0 leaves it up indefinitely
>
> These are both massively important. =A0If there were a W3C companion
> document, then it would make sense to include this sort of stuff there.
>
> More robust treatment would be nice for:
> =A0a) the limitations of consent mechanisms
> =A0b) providing appropriate feedback
>
> For the latter, this can get complicated. =A0I recall a discussion of the
> geolocation API that lead to the conclusion that no UI feedback would be
> provided. =A0I still believe that this was a poor choice. =A0This case be=
ars
> a certain amount of similarity to that discussion. =A0I should really joi=
n
> that media capture group...
>
> In any case, this probably needs at least basic treatment here, even if
> that is just by reference.

I agree we should say something about this, but I think it's best
said in the W3C specs. I will work to produce something.


> AUTHENTICATION MODELS
>
> (Crossing over into S4 of the security-arch document here to address the
> use cases from S4.1.1.2 and S4.1.1.3.)
>
> It looks like there is an assumption in play here. =A0That is, there is
> something like an IdP in use. =A0Calling a site (as opposed to a person)
> is very much a case where the usual domain trust anchors are perfectly
> adequate. =A0The site can offer an identical (or similar) certificate in
> the DTLS handshake as it did with the HTTPS TLS handshake.
>
> Obviously, it's not as simple as just having this chain to a trust
> anchor. =A0Any site that deployed TLSA (c.f. DANE WG) would require extra
> checking.
>
> I'm surprised to see nothing on this particular use case. =A0It's a
> particularly useful one.

OK. I'll add something to the security-arch document.


> CALL MEDIATION AND REPUTATION
>
> S4.1.1.3 contains a discussion about a case where the reputation of one
> site is potentially affected by it mediating a call with a third party.
> In the example, sites wouldn't want advertisements they display from
> reflecting poorly on them.
>
> I don't see how this changes the current situation significantly. =A0Site
> and ad networks work very hard to ensure that advertisements are
> appropriate to the context in which they are shown. =A0Realtime calling
> doesn't really change this situation in any meaningful way.
>
> A very big part of this is ensuring that the source of media is
> correctly identified. =A0In part, this is the same problem we already
> have. =A0In addition, sites need to take some responsibility for making
> any necessary distinction sufficiently clear on their own interface. =A0I=
n
> this context, the burden lies with the ad network.

Hmm... my experience is that sites and ad networks don't do
that much vetting of JS, actually. I've done ad network
studies where we put arbitrary JS that does a pile of
XHR-type stuff in ads and nobody seemed to notice.


> ((Aside. =A0I'm not sure that this is always true:
> =A0 [...] sites which host
> =A0 advertisements often learn very little about whether individual users
> =A0 clicked through to the ads, or even which ads were presented
>
> That's true for ads in iframes, but then the reputation concern isn't
> entirely applicable. =A0If the ads are served inline, then I would have t=
o
> assume that obfuscation of ad network content is the only protection
> against the site learning about ad content and user interactions.))

Yes, that's correct. I added "in IFRAMEs".


> COMMUNICATIONS CONSENT
>
> The idea that a target for a media flow consents is not binary in the
> sense that ICE establishes. =A0Even with the addition of an expiration
> time to ICE consent, there are two problems related to the volume of
> packets that need addressing:
>
> The rate at which STUN Binding requests are generated in ICE is
> dependent on the bandwidth available for media. =A0Because this
> information comes from a web attacker, the browser cannot trust this
> information. =A0The browser has to rate-limit Binding requests based on
> its own information. =A0In practice, this probably has to be based on a
> fixed rate.
>
> Even once the browser has validated consent, it has no idea _how much_
> media is OK to send. =A0The difference between 8kbps audio and 5Mbps high
> definition video is enough to make many an internet connection melt.
> RTCP feedback like TMMBR is good, but maybe those packets could be made
> to get "lost" by a network attacker. =A0Plus, if we are going to offer a
> data channel that doesn't use RTCP, then that option isn't available.

So, this is definitely a real issue, but I'm not sure how far we want
to go in doing something about it. My thought had been that we wouldn't
attempt to obtain explicit consent for a particular bandwidth rate
and instead assume that if the sending side is maliciously
sending over-bandwidth that the receiving side will drop
continuing consent probes and cause the connection to be
torn down. That said, I think this is something we need to
actually discuss, so I'll mark it an open issue and
let's talk about it in the meeting as part of the continuing
consent discussion.


> BANDWIDTH LIMITING
>
> Bandwidth limiting is probably an important security feature that isn't
> really covered, though you touch on it in the security considerations of
> security-arch.

See above.


> SPECULATIVE STUFF
>
> Don't think that we need masking: TURN TCP probably has enough
> protection already and that's the only TCP I can imagine having to use,
> aside from HTTPS...

I tend to agree, but I think it's worth leaving here as part of
the analytical framework.


> Don't think that we need some sort of implicit consent mechanism. =A0For
> consent, I don't think that it's unreasonable to expect ICE-lite at a
> minimum.

Me neither. I think that's sort of what the -arch document suggests
as well.


> See thread on hiding IP addresses. =A0In order to avoid tracking via IP,
> clients should not use the Internet. =A0I hear that Tor is somewhat handy
> at making it possible to have cake that you can eat too. =A0That said, I'=
m
> not sure that that particular cake tastes all that nice...

I'm not sure either, but.... what are you looking for he?


> IdP
>
> MIXED CONTENT
>
> security-arch@S5.1
>
> It might make sense to consider _all_ unsecured HTTP is being in the
> same origin for the purposes of this.

I'm not sure what you are suggesting? Seems like a big error to
treat http://www.foo.com/ and http://www.bar.com/ as the same
for consent purposes.


> Furthermore, I subscribe to the view that mixed content =3D=3D unsecured.

Hmm.... I mostly agree with this, but I think that web attackers
are real....


> The idea that a page might become mixed content is a real concern.
> Terminating the session is probably the only safe choice.
>
> Here's the thought process. =A0Initially, I thought that it might be
> sufficient to prevent modifications to the session once a page goes
> rogue. =A0At least then if a secure browser-to-browser pipe exists that
> the page could not access prior to the poisoning, this pipe can continue
> unmodified. =A0However, if one browser goes off the grid, then it can
> simply exploit any renegotiation capabilities that exist in the
> application to trigger changes through the signaling channel. =A0For
> instance, EKT might allow the attacker to update keys. =A0Given that it
> should also be possible to insert a relay (in one direction at least) by
> providing faked "candidates", this cracks it wide open. =A0The more
> renegotiation capabilities exist on the media path, the worse this is.

I generally agree with this.


> ICE TRANSACTION ID
>
> ...need only be hidden for Binding requests that are outstanding.
> Successful requests need not be hidden. =A0(Note that this means meeting
> all of the success criteria specified in RFC 5245.)

Agreed.

> ICE KEEPALIVES
>
> Are not sufficient, so don't require them. =A0Instead, require a repeat o=
f
> the connectivity check. =A0That ensures consent is refreshed and it also
> means that the statement about ICE-Lite remains true, which would not be
> so if another mechanism were used.

This seems like it's an open issue to discuss during the interim,
since I know there have been some questions here.


> UNLINKABILITY
>
> security-arch@S5.5
>
> Nothing on this in the security doc. =A0What is the goal here?

I'm not sure I understand the question. I want to be able
to make calls to people without having them linked?


> Minting a new key can be expensive. =A0How do you prevent sites from
> pushing the button too often and causing a different sort of problem?

What do you mean? The site can always cause the browser to
burn arbitrary amounts of CPU (though this typically will
eventually cause the browser to make some kind of sad
face), just by infinite looping.

-Ekr

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---------- Forwarded message ----------
From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, Jun 5, 2012 at 9:35 AM
Subject: Re: [rtcweb] Review of draft-ietf-rtcweb-security-02 and
draft-ietf-rtcweb-security-arch-01 documents
To: "DRUTA, DAN" <dd5826@att.com>


Dan,

Thanks for your review, comments below.


> I reviewed the latest version of the drafts and below are my comments/sug=
gestions/corrections (not as comprehensive as Mark's).
> In general I think the documents are well written with good references an=
d paint a rather comprehensive picture of the issues to be addressed for rt=
cweb.
>
> Draft-ietf-rtcweb-security-02
>
> 1.Introduction - =A0second paragraph after the diagram:
> "Each of their browsers exposes standardized JavaScript calling APIs whic=
h are used by the Web server to set up a call between Alice and Bob."
>
> =A0This statement can be a bit more clear in that the APIs are executed
> =A0locally (in the user agent) as java script downloaded from the Web
> =A0Server. The next paragraph eludes to the fact that logic is
> =A0controlled by the calling service but I think it will be important
> =A0to distinguish between APIs implemented by the user-agent and
> =A0APIs/functions implemented by the calling service.

I added "implemented as browser built-ins". Hope this helps.


> 4.1 Access to Local Devices.
> Since this describes the threat I think is important to point out
> the fact that even when the user provides consent it is difficult
> for them to determine the real reason they give the consent to a
> site. Take the scenario in which a site obtains consent from the
> user for an app that captures a clip and saves it on the user's hard
> drive. Later on, if the user gave permanent consent to the site, the
> site can obtain access to the camera for the purpose of streaming
> without the user knowing that. This can be confusing for users even
> if they trust the site.

Right. This is a good example of the gap between the policy the user
wishes to have enforced (my information will be used only for purposes
A and B and not purpose C) and what we can technically enforce (site X
can have my data but they do whatever they want...)


> 4.2 IP location Privacy
> "Note that as soon as the callee sends their ICE candidates, the callee l=
earns the callee's IP addresses."
> This must be "the caller learns the callee's IP address"

Good catch. Fixed.



> 4.3.2.3. Third Party Identity
> "It is possible (see[I-D.rescorla-rtcweb-generic-idp]) to use systems of =
this type to authenticate RTCWEB calls, linking them to existing user notio=
ns of
> identity (e.g., Facebook adjacencies). Calls which are authenticated in t=
his fashion are naturally resistant even to active MITM attack by the calli=
ng site."
>
> Not to get too much into semantics, I think it is important to make
> the distinction that users are authenticated (using whatever IdP)
> and calls are authorized based on the assertions provided by the Idp
> for the authenticated user.

Good point. Added something. Let me know if you think it helps.



> Draft-ietf-rtcweb-security-arch-01
>
> 1.Introduction
> "The Real-Time Communications on the Web (RTCWEB) working group is tasked=
 with standardizing protocols for real-time communications between Web brow=
sers."
>
> While there's nothing inaccurate about the statement, I would like to mak=
e it more generic because it is more than just communication between web br=
owsers. In my opinion the WEB part in the acronym is all about using the we=
b technologies to enable real time communications.
>
> A proposed edited definition would be "... standardization protocols
> for enabling real-time communications within user-agents using web
> technologies (e.g JavaScript)".

That's an improvement. Accepted.

> I have been trying to identify the appropriate section for a rather contr=
oversial requirement: =A0Lawful Intercept. It is expected that service prov=
iders for RTCWeb/WebRTC will be required to provide support to lawful inter=
cept and I think we need to capture these requirements so we can map them t=
o the security architecture.

I think Martin's assessment of the situation here is pretty much right.

-Ekr

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Tue, 5 Jun 2012 19:30:54 +0200
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Subject: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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Hi, just wondering if there could be some "workaround" for this "issue":

I've a VPN (Cisco vpnc) interface in my host. Using Chrome dev-channel
and setting a public STUN server in my WebRTC application I've
realized that during ICE stuf it also takes the VPN interface (which
makes sense) but when it tries to contact the STUN server from the VPN
interface it will have not response (due to VPN/network routes and
so...) which causes a "STUN timeout" (~2-3 seconds). Of course this
happens for *every* outgoing or incoming media session when the
browser needs to create a SDP.

I don't think there is a solution for this "issue", just wondering.

Regards.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Selective responses inline.

On 5 June 2012 09:50, Eric Rescorla <ekr@rtfm.com> wrote:
>> Therefore, it's easy to draw the implicit conclusions of the draft,
>> namely:
>> =C2=A01. a receipt consent mechanism like CORS is necessary, and
>> =C2=A02. user consent for access to input devices is necessary _and_
>> =C2=A0 =C2=A0 sufficient...for this mode.
>
> I've tried to address this some more in Section 4.1.

I like it.

>> PRIVATE COMMUNICATIONS AND CONSENT
>>
>> The above assumes that the site has access to media. =C2=A0That is,
>> permission for input devices is being granted to the site. =C2=A0However=
, it
>> is possible to imagine a mode of communications where the site mediates
>> the creation of a secure channel, but does not have access to that
>> channel.
>
> IMO there are actually two issues here:
>
> (1) consent
> (2) peer authentication
>
> To give an example, I might be happy to have Gmail be able to
> access my camera and microphone without a big consent experience,
> but I still want to be able to have calls that I know Gmail
> can't listen in on in. That way I can have easy calls, but
> when I want security I can have it. I have tried to make this
> clear in the new text.

I think in your situation you want a separate type of consent - access
to a specific "session".  The idea of what a "session" entails
probably needs better definition than this, but I'm sure you know what
I mean.  (I'd probably associate this with a single transport - and
all streams that are attached to that transport, but you don't get to
do that with the current API.  More on that in future, I promise.)

When you consent to granting the site access to input devices, that
site can then record you.  Presumably, the site also gets to record
streams of remote origin without that consent.  What you desire is
something more than just consent to access a device, or peer
authentication...it's a "hands off" signal from the user.

As I've already said, tying the "hands off" signal into peer
authentication is probably a sensible optimization.

>> AUTHENTICATION MODELS
>> [...]=C2=A0The site can offer an identical (or similar) certificate in
>> the DTLS handshake as it did with the HTTPS TLS handshake.
>
> OK. I'll add something to the security-arch document.

Where exactly?

>> CALL MEDIATION AND REPUTATION
> Hmm... my experience is that sites and ad networks don't do
> that much vetting of JS, actually. I've done ad network
> studies where we put arbitrary JS that does a pile of
> XHR-type stuff in ads and nobody seemed to notice.

I've seen a range of responses from sites.  Some sites vet advertisers
carefully, others could care less as long as the money flows.  Those
that vet have usually been stung once already.

If reputation is important to you, then it is your responsibility to
safeguard your own reputation.  If you rely on others, then you can
use technical measures (checking, etc...), or simply rely on their own
desire to safeguard their reputation.

In summary, I don't see a need for a specific technical solution to
this problem.

>> See thread on hiding IP addresses. =C2=A0In order to avoid tracking via =
IP,
>> clients should not use the Internet. =C2=A0I hear that Tor is somewhat h=
andy
>> at making it possible to have cake that you can eat too. =C2=A0That said=
, I'm
>> not sure that that particular cake tastes all that nice...
>
> I'm not sure either, but.... what are you looking for he?

Section 5.4 stipulates requirements that I don't think are reasonable.
 Including discussion on the subject, including countermeasures, with
a conclusion that there are no requirements on the API would be good.
However, guidance for site implementers would be sensible.

>> It might make sense to consider _all_ unsecured HTTP is being in the
>> same origin for the purposes of this.
>
> I'm not sure what you are suggesting? Seems like a big error to
> treat http://www.foo.com/ and http://www.bar.com/ as the same
> for consent purposes.

Sorry, that was too abstract to be actionable.

You already established that - in the presence of a network attacker -
consent to foo.com is equivalent to consent to bar.net.  Therefore, it
seems reasonable to regard the two as being equivalent.  Once you make
that leap, then it is easy to see that you don't have enough
granularity to make any consent meaningful.  So you have to conclude
that providing a (persistent) consent for non-HTTPS sites is
pointless.

>> UNLINKABILITY
>>
>> security-arch@S5.5
>>
>> Nothing on this in the security doc. =C2=A0What is the goal here?
>
> I'm not sure I understand the question. I want to be able
> to make calls to people without having them linked?

What about identity?  While you are in the business of creating an
identity system, wouldn't it be nice if the site you are using
couldn't identify the people using that site?  Imagine that you are
able to create an assertion that you are
dhouhqed08gslkn209eejit8sfsdo@rtfm.com (that well known IdP), which is
translated (by the IdP validator) to a ekr@rtfm.com only in the
browser chrome...

From martin.thomson@gmail.com  Tue Jun  5 11:19:04 2012
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On 5 June 2012 10:30, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:
> [...] but when it tries to contact the STUN server from the VPN
> interface it will have not response (due to VPN/network routes and
> so...) which causes a "STUN timeout" (~2-3 seconds). Of course this
> happens for *every* outgoing or incoming media session when the
> browser needs to create a SDP.
>
> I don't think there is a solution for this "issue", just wondering.

It's not an issue.  Some candidate pairs are always going to fail.
That shouldn't block the testing (and nomination) of other pairs.  If
the ICE implementation struggles or lags as a result, then that's a
fault in the implementation.

From juberti@google.com  Tue Jun  5 12:30:46 2012
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Date: Tue, 5 Jun 2012 15:30:24 -0400
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Subject: Re: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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--20cf3033464dcefbce04c1beafb4
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On Tue, Jun 5, 2012 at 2:19 PM, Martin Thomson <martin.thomson@gmail.com>wr=
ote:

> On 5 June 2012 10:30, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:
> > [...] but when it tries to contact the STUN server from the VPN
> > interface it will have not response (due to VPN/network routes and
> > so...) which causes a "STUN timeout" (~2-3 seconds). Of course this
> > happens for *every* outgoing or incoming media session when the
> > browser needs to create a SDP.
> >
> > I don't think there is a solution for this "issue", just wondering.
>
> It's not an issue.  Some candidate pairs are always going to fail.
> That shouldn't block the testing (and nomination) of other pairs.  If
> the ICE implementation struggles or lags as a result, then that's a

fault in the implementation.
>

If you don't have trickle candidates, you can't send the offer until you
have gotten all candidates, or given up on the ones that haven't responded
yet.

--20cf3033464dcefbce04c1beafb4
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<br><br><div class=3D"gmail_quote">On Tue, Jun 5, 2012 at 2:19 PM, Martin T=
homson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" ta=
rget=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquo=
te class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc so=
lid;padding-left:1ex">

On 5 June 2012 10:30, I=C3=B1aki Baz Castillo &lt;<a href=3D"mailto:ibc@ali=
ax.net">ibc@aliax.net</a>&gt; wrote:<br>
&gt; [...] but when it tries to contact the STUN server from the VPN<br>
<div class=3D"im">&gt; interface it will have not response (due to VPN/netw=
ork routes and<br>
&gt; so...) which causes a &quot;STUN timeout&quot; (~2-3 seconds). Of cour=
se this<br>
&gt; happens for *every* outgoing or incoming media session when the<br>
&gt; browser needs to create a SDP.<br>
&gt;<br>
&gt; I don&#39;t think there is a solution for this &quot;issue&quot;, just=
 wondering.<br>
<br>
</div>It&#39;s not an issue. =C2=A0Some candidate pairs are always going to=
 fail.<br>
That shouldn&#39;t block the testing (and nomination) of other pairs. =C2=
=A0If<br>
the ICE implementation struggles or lags as a result, then that&#39;s a=C2=
=A0</blockquote><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex">fault in the implementation.=
<br></blockquote>

<div><br></div><div>If you don&#39;t have trickle candidates, you can&#39;t=
 send the offer until you have gotten all candidates, or given up on the on=
es that haven&#39;t responded yet.=C2=A0</div></div>

--20cf3033464dcefbce04c1beafb4--

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Subject: Re: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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On 5 June 2012 12:30, Justin Uberti <juberti@google.com> wrote:
> If you don't have trickle candidates, you can't send the offer until you
> have gotten all candidates, or given up on the ones that haven't responded
> yet.

Right.  If you want to collect a server reflexive address for every
host candidate before moving on to other things, then you end up
waiting for a fairly substantial period of time.

There is no reason to gate media negotiation on transport setup.
Trickle candidates is basically a way of saying that you need this
separation.

From internet-drafts@ietf.org  Tue Jun  5 13:48:03 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Javascript Session Establishment Protocol
	Author(s)       : Justin Uberti
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-jsep-01.txt
	Pages           : 31
	Date            : 2012-06-05

   This document proposes a mechanism for allowing a Javascript
   application to fully control the signaling plane of a multimedia
   session, and discusses how this would work with existing signaling
   protocols.

   This document is an input document for discussion.  It should be
   discussed in the RTCWEB WG list, rtcweb@ietf.org.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-01.txt

Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-jsep-01.txt

The IETF datatracker page for this Internet-Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/


From ibc@aliax.net  Tue Jun  5 14:41:50 2012
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2012/6/5 Martin Thomson <martin.thomson@gmail.com>:
> It's not an issue. =C2=A0Some candidate pairs are always going to fail.
> That shouldn't block the testing (and nomination) of other pairs.

Perhaps I didn't explaing properly. What I mean is that when the
browser sends the STUN UDP request from the VPN interface, it will NOT
receive a UDP response in that interface. So this is the worst case:
timeout in receiving a STUN response.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From martin.thomson@gmail.com  Tue Jun  5 14:50:45 2012
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On 5 June 2012 14:41, I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:
> 2012/6/5 Martin Thomson <martin.thomson@gmail.com>:
>> It's not an issue. =C2=A0Some candidate pairs are always going to fail.
>> That shouldn't block the testing (and nomination) of other pairs.
>
> Perhaps I didn't explaing properly. What I mean is that when the
> browser sends the STUN UDP request from the VPN interface, it will NOT
> receive a UDP response in that interface. So this is the worst case:
> timeout in receiving a STUN response.

No misunderstanding.  That's expected behaviour.  And it's not
specific to interfaces that can't reach the specified address.  Some
UDP packets just don't make it.  (Though in this case, there are
several sent, so the odds of all of them getting lost is
astronomically low.)

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2012/6/5 Martin Thomson <martin.thomson@gmail.com>:
> No misunderstanding. =C2=A0That's expected behaviour. =C2=A0And it's not
> specific to interfaces that can't reach the specified address. =C2=A0Some
> UDP packets just don't make it. =C2=A0(Though in this case, there are
> several sent, so the odds of all of them getting lost is
> astronomically low.)

I hope there will be some way to try ICE candidates in all the
networks addresses/interfaces and, later once we have learned which
interfaces "work", then use just them for ICE/STUN discovery.

--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

From harald@alvestrand.no  Wed Jun  6 01:03:04 2012
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On 06/06/2012 01:41 AM, IÃ±aki Baz Castillo wrote:
> 2012/6/5 Martin Thomson<martin.thomson@gmail.com>:
>> No misunderstanding.  That's expected behaviour.  And it's not
>> specific to interfaces that can't reach the specified address.  Some
>> UDP packets just don't make it.  (Though in this case, there are
>> several sent, so the odds of all of them getting lost is
>> astronomically low.)
> I hope there will be some way to try ICE candidates in all the
> networks addresses/interfaces and, later once we have learned which
> interfaces "work", then use just them for ICE/STUN discovery.
>
That's what ICE/STUN discovery is for.... and once you yank out your 
cable or reconnect your wireless, it all changes anyway, so caching has 
very limited value.

Better to program apps in such a way that you can live with interfaces 
that don't answer.


From harald@alvestrand.no  Wed Jun  6 04:51:48 2012
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On 06/05/2012 03:18 PM, Colin Perkins wrote:
> On 5 Jun 2012, at 07:49, Harald Alvestrand wrote:
>> On 06/04/2012 10:03 PM, Ejzak, Richard P (Richard) wrote:
> ...
>>>   Even though RTP packets with different payload types could potentially reuse the same SSRC value, SSRC values cannot be independently assigned per m line with the current BUNDLE scheme since RTCP packets can be identified only using SSRC values (payload type is not a field in RTCP packets).  So the bundled m lines have to share a single SSRC space, thus increasing the likelihood of SSRC collision
>> Yes. At 302 SSRCs in the same RTP session the chance of a collision is 50%. Applications that don't handle SSRC collisions are broken.
> I agree that applications that don't handle SSRC collisions are broken, but RFC 3550 Section 8.1 gives a vastly lower probability of collision. Is there a mistake in the calculation in the RFC?
>
Yes, the RFC is completely wrong.

Quote 1:

    If N is the number of sources and L the length of the
    identifier (here, 32 bits), the probability that two sources
    independently pick the same value can be approximated for large N
    [26] as 1 - exp(-N**2 / 2**(L+1)).  For N=1000, the probability is
    roughly 10**-4.

I don't have the book quoted, so I can't say where it's picked from. I 
suspect that this may be the probability that for any given two sources, 
they'll pick the same value, but I can't reverse engineer this.

Quote 2:

    Again, if N is the number of sources and L the length of the
    identifier, the probability of collision is N / 2**L.  For N=1000,
    the probability is roughly 2*10**-7.

This is the probability that a collision will occur when one adds SSRC 
number N+1 to the session that already contains N SSRCs that have been 
added without collision.

The exact formula for the chance that *any* collision happens will be 
the birthday paradox formula, which is easy to derive from the one above.



From tterriberry@mozilla.com  Wed Jun  6 07:28:00 2012
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Harald Alvestrand wrote:
> The exact formula for the chance that *any* collision happens will be
> the birthday paradox formula, which is easy to derive from the one above.

This is what I thought, too, but that just makes your 50% number more 
confusing. Plugging 302 and 4294967296 into 
http://jeff.aaron.ca/cgi-bin/birthday gives me 0.001%, not 50% (I got 
the same value when I worked it out myself). But one out of a thousand 
is still a not-insignificant number. The approximation in your Quote 1 
is actually pretty close, but _higher_ than the real value (by 0.3% of 
the real value, or 0.001058% (actual) vs. 0.001062% (approximate).

From tterriberry@mozilla.com  Wed Jun  6 07:38:57 2012
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Timothy B. Terriberry wrote:
> Harald Alvestrand wrote:
>> The exact formula for the chance that *any* collision happens will be
>> the birthday paradox formula, which is easy to derive from the one above.
>
> This is what I thought, too, but that just makes your 50% number more
> confusing. Plugging 302 and 4294967296 into

Also, if I'm working this out correctly, you cross the 50% threshold at 
6563 SSRCs instead of 302.

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2012/6/6 Harald Alvestrand <harald@alvestrand.no>:
> That's what ICE/STUN discovery is for.... and once you yank out your cabl=
e
> or reconnect your wireless, it all changes anyway, so caching has very
> limited value.

Sure, but in my case it means ~3 seconds of delay for every WebRTC
session. Just wondering: it would be "nice" if the SO would notify the
browser when some change in the network settings happen, so the
browser would then invalidate its cache and try again all the
interfaces, and not just those that worked in the previous session.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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On 06/06/2012 04:27 PM, Timothy B. Terriberry wrote:
> Harald Alvestrand wrote:
>> The exact formula for the chance that *any* collision happens will be
>> the birthday paradox formula, which is easy to derive from the one 
>> above.
>
> This is what I thought, too, but that just makes your 50% number more 
> confusing. Plugging 302 and 4294967296 into 
> http://jeff.aaron.ca/cgi-bin/birthday gives me 0.001%, not 50% (I got the 
Argh. That's 4G. I was using 64K. Somehow I'd lost half the SSRC. 
Explains the difference.
My apologies.

I'll crawl under a rock now.

> same value when I worked it out myself). But one out of a thousand is 
> still a not-insignificant number. The approximation in your Quote 1 is 
> actually pretty close, but _higher_ than the real value (by 0.3% of 
> the real value, or 0.001058% (actual) vs. 0.001062% (approximate).
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From g.kiranreddy4u@gmail.com  Wed Jun  6 22:09:16 2012
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Date: Thu, 7 Jun 2012 10:38:54 +0530
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--90e6ba1efe1a8a57d904c1dae235
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Hi all,

I am new to this mailing list.
In draft-sipdoc-rtcweb-open-wire-protocol-00 section 4, for the mandatory
call control section, few fiel names were specified.
Can any one please tell me, in which specification all these filed names
are defined.
Ans also possibility for adding new field names.

Thanks,
Kiran.

--90e6ba1efe1a8a57d904c1dae235
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Hi all,<br>=A0<br>I am new to this mailing list.<br>In draft-sipdoc-rtcweb-=
open-wire-protocol-00 section 4, for the mandatory call control section, fe=
w fiel names were specified.<br>Can any one please tell me, in which specif=
ication all these filed names are defined.<br>

Ans also possibility for adding new field names.<br>=A0<br>Thanks,<br>Kiran=
.

--90e6ba1efe1a8a57d904c1dae235--

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On 6 Jun 2012, at 19:54, I=F1aki Baz Castillo wrote:

> 2012/6/6 Harald Alvestrand <harald@alvestrand.no>:
>> That's what ICE/STUN discovery is for.... and once you yank out your =
cable
>> or reconnect your wireless, it all changes anyway, so caching has =
very
>> limited value.
>=20
> Sure, but in my case it means ~3 seconds of delay for every WebRTC
> session. Just wondering: it would be "nice" if the SO would notify the
> browser when some change in the network settings happen, so the
> browser would then invalidate its cache and try again all the
> interfaces, and not just those that worked in the previous session.

Strikes me that you can get close to that with the current API.=20
In your candidate callback, you  wait untill you see what you used =
last-time and use it,=20
then if that fails you have to do a re-invite with the stuff that =
trickled in later.
(and update the cache).

You are still at the mercy of any upstream network changes (e.g. =
rotating=20
pools of public IPs - mifi switching roaming providers etc) that make =
your cache=20
meaningless.

T.



>=20
>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 7 Jun 2012 12:01:54 +0200
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Subject: Re: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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2012/6/7 Tim Panton <tim@phonefromhere.com>:
>
> On 6 Jun 2012, at 19:54, I=C3=B1aki Baz Castillo wrote:
>> Sure, but in my case it means ~3 seconds of delay for every WebRTC
>> session. Just wondering: it would be "nice" if the SO would notify the
>> browser when some change in the network settings happen, so the
>> browser would then invalidate its cache and try again all the
>> interfaces, and not just those that worked in the previous session.


> Strikes me that you can get close to that with the current API.
> In your candidate callback, you =C2=A0wait untill you see what you used l=
ast-time and use it,
> then if that fails you have to do a re-invite with the stuff that trickle=
d in later.
> (and update the cache).

It seems suitable :)




--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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Date: Thu, 7 Jun 2012 14:02:49 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] Direction to Interim meeting in Kista
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--------------030309080908010900030009
Content-Type: text/plain; charset="ISO-8859-1"
Content-Transfer-Encoding: 8bit

WG,

The Interim meeting will be in the conference room Lappland at Nordic
Forum. The street address is Torshamnsgatan 35 in Kista.

The attached image provides the recommended way to walk from Kista
Sub-way station. The walking directions are the following:

So when arriving at the subway station take the exit towards the rear of
the train if arriving from downtown. From the station you come down the
stairs and end up in the Galleria, turn left and go straight through the
galleria. You exit the galleria onto the end of a square. In front of
you are a street "Kistagången" going uphill. At the far end of the
street there is a high rise sticking up. Go towards it.

You will cross a bridge over a car street. In the next block after the
bridge there is KTH/Electrum on the left. After that building you will
have a walkway /bicycle path "Gröndlandsgången" to the left. Take that
and then right after just 30-40 meters, up some stairs. Then you arrive
on a parking lot, turn left and you get to the entrance. The meeting is
in on the entrance floor.

If you get lost, please call me +46 73 0949079 and I will try to sort
you out.
-- 

Magnus Westerlund

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

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Subject: [rtcweb] Fwd: Invitation: Monday night get-together in Stockholm
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Forwarding as Harald's message appears to not have reached the list.

/Magnus

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DQo=



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On May 30, 2012, at 5:06 , G=F6ran Eriksson AP wrote:

>>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org=20
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings
>> Sent: den 16 maj 2012 03:31
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Comments on=20
>> draft-ietf-rtcweb-use-cases-and-requirements-07
>>=20
>>=20
>> Like to add case where application has one two video streams=20
>> but one is far more important than the other. Should be a way=20
>> to make sure that preferential treatment is given the the=20
>> important stream over the less important streams.=20
>=20
> I agree that this is a relevant area to address to secure that WebRTC =
is competitive compared to
> apps done in native OS's, especiallin in enterprise context.
>=20
> The current use case document states something like "being able to use =
priority functions in network nodes".
> This is vague since it touches many matters such as whether to put =
audio and video on the same IP-flow or have them in different
> ones, which is essential when considering mapping on LTE radio =
channels; the potential use of DS to identify flows, which
> may be relevant in enterprise context, etc, how to cater for treament =
of stuff on the datagram channel, multihoming consequences, etc.
>=20
> I am not sure however how far into such matters we should go in this =
WG, but given the importance of making a
> good solution working across OS's and browser vendors and access =
technologies, I am leaning towards support a discussions about such
> details in this WG for instance using the use case document.
>=20
> What is Your take of this?=20

I think we need to discuss it in the spec. Let's imagine the case where =
we want the brows to set one DSCP for audio and a different one for =
video. I don't think we can have setting the DSCP 100% under control of =
the JS App as that has security problems. But the browser should set the =
DSCP to something and if there are multiple video streams, I think it is =
important to be able to indicate some video streams are less important =
than others - only the JS app knows which one is less important.=20




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=20

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: den 8 juni 2012 14:23
> To: G=F6ran Eriksson AP
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Comments on=20
> draft-ietf-rtcweb-use-cases-and-requirements-07
>=20
>=20
> On May 30, 2012, at 5:06 , G=F6ran Eriksson AP wrote:
>=20
> >>=20
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org
> >> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings
> >> Sent: den 16 maj 2012 03:31
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] Comments on
> >> draft-ietf-rtcweb-use-cases-and-requirements-07
> >>=20
> >>=20
> >> Like to add case where application has one two video=20
> streams but one=20
> >> is far more important than the other. Should be a way to make sure=20
> >> that preferential treatment is given the the important stream over=20
> >> the less important streams.
> >=20
> > I agree that this is a relevant area to address to secure=20
> that WebRTC=20
> > is competitive compared to apps done in native OS's,=20
> especiallin in enterprise context.
> >=20
> > The current use case document states something like "being=20
> able to use priority functions in network nodes".
> > This is vague since it touches many matters such as whether to put=20
> > audio and video on the same IP-flow or have them in different ones,=20
> > which is essential when considering mapping on LTE radio=20
> channels; the potential use of DS to identify flows, which=20
> may be relevant in enterprise context, etc, how to cater for=20
> treament of stuff on the datagram channel, multihoming=20
> consequences, etc.
> >=20
> > I am not sure however how far into such matters we should=20
> go in this=20
> > WG, but given the importance of making a good solution=20
> working across=20
> > OS's and browser vendors and access technologies, I am=20
> leaning towards support a discussions about such details in=20
> this WG for instance using the use case document.
> >=20
> > What is Your take of this?=20
>=20
> I think we need to discuss it in the spec.=20
>Let's imagine the=20
> case where we want the brows to set one DSCP for audio and a=20
> different one for video. I don't think we can have setting=20
> the DSCP 100% under control of the JS App as that has=20
> security problems. But the browser should set the DSCP to=20
> something and if there are multiple video streams, I think it=20
> is important to be able to indicate some video streams are=20
> less important than others - only the JS app knows which one=20
> is less important.=20

Agree that the JS controlling the DSCP marking is an issue- I am more leani=
ng towards
looking to solutions putting the "origin" in control. Also, the server side=
 of the "app" should also be able to in a position to understand the relati=
ve importance on the different
streams/flows in an application. But this is of course up for discussion.

Another set of use cases could be considering the data on the data channel-=
 we have scenarios where these have a significant impact on the UX and it c=
ould therefore be
relevant to consider these flows as well- after all, it may be nice talking=
 to/looking at someone but if we also expect some shared data, I am pretty =
sure we will spend time complaining on the "darn" network more than doing w=
hatever was the original purpose, :-)!

Going forward, I think Your proposal for adding use cases detailing require=
ments in this
area is a very good one and a good, next step. I expect this will be brough=
t up in next week interim meeting for a discussion about how to proceed.


>=20
>=20
>=20
> =

From g.kiranreddy4u@gmail.com  Sat Jun  9 04:26:27 2012
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From: kiran kumar <g.kiranreddy4u@gmail.com>
Date: Sat, 9 Jun 2012 16:56:06 +0530
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Subject: [rtcweb] Reg retransmissions using JSEP approach.
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--e89a8f83a12f2c761404c20863ca
Content-Type: text/plain; charset=ISO-8859-1

Hi all,

I am new to this list, so please don't mind if I am repeating the same
question, if it has already discussed in this list.

My doubt is regarding how JSEP style of implementation will differentiate
the retransmission of messges with out ROAP.

According to few mails and discussion lists, It is specified that ROAP is
going to be replaced by JSEP.
But in updated JSEP draft an example with ROAP is also provided.

If ROAP is going to be replaced by JSEP, which part of JSEP will
differentiate the retransmitted offers.
ROAP is having "seq" field to differentiate it.
Is that part going to be completely left to the application protocol ?


Thanks,
Kiran.

--e89a8f83a12f2c761404c20863ca
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div>Hi all,</div>
<div>=A0</div>
<div>I am new to this list, so please don&#39;t mind if I am repeating the =
same question, if it has already discussed in this list.</div>
<div>=A0</div>
<div>My doubt is regarding how JSEP style of implementation will differenti=
ate the retransmission of messges with out ROAP.</div>
<div>=A0</div>
<div>According to few mails and discussion lists, It is specified that=A0RO=
AP is going to be replaced by JSEP.</div>
<div>But in updated JSEP draft an example with ROAP is also provided.</div>
<div>=A0</div>
<div>If=A0ROAP is going to be replaced by JSEP, which part of JSEP will dif=
ferentiate the retransmitted offers.</div>
<div>ROAP is having &quot;seq&quot; field to differentiate it.</div>
<div>Is that part going to be completely left to the application protocol ?=
</div>
<div>=A0</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Kiran.</div>
<div>=A0</div>

--e89a8f83a12f2c761404c20863ca--

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: kiran kumar <g.kiranreddy4u@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Reg retransmissions using JSEP approach.
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Hi Kiran,

As per JSEP, it is left to the application protocol. The application protoc=
ol shall have "seq" field equivalent to differentiate the retransmitted off=
ers.

Thanks
Partha

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 kiran kumar
Sent: Saturday, June 09, 2012 4:56 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Reg retransmissions using JSEP approach.

Hi all,

I am new to this list, so please don't mind if I am repeating the same ques=
tion, if it has already discussed in this list.

My doubt is regarding how JSEP style of implementation will differentiate t=
he retransmission of messges with out ROAP.

According to few mails and discussion lists, It is specified that ROAP is g=
oing to be replaced by JSEP.
But in updated JSEP draft an example with ROAP is also provided.

If ROAP is going to be replaced by JSEP, which part of JSEP will differenti=
ate the retransmitted offers.
ROAP is having "seq" field to differentiate it.
Is that part going to be completely left to the application protocol ?


Thanks,
Kiran.


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Hi Kiran,<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">As per JSEP, it is left t=
o the application protocol. The application protocol shall have &#8220;seq&=
#8221; field equivalent to differentiate the retransmitted offers.<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Thanks<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Partha<o:p></o:p></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp;</o:p></span><=
/p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-b=
ounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>kiran kumar<br>
<b>Sent:</b> Saturday, June 09, 2012 4:56 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Reg retransmissions using JSEP approach.<o:p></o:p=
></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">Hi all,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">I am new to this list, so please don't mind if I am =
repeating the same question, if it has already discussed in this list.<o:p>=
</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">My doubt is regarding how JSEP style of implementati=
on will differentiate the retransmission of messges with out ROAP.<o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">According to few mails and discussion lists, It is s=
pecified that&nbsp;ROAP is going to be replaced by JSEP.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">But in updated JSEP draft an example with ROAP is al=
so provided.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">If&nbsp;ROAP is going to be replaced by JSEP, which =
part of JSEP will differentiate the retransmitted offers.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">ROAP is having &quot;seq&quot; field to differentiat=
e it.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Is that part going to be completely left to the appl=
ication protocol ?<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Thanks,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">Kiran.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</body>
</html>

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On Jun 9, 2012, at 7:26 , kiran kumar wrote:

> Hi all,
> =20
> I am new to this list, so please don't mind if I am repeating the same =
question, if it has already discussed in this list.
> =20
> My doubt is regarding how JSEP style of implementation will =
differentiate the retransmission of messges with out ROAP.
> =20
> According to few mails and discussion lists, It is specified that ROAP =
is going to be replaced by JSEP.
> But in updated JSEP draft an example with ROAP is also provided.
> =20
> If ROAP is going to be replaced by JSEP, which part of JSEP will =
differentiate the retransmitted offers.
> ROAP is having "seq" field to differentiate it.
> Is that part going to be completely left to the application protocol ?
> =20

Yes, I think the idea i the application would have to deal with that. It =
probably needs to keep track of the keeping between the offer / answers =
to the seq number too so that if an offer is rejected and it rolls back =
to reg previous offer, it has that offer so it can roll back. At the =
same time browser implementation  also need to keep track of the various =
offers so they can roll back. More text is needed in the drafts on all =
this.=20


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To: =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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On Jun 6, 2012, at 14:54 , I=F1aki Baz Castillo wrote:

> 2012/6/6 Harald Alvestrand <harald@alvestrand.no>:
>> That's what ICE/STUN discovery is for.... and once you yank out your =
cable
>> or reconnect your wireless, it all changes anyway, so caching has =
very
>> limited value.
>=20
> Sure, but in my case it means ~3 seconds of delay for every WebRTC
> session. Just wondering: it would be "nice" if the SO would notify the
> browser when some change in the network settings happen, so the
> browser would then invalidate its cache and try again all the
> interfaces, and not just those that worked in the previous session.
>=20

I get that the VPN takes however long to timeout, but given that other =
STUN servers do respond, why are you dealing your whole WebRTC session =
waiting for that stun server. Just ignore it once you get some response =
from another interface.=20


>=20
> --=20
> I=F1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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Hi,

Some comments on draft-ietf-rtcweb-use-cases-and-requirements-07.

Section 4.1.1.1 (Simple Video Communication Service) contains the statement=
 "Each user can also pause sending of media (audio, video, or both) and mut=
e incoming media".  I think this and the associated requirements need some =
clarification for example whether the requirement is to control the directi=
on of the media and therefore requires the generation of a new SDP offer wi=
th for example a direction of "recvonly/inactive".  The only associated req=
uirement seems to be A8 but this does not help clarify what is meant by "pa=
use" in the use case. It needs to be clarified whether the pause/resume/mut=
e are actions within the browser or are actions which result in notificatio=
ns to the remote user.

Maybe we could add a use case which involves controlling the direction of m=
ultiple streams to emulate switching between different users in a multipart=
y call scenario.


The statement "It is essential that the communication cannot be eavesdroppe=
d" appears in all use cases so could be moved to section 4.1 but I think it=
 needs to be reworded in terms of the media always being encrypted as I don=
't like the term "eavesdropped" which is not defined.

Regards
Andy




From ted.ietf@gmail.com  Sun Jun 10 09:48:22 2012
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Date: Sun, 10 Jun 2012 18:48:20 +0200
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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On Sun, Jun 10, 2012 at 5:33 PM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> Hi,
>

> Section 4.1.1.1 (Simple Video Communication Service) contains the stateme=
nt "Each user can also pause sending of media (audio, video, or both) and m=
ute incoming media". =A0I think this and the associated requirements need s=
ome clarification for example whether the requirement is to control the dir=
ection of the media and therefore requires the generation of a new SDP offe=
r with for example a direction of "recvonly/inactive".

Howdy,

As a clarifying question, are you thinking that it is not clear
because non-signaling methods could achieve some of these?  That is
"muting" incoming media could also be accomplished entirely receive
side, by simply not rendering it?

regards,

Ted

=A0The only associated requirement seems to be A8 but this does not help
clarify what is meant by "pause" in the use case. It needs to be
clarified whether the pause/resume/mute are actions within the browser
or are actions which result in notifications to the remote user.
>
> Maybe we could add a use case which involves controlling the direction of=
 multiple streams to emulate switching between different users in a multipa=
rty call scenario.
>
>
> The statement "It is essential that the communication cannot be eavesdrop=
ped" appears in all use cases so could be moved to section 4.1 but I think =
it needs to be reworded in terms of the media always being encrypted as I d=
on't like the term "eavesdropped" which is not defined.
>
> Regards
> Andy
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From andrew.hutton@siemens-enterprise.com  Sun Jun 10 13:03:11 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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Hi Ted,

See below.

Regards
Andy

> -----Original Message-----
> From: Ted Hardie [mailto:ted.ietf@gmail.com]
> Sent: 10 June 2012 18:48
> To: Hutton, Andrew
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-
> requirements-07
>=20
> On Sun, Jun 10, 2012 at 5:33 PM, Hutton, Andrew
> <andrew.hutton@siemens-enterprise.com> wrote:
> > Hi,
> >
>=20
> > Section 4.1.1.1 (Simple Video Communication Service) contains the
> statement "Each user can also pause sending of media (audio, video, or
> both) and mute incoming media". =A0I think this and the associated
> requirements need some clarification for example whether the
> requirement is to control the direction of the media and therefore
> requires the generation of a new SDP offer with for example a direction
> of "recvonly/inactive".
>=20
> Howdy,
>=20
> As a clarifying question, are you thinking that it is not clear
> because non-signaling methods could achieve some of these?  That is
> "muting" incoming media could also be accomplished entirely receive
> side, by simply not rendering it?
>=20

Yes I guess so it is not clear whether the use case results in a local acti=
on in the browser or it whether it impacts the media and SDP. I want to mak=
e sure that it is possible to control the media direction in SDP (sendrecv/=
recvonly/sendonly/inactive) and have the browser behave appropriately.


> regards,
>=20
> Ted
>=20
> =A0The only associated requirement seems to be A8 but this does not help
> clarify what is meant by "pause" in the use case. It needs to be
> clarified whether the pause/resume/mute are actions within the browser
> or are actions which result in notifications to the remote user.
> >
> > Maybe we could add a use case which involves controlling the
> direction of multiple streams to emulate switching between different
> users in a multiparty call scenario.
> >
> >
> > The statement "It is essential that the communication cannot be
> eavesdropped" appears in all use cases so could be moved to section 4.1
> but I think it needs to be reworded in terms of the media always being
> encrypted as I don't like the term "eavesdropped" which is not defined.
> >
> > Regards
> > Andy
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb

From ibc@aliax.net  Sun Jun 10 13:21:20 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Sun, 10 Jun 2012 22:20:58 +0200
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Subject: Re: [rtcweb] STUN tests (ICE candidates) and VPN (session delayed)
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2012/6/10 Cullen Jennings <fluffy@iii.ca>:
> I get that the VPN takes however long to timeout, but given that other ST=
UN servers do respond, why are you dealing your whole WebRTC session waitin=
g for that stun server. Just ignore it once you get some response from anot=
her interface.

That sounds simple and good :)
Thanks.


--=20
I=C3=B1aki Baz Castillo
<ibc@aliax.net>

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An implementation of ROAP on top of JSEP (-00) is available here:

http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep

One of the areas in which it is incomplete is retransmissions....

On 06/10/2012 04:21 PM, Cullen Jennings wrote:
> On Jun 9, 2012, at 7:26 , kiran kumar wrote:
>
>> Hi all,
>>
>> I am new to this list, so please don't mind if I am repeating the same question, if it has already discussed in this list.
>>
>> My doubt is regarding how JSEP style of implementation will differentiate the retransmission of messges with out ROAP.
>>
>> According to few mails and discussion lists, It is specified that ROAP is going to be replaced by JSEP.
>> But in updated JSEP draft an example with ROAP is also provided.
>>
>> If ROAP is going to be replaced by JSEP, which part of JSEP will differentiate the retransmitted offers.
>> ROAP is having "seq" field to differentiate it.
>> Is that part going to be completely left to the application protocol ?
>>
> Yes, I think the idea i the application would have to deal with that. It probably needs to keep track of the keeping between the offer / answers to the seq number too so that if an offer is rejected and it rolls back to reg previous offer, it has that offer so it can roll back. At the same time browser implementation  also need to keep track of the various offers so they can roll back. More text is needed in the drafts on all this.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--------------090304060605000509070702
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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    An implementation of ROAP on top of JSEP (-00) is available here:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep">http://code.google.com/p/webrtc-samples/source/browse/#svn%2Ftrunk%2Froap-jsep</a><br>
    <br>
    One of the areas in which it is incomplete is retransmissions....<br>
    <br>
    On 06/10/2012 04:21 PM, Cullen Jennings wrote:
    <blockquote cite="mid:13D57612-5ADF-4B23-B7A4-81B212D49C66@iii.ca"
      type="cite">
      <pre wrap="">
On Jun 9, 2012, at 7:26 , kiran kumar wrote:

</pre>
      <blockquote type="cite">
        <pre wrap="">Hi all,
 
I am new to this list, so please don't mind if I am repeating the same question, if it has already discussed in this list.
 
My doubt is regarding how JSEP style of implementation will differentiate the retransmission of messges with out ROAP.
 
According to few mails and discussion lists, It is specified that ROAP is going to be replaced by JSEP.
But in updated JSEP draft an example with ROAP is also provided.
 
If ROAP is going to be replaced by JSEP, which part of JSEP will differentiate the retransmitted offers.
ROAP is having "seq" field to differentiate it.
Is that part going to be completely left to the application protocol ?
 
</pre>
      </blockquote>
      <pre wrap="">
Yes, I think the idea i the application would have to deal with that. It probably needs to keep track of the keeping between the offer / answers to the seq number too so that if an offer is rejected and it rolls back to reg previous offer, it has that offer so it can roll back. At the same time browser implementation  also need to keep track of the various offers so they can roll back. More text is needed in the drafts on all this. 

_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 06/10/2012 05:33 PM, Hutton, Andrew wrote:
> Hi,
>
> Some comments on draft-ietf-rtcweb-use-cases-and-requirements-07.
>
> Section 4.1.1.1 (Simple Video Communication Service) contains the statement "Each user can also pause sending of media (audio, video, or both) and mute incoming media".  I think this and the associated requirements need some clarification for example whether the requirement is to control the direction of the media and therefore requires the generation of a new SDP offer with for example a direction of "recvonly/inactive".  The only associated requirement seems to be A8 but this does not help clarify what is meant by "pause" in the use case. It needs to be clarified whether the pause/resume/mute are actions within the browser or are actions which result in notifications to the remote user.
>
> Maybe we could add a use case which involves controlling the direction of multiple streams to emulate switching between different users in a multiparty call scenario.
>
>
> The statement "It is essential that the communication cannot be eavesdropped" appears in all use cases so could be moved to section 4.1 but I think it needs to be reworded in terms of the media always being encrypted as I don't like the term "eavesdropped" which is not defined.
If we want to define the term, we might want to use the term 
"wiretapping", and refer to RFC 2804 for the definition.

(having been involved in that discussion, I've got some personal 
fondness for that definition...)


>
> Regards
> Andy
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: kiran kumar <g.kiranreddy4u@gmail.com>
Date: Mon, 11 Jun 2012 09:59:36 +0530
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Subject: Re: [rtcweb] Reg retransmissions using JSEP approach.
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Thanks for your quick respnse,

Thanks,
Kiran.

On Sun, Jun 10, 2012 at 7:51 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On Jun 9, 2012, at 7:26 , kiran kumar wrote:
>
> > Hi all,
> >
> > I am new to this list, so please don't mind if I am repeating the same
> question, if it has already discussed in this list.
> >
> > My doubt is regarding how JSEP style of implementation will
> differentiate the retransmission of messges with out ROAP.
> >
> > According to few mails and discussion lists, It is specified that ROAP
> is going to be replaced by JSEP.
> > But in updated JSEP draft an example with ROAP is also provided.
> >
> > If ROAP is going to be replaced by JSEP, which part of JSEP will
> differentiate the retransmitted offers.
> > ROAP is having "seq" field to differentiate it.
> > Is that part going to be completely left to the application protocol ?
> >
>
> Yes, I think the idea i the application would have to deal with that. It
> probably needs to keep track of the keeping between the offer / answers to
> the seq number too so that if an offer is rejected and it rolls back to reg
> previous offer, it has that offer so it can roll back. At the same time
> browser implementation  also need to keep track of the various offers so
> they can roll back. More text is needed in the drafts on all this.
>
>

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<div>Thanks for your quick respnse,</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Kiran.<br><br></div>
<div class=3D"gmail_quote">On Sun, Jun 10, 2012 at 7:51 PM, Cullen Jennings=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blank">f=
luffy@iii.ca</a>&gt;</span> wrote:<br>
<blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PA=
DDING-LEFT:1ex" class=3D"gmail_quote">
<div class=3D"HOEnZb">
<div class=3D"h5"><br>On Jun 9, 2012, at 7:26 , kiran kumar wrote:<br><br>&=
gt; Hi all,<br>&gt;<br>&gt; I am new to this list, so please don&#39;t mind=
 if I am repeating the same question, if it has already discussed in this l=
ist.<br>

&gt;<br>&gt; My doubt is regarding how JSEP style of implementation will di=
fferentiate the retransmission of messges with out ROAP.<br>&gt;<br>&gt; Ac=
cording to few mails and discussion lists, It is specified that ROAP is goi=
ng to be replaced by JSEP.<br>

&gt; But in updated JSEP draft an example with ROAP is also provided.<br>&g=
t;<br>&gt; If ROAP is going to be replaced by JSEP, which part of JSEP will=
 differentiate the retransmitted offers.<br>&gt; ROAP is having &quot;seq&q=
uot; field to differentiate it.<br>

&gt; Is that part going to be completely left to the application protocol ?=
<br>&gt;<br><br></div></div>Yes, I think the idea i the application would h=
ave to deal with that. It probably needs to keep track of the keeping betwe=
en the offer / answers to the seq number too so that if an offer is rejecte=
d and it rolls back to reg previous offer, it has that offer so it can roll=
 back. At the same time browser implementation =A0also need to keep track o=
f the various offers so they can roll back. More text is needed in the draf=
ts on all this.<br>

<br></blockquote></div><br>

--14dae93407d75cb7c804c22acd1a--

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From: kiran kumar <g.kiranreddy4u@gmail.com>
Date: Mon, 11 Jun 2012 09:59:59 +0530
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Thanks for your quick response,

Thanks,
Kiran.

On Sun, Jun 10, 2012 at 12:04 AM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

>  Hi Kiran,****
>
> ** **
>
> As per JSEP, it is left to the application protocol. The application
> protocol shall have =93seq=94 field equivalent to differentiate the
> retransmitted offers.****
>
> ** **
>
> Thanks****
>
> Partha****
>
> ** **
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *kiran kumar
> *Sent:* Saturday, June 09, 2012 4:56 PM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Reg retransmissions using JSEP approach.****
>
> ** **
>
> Hi all,****
>
>  ****
>
> I am new to this list, so please don't mind if I am repeating the same
> question, if it has already discussed in this list.****
>
>  ****
>
> My doubt is regarding how JSEP style of implementation will differentiate
> the retransmission of messges with out ROAP.****
>
>  ****
>
> According to few mails and discussion lists, It is specified that ROAP is
> going to be replaced by JSEP.****
>
> But in updated JSEP draft an example with ROAP is also provided.****
>
>  ****
>
> If ROAP is going to be replaced by JSEP, which part of JSEP will
> differentiate the retransmitted offers.****
>
> ROAP is having "seq" field to differentiate it.****
>
> Is that part going to be completely left to the application protocol ?***=
*
>
>  ****
>
>  ****
>
> Thanks,****
>
> Kiran.****
>
>  ****
>

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<div>Thanks for your quick response,</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Kiran.<br><br></div>
<div class=3D"gmail_quote">On Sun, Jun 10, 2012 at 12:04 AM, Ravindran, Par=
thasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusnet.com"=
 target=3D"_blank">pravindran@sonusnet.com</a>&gt;</span> wrote:<br>
<blockquote style=3D"BORDER-LEFT:#ccc 1px solid;MARGIN:0px 0px 0px 0.8ex;PA=
DDING-LEFT:1ex" class=3D"gmail_quote">
<div lang=3D"EN-US" vlink=3D"purple" link=3D"blue">
<div>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt">Hi Kiran,<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt">As per JSEP, it is left to the =
application protocol. The application protocol shall have =93seq=94 field e=
quivalent to differentiate the retransmitted offers.<u></u><u></u></span></=
p>


<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt"><u></u>=A0<u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt">Thanks<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt">Partha<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"FONT-FAMILY:&#39;Calibri&#39;,&#39;sa=
ns-serif&#39;;COLOR:#1f497d;FONT-SIZE:11pt"><u></u>=A0<u></u></span></p>
<div style=3D"BORDER-BOTTOM:medium none;BORDER-LEFT:blue 1.5pt solid;PADDIN=
G-BOTTOM:0in;PADDING-LEFT:4pt;PADDING-RIGHT:0in;BORDER-TOP:medium none;BORD=
ER-RIGHT:medium none;PADDING-TOP:0in">
<div>
<div style=3D"BORDER-BOTTOM:medium none;BORDER-LEFT:medium none;PADDING-BOT=
TOM:0in;PADDING-LEFT:0in;PADDING-RIGHT:0in;BORDER-TOP:#b5c4df 1pt solid;BOR=
DER-RIGHT:medium none;PADDING-TOP:3pt">
<p class=3D"MsoNormal"><b><span style=3D"FONT-FAMILY:&#39;Tahoma&#39;,&#39;=
sans-serif&#39;;FONT-SIZE:10pt">From:</span></b><span style=3D"FONT-FAMILY:=
&#39;Tahoma&#39;,&#39;sans-serif&#39;;FONT-SIZE:10pt"> <a href=3D"mailto:rt=
cweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a> [mailt=
o:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounc=
es@ietf.org</a>] <b>On Behalf Of </b>kiran kumar<br>

<b>Sent:</b> Saturday, June 09, 2012 4:56 PM<br><b>To:</b> <a href=3D"mailt=
o:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br><b>Subject:</b>=
 [rtcweb] Reg retransmissions using JSEP approach.<u></u><u></u></span></p>

</div></div>
<div>
<div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
<div>
<p class=3D"MsoNormal">Hi all,<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">I am new to this list, so please don&#39;t mind if I=
 am repeating the same question, if it has already discussed in this list.<=
u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">My doubt is regarding how JSEP style of implementati=
on will differentiate the retransmission of messges with out ROAP.<u></u><u=
></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">According to few mails and discussion lists, It is s=
pecified that=A0ROAP is going to be replaced by JSEP.<u></u><u></u></p></di=
v>
<div>
<p class=3D"MsoNormal">But in updated JSEP draft an example with ROAP is al=
so provided.<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">If=A0ROAP is going to be replaced by JSEP, which par=
t of JSEP will differentiate the retransmitted offers.<u></u><u></u></p></d=
iv>
<div>
<p class=3D"MsoNormal">ROAP is having &quot;seq&quot; field to differentiat=
e it.<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">Is that part going to be completely left to the appl=
ication protocol ?<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">Thanks,<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">Kiran.<u></u><u></u></p></div>
<div>
<p class=3D"MsoNormal">=A0<u></u><u></u></p></div></div></div></div></div><=
/div></blockquote></div><br>

--14dae93410c3b7e3cb04c22aced9--

From magnus.westerlund@ericsson.com  Mon Jun 11 07:11:54 2012
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Subject: [rtcweb] Remote Participation details
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WG,

For tomorrows interim meeting the WIKI now contains some information
about how to participate remotely, there is two options:

http://trac.tools.ietf.org/wg/rtcweb/trac/wiki/WikiStart

If you have any questions please contact the chairs.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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From ekr@rtfm.com  Tue Jun 12 01:33:22 2012
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From: Eric Rescorla <ekr@rtfm.com>
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Subject: [rtcweb] What I think we need to do about ICE
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To recap what I was saying on the list:

1. Some method of determining when a candidate pair has failed.

2. Some method of failing over to another candidate pair after
failure in as clean a fashion as possible, preferably exploiting
the fact that we already know about other valid candidate pairs,
if possible. Example: I had a direct and relayed pair that worked
and I chose direct. Is there a way to fall back to relayed without
doing a total re-ice.

3. When I discover a new candidate pair, some way of investigating
it and maybe changing to it w/o interfering with my current active
candidate pair.

I think we need to do (1) and probably (2) now and (3) is a nice to have.

-Ekr

From bernard_aboba@hotmail.com  Tue Jun 12 01:53:36 2012
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Subject: Re: [rtcweb] What I think we need to do about ICE
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Issue 1 is addressed via continuing consent, no?=20

WRT 2, ICE prioritizes pairs so there are shortcuts that can be taken, depen=
ding on what you know. If an interface is down, no need to try those host ca=
ndidates. Similarly, no need to retry pairs that just failed. The question w=
ith relay fail back is whether the allocation is still valid.

3. It seems sensible to be able to cache recent tests and not rerun them. If=
 that were done then full offer/answers might only generate delta tests.



On Jun 12, 2012, at 1:33 AM, "Eric Rescorla" <ekr@rtfm.com> wrote:

> To recap what I was saying on the list:
>=20
> 1. Some method of determining when a candidate pair has failed.
>=20
> 2. Some method of failing over to another candidate pair after
> failure in as clean a fashion as possible, preferably exploiting
> the fact that we already know about other valid candidate pairs,
> if possible. Example: I had a direct and relayed pair that worked
> and I chose direct. Is there a way to fall back to relayed without
> doing a total re-ice.
>=20
> 3. When I discover a new candidate pair, some way of investigating
> it and maybe changing to it w/o interfering with my current active
> candidate pair.
>=20
> I think we need to do (1) and probably (2) now and (3) is a nice to have.
>=20
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From michawe@ifi.uio.no  Tue Jun 12 02:06:39 2012
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Hi,

Sorry if I'm asking a dummy question... I'm still somewhat new to 
rtcweb, and certainly quite new to this list:

It seems that the group's idea of data traffic is that this would be 
latency-critical traffic, possibly combined with other real-time 
communication. Slide 2 of the slideset at 
http://trac.tools.ietf.org/wg/rtcweb/trac/attachment/wiki/WikiStart/Non-Media%20Data%20in%20RTCWEB.ppt
says:

Use cases:
-Dynamic meta data for Conference and other real-time services
-Gaming data with low latency requirements
-Others?

This makes me wonder: once we have rtcweb in place, what stops a web 
site developer from just providing a web-based file transfer service 
between browsers? And: if that's a possibility, would we even want to 
prevent that? (I'd say no)

Similarly, I can pull files on my Skype to send e.g. photographs to a 
person I'm chatting with. I'd imagine that this sort of thing would also 
be attractive for rtcweb based applications, and I don't see a reason to 
prevent this.

What's different here is that we're in these cases dealing with greedy, 
non-latency-critical traffic, and possibly lots of it. Could this even 
be a new use case?

Cheers,
Michael


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Michael Welzl wrote:
> What's different here is that we're in these cases dealing with greedy,
> non-latency-critical traffic, and possibly lots of it. Could this even
> be a new use case?

There is currently a "Simple Video Communication Service with file 
exchange" use case (see Section 4.2.8 in
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-08). 
Are there additional derived requirements that are needed to support 
this use case?

From randell-ietf@jesup.org  Tue Jun 12 02:20:53 2012
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On 6/12/2012 5:06 AM, Michael Welzl wrote:

> This makes me wonder: once we have rtcweb in place, what stops a web
> site developer from just providing a web-based file transfer service
> between browsers? And: if that's a possibility, would we even want to
> prevent that? (I'd say no)
>
> Similarly, I can pull files on my Skype to send e.g. photographs to a
> person I'm chatting with. I'd imagine that this sort of thing would also
> be attractive for rtcweb based applications, and I don't see a reason to
> prevent this.

Correct, support for this is a design goal.

> What's different here is that we're in these cases dealing with greedy,
> non-latency-critical traffic, and possibly lots of it. Could this even
> be a new use case?

This is the reason for the priority fields for the DataChannel objects, 
to allow the application to set the relative priorities of media 
channels and data.  By default, we don't want data channels to starve 
media, but to take a share, so are reasonable for "look at this picture" 
cases.  Large i-don'tcare-when transfers should be given somewhat lower 
priorities, real (low-bandwidth) realtime data should be given higher.

-- 
Randell Jesup
randell-ietf@jesup.org

From michawe@ifi.uio.no  Tue Jun 12 02:38:41 2012
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On 6/12/12 11:11 AM, Timothy B. Terriberry wrote:
> Michael Welzl wrote:
>> What's different here is that we're in these cases dealing with greedy,
>> non-latency-critical traffic, and possibly lots of it. Could this even
>> be a new use case?
>
> There is currently a "Simple Video Communication Service with file 
> exchange" use case (see Section 4.2.8 in
> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-08). 
> Are there additional derived requirements that are needed to support 
> this use case?
First: very sorry for missing this one! I looked at this very draft 
again just yesterday, but somehow didn't notice the file transfer bit.
On a side note, when looking through the requirements derived from this 
use case now, I got the impression that it would have been more 
reader-friendly to write:
"In addition to the Simple Video Communication Service use-case, this 
use-case has the following requirements:"  and list just the new ones.

Second: I don't think that there are additional requirements derived 
from that, but it can have architectural implications. (i.e. what is 
derived is, perhaps, a new property).  =>  "file transfer" should then 
be a part of the use cases on the list in the slide set I mentioned.  
(I'm saying that because the slide asks: "others?")

Given that file transfers are "part of the plan", I just think it's 
important for all of us to understand that there may be a lot of data 
traffic too (instead of pictures, people may be sending a movie... or 
you could even have a website offering *nothing but* a file transfer 
service?! I have a feeling that this could even turn into an unintended 
"killer app" for rtcweb  :-)   ).

Cheers,
Michael


From michawe@ifi.uio.no  Tue Jun 12 02:39:26 2012
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On 6/12/12 11:20 AM, Randell Jesup wrote:
> On 6/12/2012 5:06 AM, Michael Welzl wrote:
>
>> This makes me wonder: once we have rtcweb in place, what stops a web
>> site developer from just providing a web-based file transfer service
>> between browsers? And: if that's a possibility, would we even want to
>> prevent that? (I'd say no)
>>
>> Similarly, I can pull files on my Skype to send e.g. photographs to a
>> person I'm chatting with. I'd imagine that this sort of thing would also
>> be attractive for rtcweb based applications, and I don't see a reason to
>> prevent this.
>
> Correct, support for this is a design goal.
>
>> What's different here is that we're in these cases dealing with greedy,
>> non-latency-critical traffic, and possibly lots of it. Could this even
>> be a new use case?
>
> This is the reason for the priority fields for the DataChannel 
> objects, to allow the application to set the relative priorities of 
> media channels and data.  By default, we don't want data channels to 
> starve media, but to take a share, so are reasonable for "look at this 
> picture" cases.  Large i-don'tcare-when transfers should be given 
> somewhat lower priorities, real (low-bandwidth) realtime data should 
> be given higher.
>
I see; thanks. Sounds good to me.

Cheers,
Michael


From harald@alvestrand.no  Tue Jun 12 03:14:10 2012
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Subject: [rtcweb] CNAME as a media stream identifier - what I don't think works
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OK, from the debate today it's clear that the picture is not clear in 
everyone's minds....

with the API that the WG currently has defined, the following is a legal 
set of calls.
So any spec needs to be able to describe the result - even if it is an 
error event at some point.
(Never mind if it is a *wise* set of calls.)

pc = new PeerConnection();

GetUserMedia(..., function(stream) {
      stream1 = stream;
      stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);
}

..... stuff to start connecting ....

pc.addStream(stream1);
pc.addStream(stream2);

offer = pc.createOffer(....);

With MSID, the offer will look like this:

....
m=audio
a=ssrc:1234 msid=xyzzy a0
a=ssrc:1234 msid=xyww a0
...
m=video
a=ssrc:1256 msid=xyzzy v0
a=ssrc:1256 msid=xyww v0

and the expectation on the recipient side is that you will get 
pc.onaddstream called with:

1) stream with id = xyzzy, audio and video
2) stream with id = xyww, audio and video

Cullen, this is for you:
If a mediastream is identified by CNAME, what do you expect to have happen?

                         Harald






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From: Justin Uberti <juberti@google.com>
Date: Tue, 12 Jun 2012 07:36:36 -0400
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] What I think we need to do about ICE
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On Tue, Jun 12, 2012 at 4:53 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

> Issue 1 is addressed via continuing consent, no?
>

Is this frequent enough? We probably need to be able to detect failures in
~5 seconds or so.

>
> WRT 2, ICE prioritizes pairs so there are shortcuts that can be taken,
> depending on what you know. If an interface is down, no need to try those
> host candidates. Similarly, no need to retry pairs that just failed. The
> question with relay fail back is whether the allocation is still valid.
>

How does this work at the wire level? Is this a new USE-CANDIDATE ping on
the new pair from the controlling side?

>
> 3. It seems sensible to be able to cache recent tests and not rerun them.
> If that were done then full offer/answers might only generate delta tests.
>

This assumes that both sides are keeping all unused candidates alive during
the session.

>
>
>
> On Jun 12, 2012, at 1:33 AM, "Eric Rescorla" <ekr@rtfm.com> wrote:
>
> > To recap what I was saying on the list:
> >
> > 1. Some method of determining when a candidate pair has failed.
> >
> > 2. Some method of failing over to another candidate pair after
> > failure in as clean a fashion as possible, preferably exploiting
> > the fact that we already know about other valid candidate pairs,
> > if possible. Example: I had a direct and relayed pair that worked
> > and I chose direct. Is there a way to fall back to relayed without
> > doing a total re-ice.
> >
> > 3. When I discover a new candidate pair, some way of investigating
> > it and maybe changing to it w/o interfering with my current active
> > candidate pair.
> >
> > I think we need to do (1) and probably (2) now and (3) is a nice to have.
> >
> > -Ekr
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<br><br><div class=3D"gmail_quote">On Tue, Jun 12, 2012 at 4:53 AM, Bernard=
 Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">

Issue 1 is addressed via continuing consent, no?<br></blockquote><div><br><=
/div><div>Is this frequent enough? We probably need to be able to detect fa=
ilures in ~5 seconds or so.=C2=A0</div><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<br>
WRT 2, ICE prioritizes pairs so there are shortcuts that can be taken, depe=
nding on what you know. If an interface is down, no need to try those host =
candidates. Similarly, no need to retry pairs that just failed. The questio=
n with relay fail back is whether the allocation is still valid.<br>

</blockquote><div><br></div><div>How does this work at the wire level? Is t=
his a new USE-CANDIDATE ping on the new pair from the controlling side?</di=
v><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">


<br>
3. It seems sensible to be able to cache recent tests and not rerun them. I=
f that were done then full offer/answers might only generate delta tests.<b=
r></blockquote><div><br></div><div>This assumes that both sides are keeping=
 all unused candidates alive during the session.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
<br>
On Jun 12, 2012, at 1:33 AM, &quot;Eric Rescorla&quot; &lt;<a href=3D"mailt=
o:ekr@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
<br>
&gt; To recap what I was saying on the list:<br>
&gt;<br>
&gt; 1. Some method of determining when a candidate pair has failed.<br>
&gt;<br>
&gt; 2. Some method of failing over to another candidate pair after<br>
&gt; failure in as clean a fashion as possible, preferably exploiting<br>
&gt; the fact that we already know about other valid candidate pairs,<br>
&gt; if possible. Example: I had a direct and relayed pair that worked<br>
&gt; and I chose direct. Is there a way to fall back to relayed without<br>
&gt; doing a total re-ice.<br>
&gt;<br>
&gt; 3. When I discover a new candidate pair, some way of investigating<br>
&gt; it and maybe changing to it w/o interfering with my current active<br>
&gt; candidate pair.<br>
&gt;<br>
&gt; I think we need to do (1) and probably (2) now and (3) is a nice to ha=
ve.<br>
&gt;<br>
&gt; -Ekr<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

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From bernard_aboba@hotmail.com  Tue Jun 12 06:57:18 2012
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Subject: Re: [rtcweb] What I think we need to do about ICE
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Justin said:

On Tue=2C Jun 12=2C 2012 at 4:53 AM=2C Bernard Aboba <bernard_aboba@hotmail=
.com> wrote:


Issue 1 is addressed via continuing consent=2C no?

Is this frequent enough? We probably need to be able to detect failures in =
~5 seconds or so.=20

[BA] Depends on the kind of failure.  Removal/addition of an interface can =
be detected very quickly (e.g. RFC 4436 can run in 5 ms).  Detection of a f=
ailure in the middle of the network will take longer=3B route flaps can cau=
se transient loss=2C but often resolve themselves quickly (under 30 seconds=
) so switch in less than 15 seconds will risk false positives.=20





WRT 2=2C ICE prioritizes pairs so there are shortcuts that can be taken=2C =
depending on what you know. If an interface is down=2C no need to try those=
 host candidates. Similarly=2C no need to retry pairs that just failed. The=
 question with relay fail back is whether the allocation is still valid.



How does this work at the wire level? Is this a new USE-CANDIDATE ping on t=
he new pair from the controlling side?

[BA] In the event of a failure=2C it would make sense to be aggressive to r=
estore media quickly. If you have a working pair=2C but are just looking fo=
r a better one=2C you might not want to set flags until you've confirmed th=
at the pair is actually better.=20






3. It seems sensible to be able to cache recent tests and not rerun them. I=
f that were done then full offer/answers might only generate delta tests.

This assumes that both sides are keeping all unused candidates alive during=
 the session.=20

[BA] I doubt that implementations would keep relay candidates alive if they=
 weren't chosen.  But one could cache the list of candidates from both side=
s minus the relays=2C update the host  and server reflexive candidates and =
prioritize pairs.=20
The question is whether you could skip any tests. If a pair has been recent=
ly demonstrated to be working (e.g. it is working currently=2C but you've g=
otten new pairs) you can skip the test.  Similarly=2C if it's been marked a=
s down.=20

=20









On Jun 12=2C 2012=2C at 1:33 AM=2C "Eric Rescorla" <ekr@rtfm.com> wrote:



> To recap what I was saying on the list:

>

> 1. Some method of determining when a candidate pair has failed.

>

> 2. Some method of failing over to another candidate pair after

> failure in as clean a fashion as possible=2C preferably exploiting

> the fact that we already know about other valid candidate pairs=2C

> if possible. Example: I had a direct and relayed pair that worked

> and I chose direct. Is there a way to fall back to relayed without

> doing a total re-ice.

>

> 3. When I discover a new candidate pair=2C some way of investigating

> it and maybe changing to it w/o interfering with my current active

> candidate pair.

>

> I think we need to do (1) and probably (2) now and (3) is a nice to have.

>

> -Ekr

> _______________________________________________

> rtcweb mailing list

> rtcweb@ietf.org

> https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________

rtcweb mailing list

rtcweb@ietf.org

https://www.ietf.org/mailman/listinfo/rtcweb


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Justin said:<br><div><br><div class=3D"ecxgmail_quote">On Tue=2C Jun 12=2C =
2012 at 4:53 AM=2C Bernard Aboba <span dir=3D"ltr">&lt=3B<a href=3D"mailto:=
bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt=3B</span> wrote=
:<br><blockquote class=3D"ecxgmail_quote" style=3D"border-left:1px #ccc sol=
id=3Bpadding-left:1ex">

Issue 1 is addressed via continuing consent=2C no?<br></blockquote><div><br=
></div><div>Is this frequent enough? We probably need to be able to detect =
failures in ~5 seconds or so. <br><br>[BA] Depends on the kind of failure.&=
nbsp=3B Removal/addition of an interface can be detected very quickly (e.g.=
 RFC 4436 can run in 5 ms).&nbsp=3B Detection of a failure in the middle of=
 the network will take longer=3B route flaps can cause transient loss=2C bu=
t often resolve themselves quickly (under 30 seconds) so switch in less tha=
n 15 seconds will risk false positives. <br></div><blockquote class=3D"ecxg=
mail_quote" style=3D"border-left:1px #ccc solid=3Bpadding-left:1ex">


<br>
WRT 2=2C ICE prioritizes pairs so there are shortcuts that can be taken=2C =
depending on what you know. If an interface is down=2C no need to try those=
 host candidates. Similarly=2C no need to retry pairs that just failed. The=
 question with relay fail back is whether the allocation is still valid.<br=
>

</blockquote><div><br></div><div>How does this work at the wire level? Is t=
his a new USE-CANDIDATE ping on the new pair from the controlling side?<br>=
<br>[BA] In the event of a failure=2C it would make sense to be aggressive =
to restore media quickly. If you have a working pair=2C but are just lookin=
g for a better one=2C you might not want to set flags until you've confirme=
d that the pair is actually better. <br><br></div><blockquote class=3D"ecxg=
mail_quote" style=3D"border-left:1px #ccc solid=3Bpadding-left:1ex">


<br>
3. It seems sensible to be able to cache recent tests and not rerun them. I=
f that were done then full offer/answers might only generate delta tests.<b=
r></blockquote><div><br></div><div>This assumes that both sides are keeping=
 all unused candidates alive during the session. <br><br>[BA] I doubt that =
implementations would keep relay candidates alive if they weren't chosen.&n=
bsp=3B But one could cache the list of candidates from both sides minus the=
 relays=2C update the host&nbsp=3B and server reflexive candidates and prio=
ritize pairs. <br>The question is whether you could skip any tests. If a pa=
ir has been recently demonstrated to be working (e.g. it is working current=
ly=2C but you've gotten new pairs) you can skip the test.&nbsp=3B Similarly=
=2C if it's been marked as down. <br><br>&nbsp=3B<br></div>

<blockquote class=3D"ecxgmail_quote" style=3D"border-left:1px #ccc solid=3B=
padding-left:1ex">
<div class=3D"ecxHOEnZb"><div class=3D"h5"><br>
<br>
<br>
On Jun 12=2C 2012=2C at 1:33 AM=2C "Eric Rescorla" &lt=3B<a href=3D"mailto:=
ekr@rtfm.com">ekr@rtfm.com</a>&gt=3B wrote:<br>
<br>
&gt=3B To recap what I was saying on the list:<br>
&gt=3B<br>
&gt=3B 1. Some method of determining when a candidate pair has failed.<br>
&gt=3B<br>
&gt=3B 2. Some method of failing over to another candidate pair after<br>
&gt=3B failure in as clean a fashion as possible=2C preferably exploiting<b=
r>
&gt=3B the fact that we already know about other valid candidate pairs=2C<b=
r>
&gt=3B if possible. Example: I had a direct and relayed pair that worked<br=
>
&gt=3B and I chose direct. Is there a way to fall back to relayed without<b=
r>
&gt=3B doing a total re-ice.<br>
&gt=3B<br>
&gt=3B 3. When I discover a new candidate pair=2C some way of investigating=
<br>
&gt=3B it and maybe changing to it w/o interfering with my current active<b=
r>
&gt=3B candidate pair.<br>
&gt=3B<br>
&gt=3B I think we need to do (1) and probably (2) now and (3) is a nice to =
have.<br>
&gt=3B<br>
&gt=3B -Ekr<br>
&gt=3B _______________________________________________<br>
&gt=3B rtcweb mailing list<br>
&gt=3B <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt=3B <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_=
blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div> 		 	   		  </div></body>
</html>=

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From juberti@google.com  Tue Jun 12 07:55:55 2012
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From: Justin Uberti <juberti@google.com>
Date: Tue, 12 Jun 2012 10:55:32 -0400
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Subject: Re: [rtcweb] What I think we need to do about ICE
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On Tue, Jun 12, 2012 at 9:57 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

>  Justin said:
>
> On Tue, Jun 12, 2012 at 4:53 AM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:
>
> Issue 1 is addressed via continuing consent, no?
>
>
> Is this frequent enough? We probably need to be able to detect failures in
> ~5 seconds or so.
>
> [BA] Depends on the kind of failure.  Removal/addition of an interface can
> be detected very quickly (e.g. RFC 4436 can run in 5 ms).  Detection of a
> failure in the middle of the network will take longer; route flaps can
> cause transient loss, but often resolve themselves quickly (under 30
> seconds) so switch in less than 15 seconds will risk false positives.
>

I understand what you are saying, but most people will hang up the call
after ~10 seconds of silence. Given that the downside of an ICE restart is
minimal, we probably want to be more aggressive here.

>
> WRT 2, ICE prioritizes pairs so there are shortcuts that can be taken,
> depending on what you know. If an interface is down, no need to try those
> host candidates. Similarly, no need to retry pairs that just failed. The
> question with relay fail back is whether the allocation is still valid.
>
>
> How does this work at the wire level? Is this a new USE-CANDIDATE ping on
> the new pair from the controlling side?
>
> [BA] In the event of a failure, it would make sense to be aggressive to
> restore media quickly. If you have a working pair, but are just looking for
> a better one, you might not want to set flags until you've confirmed that
> the pair is actually better.
>

>
> 3. It seems sensible to be able to cache recent tests and not rerun them.
> If that were done then full offer/answers might only generate delta tests.
>
>
> This assumes that both sides are keeping all unused candidates alive
> during the session.
>
> [BA] I doubt that implementations would keep relay candidates alive if
> they weren't chosen.  But one could cache the list of candidates from both
> sides minus the relays, update the host  and server reflexive candidates
> and prioritize pairs.
> The question is whether you could skip any tests. If a pair has been
> recently demonstrated to be working (e.g. it is working currently, but
> you've gotten new pairs) you can skip the test.  Similarly, if it's been
> marked as down.
>
>
>
>
>
>
> On Jun 12, 2012, at 1:33 AM, "Eric Rescorla" <ekr@rtfm.com> wrote:
>
> > To recap what I was saying on the list:
> >
> > 1. Some method of determining when a candidate pair has failed.
> >
> > 2. Some method of failing over to another candidate pair after
> > failure in as clean a fashion as possible, preferably exploiting
> > the fact that we already know about other valid candidate pairs,
> > if possible. Example: I had a direct and relayed pair that worked
> > and I chose direct. Is there a way to fall back to relayed without
> > doing a total re-ice.
> >
> > 3. When I discover a new candidate pair, some way of investigating
> > it and maybe changing to it w/o interfering with my current active
> > candidate pair.
> >
> > I think we need to do (1) and probably (2) now and (3) is a nice to have.
> >
> > -Ekr
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>

--20cf3074d5d6be069e04c247a9c6
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Tue, Jun 12, 2012 at 9:57 AM, Bernard=
 Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" t=
arget=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">




<div><div dir=3D"ltr">
Justin said:<br><div><br><div><div class=3D"im">On Tue, Jun 12, 2012 at 4:5=
3 AM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@h=
otmail.com" target=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrot=
e:<br>

<blockquote style=3D"border-left:1px #ccc solid;padding-left:1ex">

Issue 1 is addressed via continuing consent, no?<br></blockquote><div><br><=
/div></div><div><div class=3D"im">Is this frequent enough? We probably need=
 to be able to detect failures in ~5 seconds or so. <br><br></div>[BA] Depe=
nds on the kind of failure.=C2=A0 Removal/addition of an interface can be d=
etected very quickly (e.g. RFC 4436 can run in 5 ms).=C2=A0 Detection of a =
failure in the middle of the network will take longer; route flaps can caus=
e transient loss, but often resolve themselves quickly (under 30 seconds) s=
o switch in less than 15 seconds will risk false positives. <br>

</div></div></div></div></div></blockquote><div><br></div><div>I understand=
 what you are saying, but most people will hang up the call after ~10 secon=
ds of silence. Given that the downside of an ICE restart is minimal, we pro=
bably want to be more aggressive here.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div><div dir=3D"ltr"><div><div><div></div><=
div class=3D"im"><blockquote style=3D"border-left:1px #ccc solid;padding-le=
ft:1ex">




<br>
WRT 2, ICE prioritizes pairs so there are shortcuts that can be taken, depe=
nding on what you know. If an interface is down, no need to try those host =
candidates. Similarly, no need to retry pairs that just failed. The questio=
n with relay fail back is whether the allocation is still valid.<br>



</blockquote><div><br></div></div><div><div class=3D"im">How does this work=
 at the wire level? Is this a new USE-CANDIDATE ping on the new pair from t=
he controlling side?<br><br></div>[BA] In the event of a failure, it would =
make sense to be aggressive to restore media quickly. If you have a working=
 pair, but are just looking for a better one, you might not want to set fla=
gs until you&#39;ve confirmed that the pair is actually better.=C2=A0</div>

</div></div></div></div></blockquote><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><d=
iv dir=3D"ltr"><div><div><div><br></div><div class=3D"im"><blockquote style=
=3D"border-left:1px #ccc solid;padding-left:1ex">




<br>
3. It seems sensible to be able to cache recent tests and not rerun them. I=
f that were done then full offer/answers might only generate delta tests.<b=
r></blockquote><div><br></div></div><div><div class=3D"im">This assumes tha=
t both sides are keeping all unused candidates alive during the session. <b=
r>

<br></div>[BA] I doubt that implementations would keep relay candidates ali=
ve if they weren&#39;t chosen.=C2=A0 But one could cache the list of candid=
ates from both sides minus the relays, update the host=C2=A0 and server ref=
lexive candidates and prioritize pairs. <br>

The question is whether you could skip any tests. If a pair has been recent=
ly demonstrated to be working (e.g. it is working currently, but you&#39;ve=
 gotten new pairs) you can skip the test.=C2=A0 Similarly, if it&#39;s been=
 marked as down. <br>

<br>=C2=A0<br></div><div class=3D"im">

<blockquote style=3D"border-left:1px #ccc solid;padding-left:1ex">
<div><div><br>
<br>
<br>
On Jun 12, 2012, at 1:33 AM, &quot;Eric Rescorla&quot; &lt;<a href=3D"mailt=
o:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt; wrote:<br>
<br>
&gt; To recap what I was saying on the list:<br>
&gt;<br>
&gt; 1. Some method of determining when a candidate pair has failed.<br>
&gt;<br>
&gt; 2. Some method of failing over to another candidate pair after<br>
&gt; failure in as clean a fashion as possible, preferably exploiting<br>
&gt; the fact that we already know about other valid candidate pairs,<br>
&gt; if possible. Example: I had a direct and relayed pair that worked<br>
&gt; and I chose direct. Is there a way to fall back to relayed without<br>
&gt; doing a total re-ice.<br>
&gt;<br>
&gt; 3. When I discover a new candidate pair, some way of investigating<br>
&gt; it and maybe changing to it w/o interfering with my current active<br>
&gt; candidate pair.<br>
&gt;<br>
&gt; I think we need to do (1) and probably (2) now and (3) is a nice to ha=
ve.<br>
&gt;<br>
&gt; -Ekr<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div></div><br></div> 		 	   		  </div></div>
</blockquote></div><br>

--20cf3074d5d6be069e04c247a9c6--

From anant@mozilla.com  Tue Jun 12 08:07:38 2012
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Date: Tue, 12 Jun 2012 17:07:16 +0200
From: Anant Narayanan <anant@mozilla.com>
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 6/12/12 12:14 PM, Harald Alvestrand wrote:
> pc = new PeerConnection();
>
> GetUserMedia(..., function(stream) {
>       stream1 = stream;

As per my understanding of the gUM spec, this line:

>       stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);

indicates that the audio and video tracks are *copied* into stream2, and 
are not references. Consider, for instance, that if a resolution change 
was made to a track in stream2, the track from the same source in 
stream1 would remain unchanged.

Thus, I'd expect the following:

> ..... stuff to start connecting ....
>
> pc.addStream(stream1);
> pc.addStream(stream2);
>
> offer = pc.createOffer(....);
>
> With MSID, the offer will look like this:
>
> ....
> m=audio
> a=ssrc:1234 msid=xyzzy a0
> a=ssrc:1234 msid=xyww a0
> ...
> m=video
> a=ssrc:1256 msid=xyzzy v0
> a=ssrc:1256 msid=xyww v0

to look like:

m=audio
a=ssrc:1234 cname:144a
a=ssrc:4321 cname:1e24
...
m=video
a=ssrc:1256 cname:144a
a=ssrc:6521 cname:1e24

instead. And, this:

> and the expectation on the recipient side is that you will get
> pc.onaddstream called with:
>
> 1) stream with id = xyzzy, audio and video
> 2) stream with id = xyww, audio and video

would remain the same, except with the IDs substituted by CNAMEs.

-Anant

From bernard_aboba@hotmail.com  Tue Jun 12 08:26:44 2012
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Justin said:

I understand what you are saying=2C but most people will hang up the call a=
fter ~10 seconds of silence. Given that the downside of an ICE restart is m=
inimal=2C we probably want to be more aggressive here.

[BA] If you're asking whether you can start testing alternative pairs quick=
ly=2C sure.  With the built-in rate limiting there won't be any damage done=
 by that=2C and it will take some time to confirm a working alternative pai=
r anyway.

My comment was just about the likelihood that the issue with the existing p=
air is just a temporary glitch.  That would suggest that flags not be set o=
n the alternative exploration until you decide the existing pair is down.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Justin said:<br><div><div class=3D"ecxgmail_quote"><div><br></div><div>I un=
derstand what you are saying=2C but most people will hang up the call after=
 ~10 seconds of silence. Given that the downside of an ICE restart is minim=
al=2C we probably want to be more aggressive here.<br><br>[BA] If you're as=
king whether you can start testing alternative pairs quickly=2C sure.&nbsp=
=3B With the built-in rate limiting there won't be any damage done by that=
=2C and it will take some time to confirm a working alternative pair anyway=
.<br><br>My comment was just about the likelihood that the issue with the e=
xisting pair is just a temporary glitch.&nbsp=3B That would suggest that fl=
ags not be set on the alternative exploration until you decide the existing=
 pair is down. <br></div></div></div> 		 	   		  </div></body>
</html>=

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 06/12/2012 05:07 PM, Anant Narayanan wrote:
> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>> pc = new PeerConnection();
>>
>> GetUserMedia(..., function(stream) {
>>       stream1 = stream;
>
> As per my understanding of the gUM spec, this line:
>
>>       stream2 = new MediaStream(stream1.audioTracks, 
>> stream1.videoTracks);
>
> indicates that the audio and video tracks are *copied* into stream2, 
> and are not references. 
I have concluded from the discussions this week that we're not of one 
mind on this, and that various items, including this, make more sense if 
we see them as references.

The actual language for the constructor is:

   "Addtracktostream'saudio track list."

There's nothing here about making a copy of /track/.

Remember that the original specification (a year ago) had chained tracks 
feeding into other tracks; we simplified this model to a model where a 
track always links a source to a sink, not to another track.

I think that our previous day's discussion has concluded that if we want 
to make a change to a track, and not have that change affect another 
track, we have to make a copy, and we have to make it explicit. For the 
(I think normal) case where we don't manipulate requirements on tracks, 
or are happy to have any changes affect all copies of the track, we 
don't need the copy.

> Consider, for instance, that if a resolution change was made to a 
> track in stream2, the track from the same source in stream1 would 
> remain unchanged.
>
> Thus, I'd expect the following:
>
>> ..... stuff to start connecting ....
>>
>> pc.addStream(stream1);
>> pc.addStream(stream2);
>>
>> offer = pc.createOffer(....);
>>
>> With MSID, the offer will look like this:
>>
>> ....
>> m=audio
>> a=ssrc:1234 msid=xyzzy a0
>> a=ssrc:1234 msid=xyww a0
>> ...
>> m=video
>> a=ssrc:1256 msid=xyzzy v0
>> a=ssrc:1256 msid=xyww v0
>
> to look like:
>
> m=audio
> a=ssrc:1234 cname:144a
> a=ssrc:4321 cname:1e24
> ...
> m=video
> a=ssrc:1256 cname:144a
> a=ssrc:6521 cname:1e24
>
> instead.

This markup (which is actually already standardized) says two things:

- the sender has two synchronization contexts, 144a and 1e24
- the recipient has no information about whether or not these are 
synchronized to the same clock

This also means that the packets of each stream are transmitted twice, 
of course.
> And, this:
>
>> and the expectation on the recipient side is that you will get
>> pc.onaddstream called with:
>>
>> 1) stream with id = xyzzy, audio and video
>> 2) stream with id = xyww, audio and video
>
> would remain the same, except with the IDs substituted by CNAMEs.


That's a reasonable interpretation. Is it the one we want to make?




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  <body bgcolor="#FFFFFF" text="#000000">
    On 06/12/2012 05:07 PM, Anant Narayanan wrote:
    <blockquote cite="mid:4FD75B24.6020501@mozilla.com" type="cite">On
      6/12/12 12:14 PM, Harald Alvestrand wrote:
      <br>
      <blockquote type="cite">pc = new PeerConnection();
        <br>
        <br>
        GetUserMedia(..., function(stream) {
        <br>
        &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; stream1 = stream;
        <br>
      </blockquote>
      <br>
      As per my understanding of the gUM spec, this line:
      <br>
      <br>
      <blockquote type="cite">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; stream2 = new
        MediaStream(stream1.audioTracks, stream1.videoTracks);
        <br>
      </blockquote>
      <br>
      indicates that the audio and video tracks are *copied* into
      stream2, and are not references. </blockquote>
    I have concluded from the discussions this week that we're not of
    one mind on this, and that various items, including this, make more
    sense if we see them as references.<br>
    <br>
    The actual language for the constructor is:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <span style="color: rgb(0, 0, 0); font-family: sans-serif;
      font-size: medium; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-align: -webkit-auto; text-indent: 0px;
      text-transform: none; white-space: normal; widows: 2;
      word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none; ">&nbsp; "Add<span
        class="Apple-converted-space">&nbsp;</span></span><var style="color:
      rgb(0, 0, 0); font-family: sans-serif; font-size: medium;
      font-variant: normal; font-weight: normal; letter-spacing: normal;
      line-height: normal; orphans: 2; text-align: -webkit-auto;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); ">track</var><span style="color: rgb(0, 0, 0); font-family:
      sans-serif; font-size: medium; font-style: normal; font-variant:
      normal; font-weight: normal; letter-spacing: normal; line-height:
      normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px;
      text-transform: none; white-space: normal; widows: 2;
      word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none; "><span
        class="Apple-converted-space">&nbsp;</span>to<span
        class="Apple-converted-space">&nbsp;</span></span><var style="color:
      rgb(0, 0, 0); font-family: sans-serif; font-size: medium;
      font-variant: normal; font-weight: normal; letter-spacing: normal;
      line-height: normal; orphans: 2; text-align: -webkit-auto;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); ">stream&#8217;s</var><span style="color: rgb(0, 0, 0);
      font-family: sans-serif; font-size: medium; font-style: normal;
      font-variant: normal; font-weight: normal; letter-spacing: normal;
      line-height: normal; orphans: 2; text-align: -webkit-auto;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: 2; word-spacing: 0px; -webkit-text-size-adjust: auto;
      -webkit-text-stroke-width: 0px; background-color: rgb(255, 255,
      255); display: inline !important; float: none; "><span
        class="Apple-converted-space">&nbsp;</span>audio track list."<br>
      <br>
      There's nothing here about making a copy of <i>track</i>.<br>
      <br>
      Remember that the original specification (a year ago) had chained
      tracks feeding into other tracks; we simplified this model to a
      model where a track always links a source to a sink, not to
      another track.<br>
      <br>
      I think that our previous day's discussion has concluded that if
      we want to make a change to a track, and not have that change
      affect another track, we have to make a copy, and we have to make
      it explicit. For the (I think normal) case where we don't
      manipulate requirements on tracks, or are happy to have any
      changes affect all copies of the track, we don't need the copy.<br>
      <br>
    </span>
    <blockquote cite="mid:4FD75B24.6020501@mozilla.com" type="cite">Consider,
      for instance, that if a resolution change was made to a track in
      stream2, the track from the same source in stream1 would remain
      unchanged.
      <br>
      <br>
      Thus, I'd expect the following:
      <br>
      <br>
      <blockquote type="cite">..... stuff to start connecting ....
        <br>
        <br>
        pc.addStream(stream1);
        <br>
        pc.addStream(stream2);
        <br>
        <br>
        offer = pc.createOffer(....);
        <br>
        <br>
        With MSID, the offer will look like this:
        <br>
        <br>
        ....
        <br>
        m=audio
        <br>
        a=ssrc:1234 msid=xyzzy a0
        <br>
        a=ssrc:1234 msid=xyww a0
        <br>
        ...
        <br>
        m=video
        <br>
        a=ssrc:1256 msid=xyzzy v0
        <br>
        a=ssrc:1256 msid=xyww v0
        <br>
      </blockquote>
      <br>
      to look like:
      <br>
      <br>
      m=audio
      <br>
      a=ssrc:1234 cname:144a
      <br>
      a=ssrc:4321 cname:1e24
      <br>
      ...
      <br>
      m=video
      <br>
      a=ssrc:1256 cname:144a
      <br>
      a=ssrc:6521 cname:1e24
      <br>
      <br>
      instead.</blockquote>
    <br>
    This markup (which is actually already standardized) says two
    things:<br>
    <br>
    - the sender has two synchronization contexts, 144a and 1e24<br>
    - the recipient has no information about whether or not these are
    synchronized to the same clock<br>
    <br>
    This also means that the packets of each stream are transmitted
    twice, of course.<br>
    <blockquote cite="mid:4FD75B24.6020501@mozilla.com" type="cite">
      And, this:
      <br>
      <br>
      <blockquote type="cite">and the expectation on the recipient side
        is that you will get
        <br>
        pc.onaddstream called with:
        <br>
        <br>
        1) stream with id = xyzzy, audio and video
        <br>
        2) stream with id = xyww, audio and video
        <br>
      </blockquote>
      <br>
      would remain the same, except with the IDs substituted by CNAMEs.
      <br>
    </blockquote>
    <br>
    <br>
    That's a reasonable interpretation. Is it the one we want to make?<br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------070206030204020308080009--

From stefan.lk.hakansson@ericsson.com  Tue Jun 12 12:55:46 2012
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 06/12/2012 05:26 PM, Harald Alvestrand wrote:
> On 06/12/2012 05:07 PM, Anant Narayanan wrote:
>> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>>> pc = new PeerConnection();
>>>
>>> GetUserMedia(..., function(stream) {
>>> stream1 = stream;
>>
>> As per my understanding of the gUM spec, this line:
>>
>>> stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);
>>
>> indicates that the audio and video tracks are *copied* into stream2,
>> and are not references.
> I have concluded from the discussions this week that we're not of one
> mind on this, and that various items, including this, make more sense if
> we see them as references.

They would have to be separate enough to allow individual enable/disable 
in any case.

From stefan.lk.hakansson@ericsson.com  Tue Jun 12 13:38:00 2012
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 06/12/2012 05:07 PM, Anant Narayanan wrote:
> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>> pc = new PeerConnection();
>>
>> GetUserMedia(..., function(stream) {
>>        stream1 = stream;
>
> As per my understanding of the gUM spec, this line:
>
>>        stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);
>
> indicates that the audio and video tracks are *copied* into stream2, and
> are not references. Consider, for instance, that if a resolution change
> was made to a track in stream2, the track from the same source in
> stream1 would remain unchanged.

This is a bit unclear to me. What would a resolution change really mean 
in this case? To me the resolution (when a camera is used as source) 
would be the resolution delivered by the camera. Can it deliver two (or 
more) resolutions at the same time (perhaps also with different frame 
rates, aspect rations and so on)?

To me it makes most sense (if the application is going to explicitly ask 
for specific resolution, frame rate, aspec ration) to ask for it at 
getUserMedia time. If the app does not ask for this explicitly, at least 
the suitable resolution would be known by the browser whenever the video 
is played in a video element (from the size of it).



From andrew.hutton@siemens-enterprise.com  Tue Jun 12 13:39:38 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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References: <5A2F246E-9D77-4B72-B3F1-681B00FA99FD@cisco.com> <F3D4ABD6AB47084B84337CF4F3446A464B2F11B52E@ESESSCMS0362.eemea.ericsson.se> <AAB6A6B6-BEC7-436C-B1EB-2C117596C7C6@cisco.com> <F3D4ABD6AB47084B84337CF4F3446A464B2F2631AC@ESESSCMS0362.eemea.ericsson.se> <9F33F40F6F2CD847824537F3C4E37DDF021B53@MCHP04MSX.global-ad.net> <4FD516B3.40501@alvestrand.no>
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Subject: Re: [rtcweb] More Comments on	draft-ietf-rtcweb-use-cases-and-requirements-07
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Hi,

My suggestion would be to remove the statements from each use case regardin=
g eavesdropping and replace it with bullets in section 4.1 for consideratio=
ns which apply to all use cases. These should state that all media streams =
and data channels must be integrity and confidentiality protected.

Regards
Andy


> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Harald Alvestrand
> Sent: 10 June 2012 23:51
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-
> requirements-07
>=20
> On 06/10/2012 05:33 PM, Hutton, Andrew wrote:
> > Hi,
> >
> > Some comments on draft-ietf-rtcweb-use-cases-and-requirements-07.
> >
> > Section 4.1.1.1 (Simple Video Communication Service) contains the
> statement "Each user can also pause sending of media (audio, video, or
> both) and mute incoming media".  I think this and the associated
> requirements need some clarification for example whether the
> requirement is to control the direction of the media and therefore
> requires the generation of a new SDP offer with for example a direction
> of "recvonly/inactive".  The only associated requirement seems to be A8
> but this does not help clarify what is meant by "pause" in the use
> case. It needs to be clarified whether the pause/resume/mute are
> actions within the browser or are actions which result in notifications
> to the remote user.
> >
> > Maybe we could add a use case which involves controlling the
> direction of multiple streams to emulate switching between different
> users in a multiparty call scenario.
> >
> >
> > The statement "It is essential that the communication cannot be
> eavesdropped" appears in all use cases so could be moved to section 4.1
> but I think it needs to be reworded in terms of the media always being
> encrypted as I don't like the term "eavesdropped" which is not defined.
> If we want to define the term, we might want to use the term
> "wiretapping", and refer to RFC 2804 for the definition.
>=20
> (having been involved in that discussion, I've got some personal
> fondness for that definition...)
>=20
>=20
> >
> > Regards
> > Andy
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From Markus.Isomaki@nokia.com  Tue Jun 12 13:54:12 2012
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From: <Markus.Isomaki@nokia.com>
To: <rtcweb@ietf.org>
Thread-Topic: Interface switching use case and the basic needs
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Subject: [rtcweb] Interface switching use case and the basic needs
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 Hi,

Here is some input on the interface switching use case from mobile device p=
erspective:

Most today's mobile devices (smartphones, tablets) work so, that only a sin=
gle interface is connected at a time. This is for power conservation reason=
s, plus no-one has figured out what to do with the second active interface.=
 In practice the two interfaces are always cellular (of some sort) and Wi-F=
i. A typical policy is that any Wi-Fi is preferred over cellular, but in th=
eory some Wi-Fi networks could have a lower priority too. If a higher prior=
ity interface (network) becomes available, the device connects to it, and a=
fter that, switches off the lower priority one. When the preferred network =
becomes unavailable, the device tries to connect to the second best option.=
=20

Applications usually have no control on this. For instance, if the user is =
having a VoIP call over cellular and walks to Wi-Fi coverage, interface swi=
thing is done (after the Wi-Fi is discovered, which may take a few minutes =
depending on the scanning interval), and the call is lost (unless the app u=
ses some magic to keep it up). Depending on the environments, apps usually =
listen to events indicating interfaces going up and down. Many apps today a=
re capable of dealing with this. For instance apps that have persistent TCP=
 connections simply re-establish them over the new interface. Depending on =
the app protocol they will also have to do things like re-register (SIP) on=
 top of that. If there's nothing time critical going on, the user won't not=
ice anything. In the worst case the other end is just sending something ove=
r TCP when the connection is abandoned and that requires some app layer rec=
overy.=20

I don't know how well browsers and web apps in general deal with this. HTTP=
 is easy to deal with, as its state is not tied to e.g. a particular TCP co=
nnection. Pending requests, for instance hanging GETs, can be re-issued ove=
r a new TCP connection. How about Websocket protocol? Is it possible to sim=
ply re-establisth its TCP connection without the JS app using it noticing? =
I suppose so, but also the recovery of lost data has to be dealt with. The =
main question is whether the JS app would need to know/react in any way. In=
 WebRTC the real-time sessions would often be "signaled" over Websockets or=
 hanging HTTP GETs, so making that part recover smoothly from underlying TC=
P connections getting dropped would be essential.

The real-time media recovery/update would be a great feature, but let's kee=
p in mind also the more basic needs. Can they be dealt with a smart browser=
 implementation, or is there a need to expose something to the JS app too?

Markus



From martin.thomson@gmail.com  Tue Jun 12 15:52:05 2012
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Subject: Re: [rtcweb] Interface switching use case and the basic needs
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On 12 June 2012 13:54,  <Markus.Isomaki@nokia.com> wrote:
> Most today's mobile devices (smartphones, tablets) work so, that only a single interface is connected at a time. This is for

BTW, this one interface at a time behaviour is really annoying in some
situations.  WiFi networks frequently require sign-on, which means
that you migrate to the WiFi and you spend ages mucking around with
the login, or you have to turn off WiFi in order to reassociate with
the MME.  Either way, you have a significant down time window.

This happened to me at SFO on the way over, when I realized that my
"mobile boarding pass" was necessary to enter the security line.  It
took a couple of minutes to resolve.  This is a bad user experience as
a result of this sort of rule, applied blindly.

 > How about Websocket protocol?

That doesn't work.  The application is forced to recover.  Most of the
reasons for using websockets involve tying some sort of state to the
connection.  Sure, that's not robust, but it is easier to fail at that
point than spend the effort dealing with either statelessness (for
which you are using HTTP anyway) or state recovery, which is expensive
to implement.

> The real-time media recovery/update would be a great feature, but let's keep in mind also the more basic needs. Can they be dealt with a smart browser implementation, or is there a need to expose something to the JS app too?

Sure, the JS app gets a disconnect from websockets.  It had better
deal.  The fact that you have had a switchover does require
effectively a complete reboot of ICE, which a realtime app ultimately
needs to deal with by signaling candidates...at a minimum.

This is covered by what EKR sent out and I'm sure that he will have a
proposal for the API around this.  If he doesn't do that soon, I'll
propose something he wont like.

--Martin

From tterriberry@mozilla.com  Tue Jun 12 22:39:42 2012
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Subject: [rtcweb] RTP clock rates
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In discussion at the interim meeting yesterday, Harald expressed a 
desire to prohibit a sender from changing the Payload Type associated 
with an SSRC to one that runs at a different clock rate. This raised the 
question of what clock rates are supported for various codecs where we 
might want to switch, specifically Opus and DTMF.

draft-spittka-payload-rtp-opus-00 specifies that the clock rate for Opus 
will always be 48 kHz. According to RFC 4733, DTMF supports any integer 
clock rate. The "duration" field for a DTMF telephone event is a 16-bit 
unsigned integer, giving a maximum duration of 1.365 seconds per packet. 
Events can be extended beyond this by splitting them into segments and 
sending a separate packet for each segment (using the 'E' bit to mark 
the last one).

So I don't think this prohibition imposes any serious limitations, at 
least with regards to Opus and DTMF.

From harald@alvestrand.no  Wed Jun 13 00:23:01 2012
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Subject: Re: [rtcweb] RTP clock rates
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On 06/13/2012 07:39 AM, Timothy B. Terriberry wrote:
> In discussion at the interim meeting yesterday, Harald expressed a 
> desire to prohibit a sender from changing the Payload Type associated 
> with an SSRC to one that runs at a different clock rate.
To be more precise: I suggested that we should incorporate by reference 
the advice in draft-ietf-avtext-multiple-clock-rates-05:

4.1.  RTP Sender (with RTCP)

    An RTP Sender with RTCP turned on MUST use a different SSRC for each
    different clock rate.  An RTCP BYE MUST be sent and a new SSRC MUST
    be used if the clock rate switches back to a value already seen in
    the RTP stream.

> This raised the question of what clock rates are supported for various 
> codecs where we might want to switch, specifically Opus and DTMF.
>
> draft-spittka-payload-rtp-opus-00 specifies that the clock rate for 
> Opus will always be 48 kHz. According to RFC 4733, DTMF supports any 
> integer clock rate. The "duration" field for a DTMF telephone event is 
> a 16-bit unsigned integer, giving a maximum duration of 1.365 seconds 
> per packet. Events can be extended beyond this by splitting them into 
> segments and sending a separate packet for each segment (using the 'E' 
> bit to mark the last one).
>
> So I don't think this prohibition imposes any serious limitations, at 
> least with regards to Opus and DTMF.

This of course means that if one wants to use DTMF with multiple codecs 
that have different clock rates, there must be multiple payload types 
that identify as DTMF - one for each clock rate.

The same issue arises with CN (Comfort Noise).

Not a problem, as long as implementors are aware, I think.

                               Harald


From pravindran@sonusnet.com  Wed Jun 13 00:43:00 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Cloning  mechanism in JS
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Date: Wed, 13 Jun 2012 07:42:52 +0000
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Subject: [rtcweb] Cloning  mechanism in JS
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Hi all,

I'm writing this mail, as I don't have facility to write in the Jabber room=
.

Cloning mechanism is 1:1 mapping with SIP forking (serial & parallel). The =
cloned peerconnection aggregation is managed by JavaScript and which peerco=
nnection get priority is matter of JS decision.

Each cloned peerconnection will have the initial offer. The number of cloni=
ng shall be decided by the application. As a simple thumb rule, before ever=
y the (early) answer, cloning of the peerconnection shall be done till the =
application wants it. This model helps for other RTP topologies like RTP mi=
xer in WebRTC in the future.

In a way, PRANSWER state itself is not required.

Thanks
Partha=20

From harald@alvestrand.no  Wed Jun 13 01:16:42 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Question about usage of data traffic
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The reason for the initial data use cases was that we wanted to focus on 
the stuff that *couldn't* be done by bouncing data off a server; 
non-time-sensitive bulk file transfer can certainly be done via a 
server, even though that may not be optimal.

That said, we've always expected that once the data transfer capability 
is in place, we'll have plenty of people wanting to use it in 
"unexpected" ways.
The important RTCWEB-specific requirement from the file transfer use 
case is probably that IF file transfer is done in a peer-to-peer 
fashion, within the RTCWEB context, it MUST be able to run without 
degrading the quality of the real time media.

On 06/12/2012 11:38 AM, Michael Welzl wrote:
> On 6/12/12 11:11 AM, Timothy B. Terriberry wrote:
>> Michael Welzl wrote:
>>> What's different here is that we're in these cases dealing with greedy,
>>> non-latency-critical traffic, and possibly lots of it. Could this even
>>> be a new use case?
>>
>> There is currently a "Simple Video Communication Service with file 
>> exchange" use case (see Section 4.2.8 in
>> http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-08). 
>> Are there additional derived requirements that are needed to support 
>> this use case?
> First: very sorry for missing this one! I looked at this very draft 
> again just yesterday, but somehow didn't notice the file transfer bit.
> On a side note, when looking through the requirements derived from 
> this use case now, I got the impression that it would have been more 
> reader-friendly to write:
> "In addition to the Simple Video Communication Service use-case, this 
> use-case has the following requirements:"  and list just the new ones.
>
> Second: I don't think that there are additional requirements derived 
> from that, but it can have architectural implications. (i.e. what is 
> derived is, perhaps, a new property).  =>  "file transfer" should then 
> be a part of the use cases on the list in the slide set I mentioned.  
> (I'm saying that because the slide asks: "others?")
>
> Given that file transfers are "part of the plan", I just think it's 
> important for all of us to understand that there may be a lot of data 
> traffic too (instead of pictures, people may be sending a movie... or 
> you could even have a website offering *nothing but* a file transfer 
> service?! I have a feeling that this could even turn into an 
> unintended "killer app" for rtcweb  :-)   ).
>
> Cheers,
> Michael
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From martin.thomson@gmail.com  Wed Jun 13 01:25:38 2012
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Subject: [rtcweb] Cloning === bad
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I think that there is a bad assumption at play in this proposal that I
think is important to highlight.

That assumption is, that SIP is the only mechanism by which we are
signaling. This is already a problem because it is possible to enter a
deadlock state in the offer answer state machine that requires
application-level intelligence to address.

From harald@alvestrand.no  Wed Jun 13 02:26:05 2012
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On 06/13/2012 10:25 AM, Martin Thomson wrote:
> I think that there is a bad assumption at play in this proposal that I
> think is important to highlight.
>
> That assumption is, that SIP is the only mechanism by which we are
> signaling. This is already a problem because it is possible to enter a
> deadlock state in the offer answer state machine that requires
> application-level intelligence to address.
I don't believe there's any such assumption here. Most of the demos and 
applications that have been shown on top of WebRTC don't have a single 
line referencing SIP in them.

The assumption I hear made (which I also have some doubts about) is that 
it *must* be *possible* to establish interworking with all the variants 
of functionality done by equipment using the SIP prtocol for signalling 
- that is, the answer to the question of "what do I do in the case that 
SIP uses parallel forking for" must have an answer.

Today's poll result on forking seems to be "in version 1.0, we'll take 
the same action as SIP devices that don't support parallel forking" - ie 
"make it someone else's problem".

> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From ekr@rtfm.com  Wed Jun 13 03:01:14 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 13 Jun 2012 03:00:33 -0700
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Subject: [rtcweb] Fast restart problem statement
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I feel like there are actually three different problem statements:

* Web-only invisible restart: the ability for a pair of web-based applications
to re-start a call upon re-load in a user-invisible way, i.e., without prompting
the user to answer the call, authorize media, etc.

* Web-only fast restart: Web-only invisible restart but optimized to be
faster than an initial call.

* Interoperable invisible restart: Like Web-only invisible restart but where
one side is RTCWEB and the other side is an unmodified SIP/Jingle
device.

* Interoperable fast restart: like Interoperable invisible restart but optimized
to be faster than an initial call.

-Ekr

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From: <Markus.Isomaki@nokia.com>
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Thread-Topic: [rtcweb] Interface switching use case and the basic needs
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Subject: Re: [rtcweb] Interface switching use case and the basic needs
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From Markus.Isomaki@nokia.com  Wed Jun 13 03:09:08 2012
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From: <Markus.Isomaki@nokia.com>
To: <lishitao@huawei.com>, <rtcweb@ietf.org>
Thread-Topic: Interface switching use case and the basic needs
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Subject: Re: [rtcweb] Interface switching use case and the basic needs
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lishitao@huawei.com wrote:
>
>So my question is, does this Interface switching use case support dual rad=
io or
>not?
>
>If it support, we can have a solution based on the assumption that the
>browser will receive two access info, and both can be used, or If it does =
not
>support, I don't think the access transfer delay can be tolerated by the u=
ser.
>Even we have some solution in the JS and browser layer, the user may switc=
h
>off the call before the call has recovered.
>

Currently the interface switching in most OS's happens in hard-handover way=
, i.e. the two interfaces are not in use at the same time. Even if the two =
radios were up at the same time for a while, I believe on the IP level the =
default route points only to a single interface at a time. So the browsers =
and web apps should assume break-before-make handovers do happen and try to=
 deal with them.

Having said that, it is indeed a good idea to keep the old interface up at =
least for a while after the new one is activated. That gives protocols (lik=
e MPTCP) and apps at least the chance to do a make-before-break transition.=
 There is no problem keeping up Wi-Fi and cellular interfaces at the same t=
ime from the radio perspective. So, this is another scenario that browsers =
and web apps (and the JS APIs) should support, in case they run on a nice p=
latform able to offer it them, and the network coverage also makes it possi=
ble. I don't know how complex events that APIs can convey, but this would b=
e something like: "A new higher priority interface is available. The lower-=
priority interface is still up for a while, but you'd better move your traf=
fic/state to the new one, as the old one may be gone at any point."=20

Markus=20

From pravindran@sonusnet.com  Wed Jun 13 03:46:22 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: Back-to-back RTP endpoint topology in RTCWeb
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Hi Magnus & all,

I like the list of RTP topology mentioned in your interim presentation. I t=
hought of asking for one more topology namely Back-to-back RTP endpoint top=
ology in RTCWeb. The topology is as follows:

     RTP endpoint-----Back-to-back RTP Endpoint-----RTP Endpoint

The reason for Back-to-back RTP Endpoint topology are

1) WebRTC RTP Endpoint to legacy RTP Endpoint (RFC 3550) interop
2) IP address hiding in WebRTC session

Even though, Back-to-back RTP Endpoint *MUST* follow RTP Endpoint behavior,=
 the requirement will be bit more than that as the linkage with identity, I=
CE consent, RTCP mandate, set of RTP draft mandates and other stuffs. Also,=
 the guidelines document for back-to-back RTP Endpoint shall avoid the blam=
e of intermediate devices always break the architecture.=20

I'm interested in hearing whether Back-to-back RTP endpoint topology is of =
interest in RTCWeb.

Thanks
Partha

From keith.drage@alcatel-lucent.com  Wed Jun 13 04:34:12 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 13 Jun 2012 13:33:56 +0200
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Quick question.

In this document are "data stream" and "datagram stream" one and the same t=
hing, or not?

If yes can we harmonise the term?

If no, can we define what the distinct terms mean?

Keith

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Subject: Re: [rtcweb] Fast restart problem statement
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On 6/13/12 12:00 PM, Eric Rescorla wrote:
> I feel like there are actually three different problem statements:

It looks like you listed four :)

> * Web-only invisible restart: the ability for a pair of web-based applications
> to re-start a call upon re-load in a user-invisible way, i.e., without prompting
> the user to answer the call, authorize media, etc.
>
> * Web-only fast restart: Web-only invisible restart but optimized to be
> faster than an initial call.

Why wouldn't we always make invisible restarts as fast as possible?

-Anant

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Date: Wed, 13 Jun 2012 13:45:50 +0200
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Markus.Isomaki@nokia.com
> Sent: den 13 juni 2012 12:09
> To: lishitao@huawei.com; rtcweb@ietf.org
> Subject: Re: [rtcweb] Interface switching use case and the basic needs
>=20
>=20
> lishitao@huawei.com wrote:
> >
> >So my question is, does this Interface switching use case=20
> support dual=20
> >radio or not?
> >
> >If it support, we can have a solution based on the=20
> assumption that the=20
> >browser will receive two access info, and both can be used, or If it=20
> >does not support, I don't think the access transfer delay=20
> can be tolerated by the user.
> >Even we have some solution in the JS and browser layer, the user may=20
> >switch off the call before the call has recovered.
> >
>=20
> Currently the interface switching in most OS's happens in=20
> hard-handover way, i.e. the two interfaces are not in use at=20
> the same time. Even if the two radios were up at the same=20
> time for a while, I believe on the IP level the default route=20
> points only to a single interface at a time. So the browsers=20
> and web apps should assume break-before-make handovers do=20
> happen and try to deal with them.
>=20
> Having said that, it is indeed a good idea to keep the old=20
> interface up at least for a while after the new one is=20
> activated. That gives protocols (like MPTCP) and apps at=20
> least the chance to do a make-before-break transition. There=20
> is no problem keeping up Wi-Fi and cellular interfaces at the=20
> same time from the radio perspective. So, this is another=20
> scenario that browsers and web apps (and the JS APIs) should=20
> support, in case they run on a nice platform able to offer it=20
> them, and the network coverage also makes it possible. I=20
> don't know how complex events that APIs can convey, but this=20
> would be something like: "A new higher priority interface is=20
> available. The lower-priority interface is still up for a=20
> while, but you'd better move your traffic/state to the new=20
> one, as the old one may be gone at any point."=20

Who is setting the "priority" of the interface? Should not the web app have=
 a say whether the traffic
should be kept on a particular interface or not?


>=20
> Markus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From ekr@rtfm.com  Wed Jun 13 04:46:11 2012
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On Wed, Jun 13, 2012 at 4:43 AM, Anant Narayanan <anant@mozilla.com> wrote:
> On 6/13/12 12:00 PM, Eric Rescorla wrote:
>>
>> I feel like there are actually three different problem statements:
>
>
> It looks like you listed four :)
>
>
>> * Web-only invisible restart: the ability for a pair of web-based
>> applications
>> to re-start a call upon re-load in a user-invisible way, i.e., without
>> prompting
>> the user to answer the call, authorize media, etc.
>>
>> * Web-only fast restart: Web-only invisible restart but optimized to be
>> faster than an initial call.
>
>
> Why wouldn't we always make invisible restarts as fast as possible?

If it required defining new protocol mechanisms, for instance.

-Ekr

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I defer to Cullen (as a chair of this group) on this issue.  I don't =
personally care one way or the other.

-- dan

On Mar 2, 2012, at 8:14 PM, DRAGE, Keith (Keith) wrote:

> Pedantic process issue.
>=20
> Do we need a proposed standard to create an IANA registry which only =
needs expert review to add a value, or will informational suffice?
>=20
> Keith
>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>> Of Dan Burnett
>> Sent: 02 March 2012 21:18
>> To: <rtcweb@ietf.org>
>> Subject: [rtcweb] New draft-burnett-rtcweb-constraints-registry-00
>>=20
>> Group,
>>=20
>> I have just submitted a new draft [1] to create a registry for HTML =
Media
>> constraints for use by the W3C getusermedia and WebRTC specs.  An =
early
>> email proposal on this subject (not the registry, but the use of it) =
is
>> available [2] for background.  I will be updating that with a fuller
>> proposal that incorporates changes discussed on the Media Capture =
Task
>> Force teleconference [3], so don't be surprised if you notice slight
>> differences in the syntax in the registry from what was proposed in =
email
>> [2].
>>=20
>> I don't know how much discussion will happen on this list for this =
draft,
>> but at the least this group should be aware of the W3C need for it.
>>=20
>> -- dan
>>=20
>> [1] =
https://datatracker.ietf.org/doc/draft-burnett-rtcweb-constraints-
>> registry/
>> [2] =
http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0198.html
>> [3] =
http://lists.w3.org/Archives/Public/public-media-capture/2012Feb/att-
>> 0053/minutes-2012-02-28.html
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] RTP clock rates
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Harald said:=20

> To be more precise: I suggested that we should incorporate by reference=20
> the advice in draft-ietf-avtext-multiple-clock-rates-05:
>=20
> 4.1.  RTP Sender (with RTCP)
>=20
>     An RTP Sender with RTCP turned on MUST use a different SSRC for each
>     different clock rate.  An RTCP BYE MUST be sent and a new SSRC MUST
>     be used if the clock rate switches back to a value already seen in
>     the RTP stream.

[BA] Yes=2C I think this makes sense.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Harald said: <br><br><div>&gt=3B To be more precise: I suggested that we sh=
ould incorporate by reference <br>&gt=3B the advice in draft-ietf-avtext-mu=
ltiple-clock-rates-05:<br>&gt=3B <br>&gt=3B 4.1.  RTP Sender (with RTCP)<br=
>&gt=3B <br>&gt=3B     An RTP Sender with RTCP turned on MUST use a differe=
nt SSRC for each<br>&gt=3B     different clock rate.  An RTCP BYE MUST be s=
ent and a new SSRC MUST<br>&gt=3B     be used if the clock rate switches ba=
ck to a value already seen in<br>&gt=3B     the RTP stream.<br><br>[BA] Yes=
=2C I think this makes sense. <br></div> 		 	   		  </div></body>
</html>=

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From Markus.Isomaki@nokia.com  Wed Jun 13 05:35:35 2012
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Subject: Re: [rtcweb] Interface switching use case and the basic needs
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Hi,

goran.ap.eriksson@ericsson.com wrote:
>>"A new higher priority interface is available. The
>> lower-priority interface is still up for a while, but you'd better
>> move your traffic/state to the new one, as the old one may be gone at
>> any point."
>
>Who is setting the "priority" of the interface? Should not the web app hav=
e a
>say whether the traffic should be kept on a particular interface or not?
>

This is usually a system-wide policy enforced by the connection manager. In=
dividual apps can't influence it. We do have a capability in Symbian by whi=
ch the apps can set preferences, but it has not been used very much - app d=
evelopers have no clue about interfaces. So far it has been quite a strict =
requirement that we want to keep only one radio interface active at a time =
to save power. The apps will just have to live with that.

There can be some special apps like IMS and MMS that have their own policie=
s, mainly because they don't work over Wi-Fi (no security, no NAT traversal=
, accounting is based on IP address etc. reasons). But normal apps have ver=
y limited powers. Web apps probably the worst since they are additionally r=
estricted by the web runtime's capabilities. =20

Markus

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> The reason for the initial data use cases was that we wanted to focus on=
=20
> the stuff that *couldn't* be done by bouncing data off a server=3B=20
> non-time-sensitive bulk file transfer can certainly be done via a=20
> server=2C even though that may not be optimal.
>
> That said=2C we've always expected that once the data transfer capability=
=20
> is in place=2C we'll have plenty of people wanting to use it in=20
> "unexpected" ways.
>
> The important RTCWEB-specific requirement from the file transfer use=20
> case is probably that IF file transfer is done in a peer-to-peer=20
> fashion=2C within the RTCWEB context=2C it MUST be able to run without=20
> degrading the quality of the real time media.

[BA] We need to be careful about the definition of "degrade".   If the goal=
 is to support the best video quality=20
possible (e.g. highest frame rate/quality/resolution supportable on the pat=
h)=2C then starting a file transfer on=20
the same path may require the video to reduce bandwidth usage.  This by its=
elf does not seem problematic.=20

However=2C it would be problematic if the file transfer builds a queue that=
 results in unacceptable latency. A
large file transfer over TCP via the server will tend to do this -- but a f=
ile transfer handled over the data channel
should not.=20

=20

 		 	   		  =

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<div dir=3D"ltr"><br><div>&gt=3B The reason for the initial data use cases =
was that we wanted to focus on <br>&gt=3B the stuff that *couldn't* be done=
 by bouncing data off a server=3B <br>&gt=3B non-time-sensitive bulk file t=
ransfer can certainly be done via a <br>&gt=3B server=2C even though that m=
ay not be optimal.<br>&gt=3B<br>&gt=3B That said=2C we've always expected t=
hat once the data transfer capability <br>&gt=3B is in place=2C we'll have =
plenty of people wanting to use it in <br>&gt=3B "unexpected" ways.<br>&gt=
=3B<br>&gt=3B The important RTCWEB-specific requirement from the file trans=
fer use <br>&gt=3B case is probably that IF file transfer is done in a peer=
-to-peer <br>&gt=3B fashion=2C within the RTCWEB context=2C it MUST be able=
 to run without <br>&gt=3B degrading the quality of the real time media.<br=
><br>[BA] We need to be careful about the definition of "degrade".&nbsp=3B&=
nbsp=3B If the goal is to support the best video quality <br>possible (e.g.=
 highest frame rate/quality/resolution supportable on the path)=2C then sta=
rting a file transfer on <br>the same path may require the video to reduce =
bandwidth usage.&nbsp=3B This by itself does not seem problematic. <br><br>=
However=2C it would be problematic if the file transfer builds a queue that=
 results in unacceptable latency. A<br>large file transfer over TCP via the=
 server will tend to do this -- but a file transfer handled over the data c=
hannel<br>should not. <br><br>&nbsp=3B<br></div></div>
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Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 06/10/2012 10:03 PM, Hutton, Andrew wrote:
> Hi Ted,
>
> See below.
>
> Regards Andy
>
>> -----Original Message----- From: Ted Hardie
>> [mailto:ted.ietf@gmail.com] Sent: 10 June 2012 18:48 To: Hutton,
>> Andrew Cc: rtcweb@ietf.org Subject: Re: [rtcweb] More Comments on
>> draft-ietf-rtcweb-use-cases-and- requirements-07
>>
>> On Sun, Jun 10, 2012 at 5:33 PM, Hutton, Andrew
>> <andrew.hutton@siemens-enterprise.com>  wrote:
>>> Hi,
>>>
>>
>>> Section 4.1.1.1 (Simple Video Communication Service) contains
>>> the
>> statement "Each user can also pause sending of media (audio, video,
>> or both) and mute incoming media".  I think this and the
>> associated requirements need some clarification for example whether
>> the requirement is to control the direction of the media and
>> therefore requires the generation of a new SDP offer with for
>> example a direction of "recvonly/inactive".
>>
>> Howdy,
>>
>> As a clarifying question, are you thinking that it is not clear
>> because non-signaling methods could achieve some of these?  That
>> is "muting" incoming media could also be accomplished entirely
>> receive side, by simply not rendering it?
>>
>
> Yes I guess so it is not clear whether the use case results in a
> local action in the browser or it whether it impacts the media and
> SDP. I want to make sure that it is possible to control the media
> direction in SDP (sendrecv/recvonly/sendonly/inactive) and have the
> browser behave appropriately.

This is my personal understanding:

- the API (and this part has been there for a long time) allows the 
application to disable/enable individual tracks of MediaStreams
- So say you have a MediaStream (one audio track, one video track in 
this example) that is rendered using a video element; disabling audio 
would lead to silence, disabling video to that the video stops
- The question is: what should happen if the MediaStream is connected to 
a PeerConnection and sent to a peer?
- using recvonly does not seem appropriate as that would stop all ssrc:s 
belonging to the same m-line - and the app only disabled one
- one option could be to simply stop sending the RTP packets for this 
track. I don't know how reliable that would be
- I think there has been proposals signaling this in SDP or in RTCP, but 
I don't think there was ever any conclusion

For the other part, muting incoming media, this is simple: the app (at 
receiving browser) could just disable the tracks it wants in the 
incoming MediaStreams. A related question is if this should be signaled 
back to the sender (so it could stop sending). This is essentially a 
question of efficiency/optimization, and nothing stops the app from 
signaling that now using an app specific protocol (perhaps carried over 
the data channel), but perhaps this should be brought into SDP or RTCP.

>
>
>> regards,
>>
>> Ted
>>
>> The only associated requirement seems to be A8 but this does not
>> help clarify what is meant by "pause" in the use case. It needs to
>> be clarified whether the pause/resume/mute are actions within the
>> browser or are actions which result in notifications to the remote
>> user.
>>>
>>> Maybe we could add a use case which involves controlling the
>> direction of multiple streams to emulate switching between
>> different users in a multiparty call scenario.
>>>
>>>
>>> The statement "It is essential that the communication cannot be
>> eavesdropped" appears in all use cases so could be moved to section
>> 4.1 but I think it needs to be reworded in terms of the media
>> always being encrypted as I don't like the term "eavesdropped"
>> which is not defined.
>>>
>>> Regards Andy
>>>
>>>
>>>
>>> _______________________________________________ rtcweb mailing
>>> list rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] draft-ietf-rtcweb-data-channel-00
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On 6/13/12 1:33 PM, DRAGE, Keith (Keith) wrote:
> Quick question.
>
> In this document are "data stream" and "datagram stream" one and the same thing, or not?
>
> If yes can we harmonise the term?
thanks
  we are going to harmonies the term in the next 01 version

Salvatore
>
> If no, can we define what the distinct terms mean?
>
> Keith
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Thread-Topic: Combining consent freshness and liveness
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Taking a closer look at the "Combined Consent/Liveness Proposal II" (slide =
16) from Eric's RTCWeb security presentation today, it seems to align close=
ly with what said in WebEx -- determining consent freshness must be under t=
he browser control and determining session liveness must be under the appli=
cation control. If I understood correctly, the proposal is to have the cons=
ent timer Tc under the browser control and the packet receipt timer Tr unde=
r the application control. So, I full support it.

I can work with Eric to incorporate it into draft-muthu-behave-consent-fres=
hness. However, I would like us to reach a (rough) consensus on whether the=
 STUN transactions for consent freshness / liveness should include or omit =
the message integrity and thereby use the STUN Binding requests/responses o=
r a new method.

Muthu

--_000_E721D8C6A2E1544DB2DEBC313AF54DE2012792EBxmbrcdx02ciscoc_
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<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Taking a closer look at the &quot;Combined Consent/Livenes=
s Proposal II&quot; (slide 16) from Eric's RTCWeb security presentation tod=
ay, it seems to align closely with what said in WebEx --
 determining consent freshness must be under the browser control and determ=
ining session liveness must be under the application control. If I understo=
od correctly, the proposal is to have the consent timer Tc under the browse=
r control and the packet receipt
 timer Tr under the application control. So, I full support it.<o:p></o:p><=
/span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">I can work with Eric to incorporate it into draft-muthu-be=
have-consent-freshness. However, I would like us to reach a (rough) consens=
us on whether the STUN transactions for consent
 freshness / liveness should include or omit the message integrity and ther=
eby use the STUN Binding requests/responses or a new method.<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">Muthu<o:p></o:p></span></p>
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From: Lishitao <lishitao@huawei.com>
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Thread-Topic: Interface switching use case and the basic needs
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From harald@alvestrand.no  Wed Jun 13 23:29:40 2012
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Subject: Re: [rtcweb] Combining consent freshness and liveness
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On 06/13/2012 08:34 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> Taking a closer look at the "Combined Consent/Liveness Proposal II" 
> (slide 16) from Eric's RTCWeb security presentation today, it seems to 
> align closely with what said in WebEx -- determining consent freshness 
> must be under the browser control and determining session liveness 
> must be under the application control. If I understood correctly, the 
> proposal is to have the consent timer Tc under the browser control and 
> the packet receipt timer Tr under the application control. So, I full 
> support it.
>
To the extent that I understand it (and my understanding is the same as 
yours), I agree.
>
> I can work with Eric to incorporate it into 
> draft-muthu-behave-consent-freshness. However, I would like us to 
> reach a (rough) consensus on whether the STUN transactions for consent 
> freshness / liveness should include or omit the message integrity and 
> thereby use the STUN Binding requests/responses or a new method.
>
As I've said before, I don't see a value in omitting the message 
integrity that offsets the (high) cost of introducing a new method. I 
think we should use the binding request/response.



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    On 06/13/2012 08:34 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
    <blockquote
cite="mid:E721D8C6A2E1544DB2DEBC313AF54DE2012792EB@xmb-rcd-x02.cisco.com"
      type="cite">
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;;">Taking a closer look at the
            "Combined Consent/Liveness Proposal II" (slide 16) from
            Eric's RTCWeb security presentation today, it seems to align
            closely with what said in WebEx -- determining consent
            freshness must be under the browser control and determining
            session liveness must be under the application control. If I
            understood correctly, the proposal is to have the consent
            timer Tc under the browser control and the packet receipt
            timer Tr under the application control. So, I full support
            it.</span></p>
      </div>
    </blockquote>
    To the extent that I understand it (and my understanding is the same
    as yours), I agree.<br>
    <blockquote
cite="mid:E721D8C6A2E1544DB2DEBC313AF54DE2012792EB@xmb-rcd-x02.cisco.com"
      type="cite">
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        <p class="MsoNormal"><span
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        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;;">I can work with Eric to
            incorporate it into draft-muthu-behave-consent-freshness.
            However, I would like us to reach a (rough) consensus on
            whether the STUN transactions for consent freshness /
            liveness should include or omit the message integrity and
            thereby use the STUN Binding requests/responses or a new
            method.</span></p>
      </div>
    </blockquote>
    As I've said before, I don't see a value in omitting the message
    integrity that offsets the (high) cost of introducing a new method.
    I think we should use the binding request/response.<br>
    <br>
    <br>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Back-to-back RTP endpoint topology in RTCWeb
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Hi Partha,

I think there is several different realizations of this.

If you goal is only to hide one end-points IP address to the other it
can be the relay topology, with an 1-1 participation. In fact you don't
really need to discuss this as an RTP topology at all as all operations
are on IP and Transport level. A TURN server will do perform this hiding
in one direction. A web service controlled relay node could do this for
both end-points.

If your goal is something additional I would claim that this falls in
the broad category of things gateways do. The problem is as soon as you
enable any RTP/RTCP level of modifications you end up as node needing to
be trusted and part of the security contexts.

I didn't bring up gateways as they are a bit undefined in their
operations as it depends on which aspect they need to change. My general
classification of gateways are that they are devices that lets the
combination of the peer end-point plus the gateway look like a node
expected or at least supported by the communicating party.

Cheers

Magnus


On 2012-06-13 12:46, Ravindran, Parthasarathi wrote:
> Hi Magnus & all,
> 
> I like the list of RTP topology mentioned in your interim
> presentation. I thought of asking for one more topology namely
> Back-to-back RTP endpoint topology in RTCWeb. The topology is as
> follows:
> 
> RTP endpoint-----Back-to-back RTP Endpoint-----RTP Endpoint
> 
> The reason for Back-to-back RTP Endpoint topology are
> 
> 1) WebRTC RTP Endpoint to legacy RTP Endpoint (RFC 3550) interop 2)
> IP address hiding in WebRTC session
> 
> Even though, Back-to-back RTP Endpoint *MUST* follow RTP Endpoint
> behavior, the requirement will be bit more than that as the linkage
> with identity, ICE consent, RTCP mandate, set of RTP draft mandates
> and other stuffs. Also, the guidelines document for back-to-back RTP
> Endpoint shall avoid the blame of intermediate devices always break
> the architecture.
> 
> I'm interested in hearing whether Back-to-back RTP endpoint topology
> is of interest in RTCWeb.
> 
> Thanks Partha


-- 

Magnus Westerlund

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


From andrew.hutton@siemens-enterprise.com  Thu Jun 14 02:15:18 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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Subject: Re: [rtcweb] More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 13 June 2012 15:37 Stefan Hakansson wrote:

>=20
> This is my personal understanding:
>=20
> - the API (and this part has been there for a long time) allows the
> application to disable/enable individual tracks of MediaStreams
> - So say you have a MediaStream (one audio track, one video track in
> this example) that is rendered using a video element; disabling audio
> would lead to silence, disabling video to that the video stops
> - The question is: what should happen if the MediaStream is connected
> to
> a PeerConnection and sent to a peer?
> - using recvonly does not seem appropriate as that would stop all
> ssrc:s
> belonging to the same m-line - and the app only disabled one
> - one option could be to simply stop sending the RTP packets for this
> track. I don't know how reliable that would be.

In the case of outgoing media then I think mute should simply disconnect th=
e source (e.g. microphone) and there is no impact on RTP other than dependi=
ng on the codec silence will be transmitted or SID frame etc. There is also=
 no impact on RTCP which continues.

This is different to setting the direction to be recvonly which would compl=
etely stop the outgoing RTP packets and would normally be signaled to the r=
emote party. This should of course be achieved using setLocalDescription.


> - I think there has been proposals signaling this in SDP or in RTCP,
> but
> I don't think there was ever any conclusion

If we are talking about a local mute then I don't see any need to signal th=
is but again this is different to changing the direction attribute in SDP.

>=20
> For the other part, muting incoming media, this is simple: the app (at
> receiving browser) could just disable the tracks it wants in the
> incoming MediaStreams. A related question is if this should be signaled
> back to the sender (so it could stop sending). This is essentially a
> question of efficiency/optimization, and nothing stops the app from
> signaling that now using an app specific protocol (perhaps carried over
> the data channel), but perhaps this should be brought into SDP or RTCP.
>=20

If the application wants to ask the remote party to pause sending of RTP th=
en it should surely do this in SDP by setting the direction attribute appro=
priately (i.e. sendonly or inactive).

Regards
Andy


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Subject: [rtcweb] Terminology "Data[gram] stream" (Re: draft-ietf-rtcweb-data-channel-00)
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On 06/13/2012 04:36 PM, Salvatore Loreto wrote:
> On 6/13/12 1:33 PM, DRAGE, Keith (Keith) wrote:
>> Quick question.
>>
>> In this document are "data stream" and "datagram stream" one and the 
>> same thing, or not?
>>
>> If yes can we harmonise the term?
> thanks
>  we are going to harmonies the term in the next 01 version

Personal preference, weak:

Could we harmonize the data channel draft on the term "datagram stream"?
It's slightly less likely to be confusing/irritating to people who think 
that media is data, too....

I'm willing to be convinced the other one is better, but if nobody else 
has a preference, we might as well do it this way (and I can add the 
term to the overview document, for document set consistency).


>
> Salvatore
>>
>> If no, can we define what the distinct terms mean?
>>
>> Keith
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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Changing the subject line again, since we're discussing how to do mute, 
not whether to do it.....

On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
> On 13 June 2012 15:37 Stefan Hakansson wrote:
>
>> This is my personal understanding:
>>
>> - the API (and this part has been there for a long time) allows the
>> application to disable/enable individual tracks of MediaStreams
>> - So say you have a MediaStream (one audio track, one video track in
>> this example) that is rendered using a video element; disabling audio
>> would lead to silence, disabling video to that the video stops
>> - The question is: what should happen if the MediaStream is connected
>> to
>> a PeerConnection and sent to a peer?
>> - using recvonly does not seem appropriate as that would stop all
>> ssrc:s
>> belonging to the same m-line - and the app only disabled one
>> - one option could be to simply stop sending the RTP packets for this
>> track. I don't know how reliable that would be.
> In the case of outgoing media then I think mute should simply disconnect the source (e.g. microphone) and there is no impact on RTP other than depending on the codec silence will be transmitted or SID frame etc. There is also no impact on RTCP which continues.
>
> This is different to setting the direction to be recvonly which would completely stop the outgoing RTP packets and would normally be signaled to the remote party. This should of course be achieved using setLocalDescription.

Just to add to the complexity of the "mute" case:

When an audio stream is muted, should there be comfort noise sent?
If yes, what should determine the noise level?

(people who've never had to deal with comfort noise can safely tune out 
at this point)

>
>
>> - I think there has been proposals signaling this in SDP or in RTCP,
>> but
>> I don't think there was ever any conclusion
> If we are talking about a local mute then I don't see any need to signal this but again this is different to changing the direction attribute in SDP.
>
>> For the other part, muting incoming media, this is simple: the app (at
>> receiving browser) could just disable the tracks it wants in the
>> incoming MediaStreams. A related question is if this should be signaled
>> back to the sender (so it could stop sending). This is essentially a
>> question of efficiency/optimization, and nothing stops the app from
>> signaling that now using an app specific protocol (perhaps carried over
>> the data channel), but perhaps this should be brought into SDP or RTCP.
>>
> If the application wants to ask the remote party to pause sending of RTP then it should surely do this in SDP by setting the direction attribute appropriately (i.e. sendonly or inactive).
>
> Regards
> Andy
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From salvatore.loreto@ericsson.com  Thu Jun 14 02:22:42 2012
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From: Salvatore Loreto <salvatore.loreto@ericsson.com>
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Subject: Re: [rtcweb] Terminology "Data[gram] stream" (Re: draft-ietf-rtcweb-data-channel-00)
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On 6/14/12 12:19 PM, Harald Alvestrand wrote:
> On 06/13/2012 04:36 PM, Salvatore Loreto wrote:
>> On 6/13/12 1:33 PM, DRAGE, Keith (Keith) wrote:
>>> Quick question.
>>>
>>> In this document are "data stream" and "datagram stream" one and the
>>> same thing, or not?
>>>
>>> If yes can we harmonise the term?
>> thanks
>>   we are going to harmonies the term in the next 01 version
> Personal preference, weak:
>
> Could we harmonize the data channel draft on the term "datagram stream"?
> It's slightly less likely to be confusing/irritating to people who think
> that media is data, too....
that is exactly what I have in mind to do!

>
> I'm willing to be convinced the other one is better, but if nobody else
> has a preference, we might as well do it this way (and I can add the
> term to the overview document, for document set consistency).
that would be great!



>
>
>> Salvatore
>>> If no, can we define what the distinct terms mean?
>>>
>>> Keith
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/14/2012 11:21 AM, Harald Alvestrand wrote:
> Changing the subject line again, since we're discussing how to do
> mute, not whether to do it.....
>
> On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
>> On 13 June 2012 15:37 Stefan Hakansson wrote:
>>
>>> This is my personal understanding:
>>>
>>> - the API (and this part has been there for a long time) allows
>>> the application to disable/enable individual tracks of
>>> MediaStreams - So say you have a MediaStream (one audio track,
>>> one video track in this example) that is rendered using a video
>>> element; disabling audio would lead to silence, disabling video
>>> to that the video stops - The question is: what should happen if
>>> the MediaStream is connected to a PeerConnection and sent to a
>>> peer? - using recvonly does not seem appropriate as that would
>>> stop all ssrc:s belonging to the same m-line - and the app only
>>> disabled one - one option could be to simply stop sending the RTP
>>> packets for this track. I don't know how reliable that would be.
>> In the case of outgoing media then I think mute should simply
>> disconnect the source (e.g. microphone) and there is no impact on
>> RTP other than depending on the codec silence will be transmitted
>> or SID frame etc. There is also no impact on RTCP which continues.
>>
>> This is different to setting the direction to be recvonly which
>> would completely stop the outgoing RTP packets and would normally
>> be signaled to the remote party. This should of course be achieved
>> using setLocalDescription.
>
> Just to add to the complexity of the "mute" case:
>
> When an audio stream is muted, should there be comfort noise sent? If
> yes, what should determine the noise level?

To me it feels as that ideally we should stop sending RTP (and make the 
other end know that the track is disabled). If we go to comfort noise 
state, the track in the MediaStream at the other end would still be 
active (even though what is played could be very low level comfort 
noise), but to reflect the local state the remote track should go to 
"muted", shouldn't it?


>
> (people who've never had to deal with comfort noise can safely tune
> out at this point)
>
>>
>>
>>> - I think there has been proposals signaling this in SDP or in
>>> RTCP, but I don't think there was ever any conclusion
>> If we are talking about a local mute then I don't see any need to
>> signal this but again this is different to changing the direction
>> attribute in SDP.
>>
>>> For the other part, muting incoming media, this is simple: the
>>> app (at receiving browser) could just disable the tracks it wants
>>> in the incoming MediaStreams. A related question is if this
>>> should be signaled back to the sender (so it could stop sending).
>>> This is essentially a question of efficiency/optimization, and
>>> nothing stops the app from signaling that now using an app
>>> specific protocol (perhaps carried over the data channel), but
>>> perhaps this should be brought into SDP or RTCP.
>>>
>> If the application wants to ask the remote party to pause sending
>> of RTP then it should surely do this in SDP by setting the
>> direction attribute appropriately (i.e. sendonly or inactive).
>>
>> Regards Andy
>>
>> _______________________________________________ rtcweb mailing
>> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Combining consent freshness and liveness
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On 13 June 2012 23:29, Harald Alvestrand <harald@alvestrand.no> wrote:
> As I've said before, I don't see a value in omitting the message integrity
> that offsets the (high) cost of introducing a new method. I think we should
> use the binding request/response.

While it was acknowledged that the security analysis didn't depend on
having the integrity check, I completely agree with Harald.  ...for at
least these reasons:
 * As discussed, the additional cost of introducing a method, which I
cannot emphasize enough is quite high.
 * Reusing the binding request allows browsers to interoperate with
existing ICE agents; even ICE-lite agents.  That's really important.

I'm sympathetic to the cost of calculating SHA-1, but is not so
expensive (either in terms of packets or in terms of CPU cycles) that
it warrants breaking these.

From bernard_aboba@hotmail.com  Thu Jun 14 11:58:43 2012
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Subject: Re: [rtcweb] Combining consent freshness and liveness
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Agree that introducing a new method is not a good idea.  ICE has been imple=
mented
and deployed=2C and so we should not be breaking interoperability without a=
 very good
reason. =20

The cost of SHA-1 computation is not a good enough justification for the pa=
in a=20
new method would cause.=20

> Date: Thu=2C 14 Jun 2012 09:08:18 -0700
> From: martin.thomson@gmail.com
> To: harald@alvestrand.no
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] Combining consent freshness and liveness
>=20
> On 13 June 2012 23:29=2C Harald Alvestrand <harald@alvestrand.no> wrote:
> > As I've said before=2C I don't see a value in omitting the message inte=
grity
> > that offsets the (high) cost of introducing a new method. I think we sh=
ould
> > use the binding request/response.
>=20
> While it was acknowledged that the security analysis didn't depend on
> having the integrity check=2C I completely agree with Harald.  ...for at
> least these reasons:
>  * As discussed=2C the additional cost of introducing a method=2C which I
> cannot emphasize enough is quite high.
>  * Reusing the binding request allows browsers to interoperate with
> existing ICE agents=3B even ICE-lite agents.  That's really important.
>=20
> I'm sympathetic to the cost of calculating SHA-1=2C but is not so
> expensive (either in terms of packets or in terms of CPU cycles) that
> it warrants breaking these.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Agree that introducing a new method is not a good idea.&nbsp=3B ICE has bee=
n implemented<br>and deployed=2C and so we should not be breaking interoper=
ability without a very good<br>reason.&nbsp=3B <br><br>The cost of SHA-1 co=
mputation is not a good enough justification for the pain a <br>new method =
would cause. <br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B Date=
: Thu=2C 14 Jun 2012 09:08:18 -0700<br>&gt=3B From: martin.thomson@gmail.co=
m<br>&gt=3B To: harald@alvestrand.no<br>&gt=3B CC: rtcweb@ietf.org<br>&gt=
=3B Subject: Re: [rtcweb] Combining consent freshness and liveness<br>&gt=
=3B <br>&gt=3B On 13 June 2012 23:29=2C Harald Alvestrand &lt=3Bharald@alve=
strand.no&gt=3B wrote:<br>&gt=3B &gt=3B As I've said before=2C I don't see =
a value in omitting the message integrity<br>&gt=3B &gt=3B that offsets the=
 (high) cost of introducing a new method. I think we should<br>&gt=3B &gt=
=3B use the binding request/response.<br>&gt=3B <br>&gt=3B While it was ack=
nowledged that the security analysis didn't depend on<br>&gt=3B having the =
integrity check=2C I completely agree with Harald.  ...for at<br>&gt=3B lea=
st these reasons:<br>&gt=3B  * As discussed=2C the additional cost of intro=
ducing a method=2C which I<br>&gt=3B cannot emphasize enough is quite high.=
<br>&gt=3B  * Reusing the binding request allows browsers to interoperate w=
ith<br>&gt=3B existing ICE agents=3B even ICE-lite agents.  That's really i=
mportant.<br>&gt=3B <br>&gt=3B I'm sympathetic to the cost of calculating S=
HA-1=2C but is not so<br>&gt=3B expensive (either in terms of packets or in=
 terms of CPU cycles) that<br>&gt=3B it warrants breaking these.<br>&gt=3B =
_______________________________________________<br>&gt=3B rtcweb mailing li=
st<br>&gt=3B rtcweb@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinf=
o/rtcweb<br></div> 		 	   		  </div></body>
</html>=

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From: Justin Uberti <juberti@google.com>
Date: Thu, 14 Jun 2012 15:36:55 -0400
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To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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--14dae934064fb2900404c273d3e2
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For "mute", stopping sending of RTP may trigger concealment on the remote
side. We currently send silent audio when muted to prevent this.
For "hold", you are notifying the remote side (e.g. a=inactive), so it
makes sense to stop sending RTP.

On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 06/14/2012 11:21 AM, Harald Alvestrand wrote:
>
>> Changing the subject line again, since we're discussing how to do
>> mute, not whether to do it.....
>>
>> On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
>>
>>> On 13 June 2012 15:37 Stefan Hakansson wrote:
>>>
>>>  This is my personal understanding:
>>>>
>>>> - the API (and this part has been there for a long time) allows
>>>> the application to disable/enable individual tracks of
>>>> MediaStreams - So say you have a MediaStream (one audio track,
>>>> one video track in this example) that is rendered using a video
>>>> element; disabling audio would lead to silence, disabling video
>>>> to that the video stops - The question is: what should happen if
>>>> the MediaStream is connected to a PeerConnection and sent to a
>>>> peer? - using recvonly does not seem appropriate as that would
>>>> stop all ssrc:s belonging to the same m-line - and the app only
>>>> disabled one - one option could be to simply stop sending the RTP
>>>> packets for this track. I don't know how reliable that would be.
>>>>
>>> In the case of outgoing media then I think mute should simply
>>> disconnect the source (e.g. microphone) and there is no impact on
>>> RTP other than depending on the codec silence will be transmitted
>>> or SID frame etc. There is also no impact on RTCP which continues.
>>>
>>> This is different to setting the direction to be recvonly which
>>> would completely stop the outgoing RTP packets and would normally
>>> be signaled to the remote party. This should of course be achieved
>>> using setLocalDescription.
>>>
>>
>> Just to add to the complexity of the "mute" case:
>>
>> When an audio stream is muted, should there be comfort noise sent? If
>> yes, what should determine the noise level?
>>
>
> To me it feels as that ideally we should stop sending RTP (and make the
> other end know that the track is disabled). If we go to comfort noise
> state, the track in the MediaStream at the other end would still be active
> (even though what is played could be very low level comfort noise), but to
> reflect the local state the remote track should go to "muted", shouldn't it?
>
>
>
>
>> (people who've never had to deal with comfort noise can safely tune
>> out at this point)
>>
>>
>>>
>>>  - I think there has been proposals signaling this in SDP or in
>>>> RTCP, but I don't think there was ever any conclusion
>>>>
>>> If we are talking about a local mute then I don't see any need to
>>> signal this but again this is different to changing the direction
>>> attribute in SDP.
>>>
>>>  For the other part, muting incoming media, this is simple: the
>>>> app (at receiving browser) could just disable the tracks it wants
>>>> in the incoming MediaStreams. A related question is if this
>>>> should be signaled back to the sender (so it could stop sending).
>>>> This is essentially a question of efficiency/optimization, and
>>>> nothing stops the app from signaling that now using an app
>>>> specific protocol (perhaps carried over the data channel), but
>>>> perhaps this should be brought into SDP or RTCP.
>>>>
>>>>  If the application wants to ask the remote party to pause sending
>>> of RTP then it should surely do this in SDP by setting the
>>> direction attribute appropriately (i.e. sendonly or inactive).
>>>
>>> Regards Andy
>>>
>>> ______________________________**_________________ rtcweb mailing
>>> list rtcweb@ietf.org https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>>
>>
>> ______________________________**_________________ rtcweb mailing list
>> rtcweb@ietf.org https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--14dae934064fb2900404c273d3e2
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div style=3D"font-family:arial,helvetica,sans-serif"><font>For &quot;mute&=
quot;, stopping sending of RTP may trigger concealment on the remote side. =
We currently send silent audio when muted to prevent this.<div>
For &quot;hold&quot;, you are notifying the remote side (e.g. a=3Dinactive)=
, so it makes sense to stop sending RTP.<br><br><div class=3D"gmail_quote">=
On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansson LK <span dir=3D"ltr">&lt;=
<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank">stefa=
n.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"HOEnZb"><div class=3D"h5">On 0=
6/14/2012 11:21 AM, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Changing the subject line again, since we&#39;re discussing how to do<br>
mute, not whether to do it.....<br>
<br>
On 06/14/2012 11:15 AM, Hutton, Andrew wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 13 June 2012 15:37 Stefan Hakansson wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
This is my personal understanding:<br>
<br>
- the API (and this part has been there for a long time) allows<br>
the application to disable/enable individual tracks of<br>
MediaStreams - So say you have a MediaStream (one audio track,<br>
one video track in this example) that is rendered using a video<br>
element; disabling audio would lead to silence, disabling video<br>
to that the video stops - The question is: what should happen if<br>
the MediaStream is connected to a PeerConnection and sent to a<br>
peer? - using recvonly does not seem appropriate as that would<br>
stop all ssrc:s belonging to the same m-line - and the app only<br>
disabled one - one option could be to simply stop sending the RTP<br>
packets for this track. I don&#39;t know how reliable that would be.<br>
</blockquote>
In the case of outgoing media then I think mute should simply<br>
disconnect the source (e.g. microphone) and there is no impact on<br>
RTP other than depending on the codec silence will be transmitted<br>
or SID frame etc. There is also no impact on RTCP which continues.<br>
<br>
This is different to setting the direction to be recvonly which<br>
would completely stop the outgoing RTP packets and would normally<br>
be signaled to the remote party. This should of course be achieved<br>
using setLocalDescription.<br>
</blockquote>
<br>
Just to add to the complexity of the &quot;mute&quot; case:<br>
<br>
When an audio stream is muted, should there be comfort noise sent? If<br>
yes, what should determine the noise level?<br>
</blockquote>
<br></div></div>
To me it feels as that ideally we should stop sending RTP (and make the oth=
er end know that the track is disabled). If we go to comfort noise state, t=
he track in the MediaStream at the other end would still be active (even th=
ough what is played could be very low level comfort noise), but to reflect =
the local state the remote track should go to &quot;muted&quot;, shouldn&#3=
9;t it?<div class=3D"HOEnZb">

<div class=3D"h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
(people who&#39;ve never had to deal with comfort noise can safely tune<br>
out at this point)<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
- I think there has been proposals signaling this in SDP or in<br>
RTCP, but I don&#39;t think there was ever any conclusion<br>
</blockquote>
If we are talking about a local mute then I don&#39;t see any need to<br>
signal this but again this is different to changing the direction<br>
attribute in SDP.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
For the other part, muting incoming media, this is simple: the<br>
app (at receiving browser) could just disable the tracks it wants<br>
in the incoming MediaStreams. A related question is if this<br>
should be signaled back to the sender (so it could stop sending).<br>
This is essentially a question of efficiency/optimization, and<br>
nothing stops the app from signaling that now using an app<br>
specific protocol (perhaps carried over the data channel), but<br>
perhaps this should be brought into SDP or RTCP.<br>
<br>
</blockquote>
If the application wants to ask the remote party to pause sending<br>
of RTP then it should surely do this in SDP by setting the<br>
direction attribute appropriately (i.e. sendonly or inactive).<br>
<br>
Regards Andy<br>
<br>
______________________________<u></u>_________________ rtcweb mailing<br>
list <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blan=
k">https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________ rtcweb mailing list<=
br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a> <a=
 href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">ht=
tps://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></font></div>

--14dae934064fb2900404c273d3e2--

From andrew.hutton@siemens-enterprise.com  Thu Jun 14 13:01:14 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
Thread-Index: AQHNSi8V99bC4xTMh0uQV/Y6A7X7Fpb6FDaAgAAoTow=
Date: Thu, 14 Jun 2012 20:01:10 +0000
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on	draft-ietf-rtcweb-use-cases-and-requirements-07)
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I agree with Justin that mute is a local action and should not stop the RTP=
 or change the state of the remote track.

Regards
Andy



On 14 Jun 2012, at 20:37, "Justin Uberti" <juberti@google.com<mailto:jubert=
i@google.com>> wrote:

For "mute", stopping sending of RTP may trigger concealment on the remote s=
ide. We currently send silent audio when muted to prevent this.
For "hold", you are notifying the remote side (e.g. a=3Dinactive), so it ma=
kes sense to stop sending RTP.

On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansson LK <stefan.lk.hakansson@e=
ricsson.com<mailto:stefan.lk.hakansson@ericsson.com>> wrote:
On 06/14/2012 11:21 AM, Harald Alvestrand wrote:
Changing the subject line again, since we're discussing how to do
mute, not whether to do it.....

On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
On 13 June 2012 15:37 Stefan Hakansson wrote:

This is my personal understanding:

- the API (and this part has been there for a long time) allows
the application to disable/enable individual tracks of
MediaStreams - So say you have a MediaStream (one audio track,
one video track in this example) that is rendered using a video
element; disabling audio would lead to silence, disabling video
to that the video stops - The question is: what should happen if
the MediaStream is connected to a PeerConnection and sent to a
peer? - using recvonly does not seem appropriate as that would
stop all ssrc:s belonging to the same m-line - and the app only
disabled one - one option could be to simply stop sending the RTP
packets for this track. I don't know how reliable that would be.
In the case of outgoing media then I think mute should simply
disconnect the source (e.g. microphone) and there is no impact on
RTP other than depending on the codec silence will be transmitted
or SID frame etc. There is also no impact on RTCP which continues.

This is different to setting the direction to be recvonly which
would completely stop the outgoing RTP packets and would normally
be signaled to the remote party. This should of course be achieved
using setLocalDescription.

Just to add to the complexity of the "mute" case:

When an audio stream is muted, should there be comfort noise sent? If
yes, what should determine the noise level?

To me it feels as that ideally we should stop sending RTP (and make the oth=
er end know that the track is disabled). If we go to comfort noise state, t=
he track in the MediaStream at the other end would still be active (even th=
ough what is played could be very low level comfort noise), but to reflect =
the local state the remote track should go to "muted", shouldn't it?




(people who've never had to deal with comfort noise can safely tune
out at this point)



- I think there has been proposals signaling this in SDP or in
RTCP, but I don't think there was ever any conclusion
If we are talking about a local mute then I don't see any need to
signal this but again this is different to changing the direction
attribute in SDP.

For the other part, muting incoming media, this is simple: the
app (at receiving browser) could just disable the tracks it wants
in the incoming MediaStreams. A related question is if this
should be signaled back to the sender (so it could stop sending).
This is essentially a question of efficiency/optimization, and
nothing stops the app from signaling that now using an app
specific protocol (perhaps carried over the data channel), but
perhaps this should be brought into SDP or RTCP.

If the application wants to ask the remote party to pause sending
of RTP then it should surely do this in SDP by setting the
direction attribute appropriately (i.e. sendonly or inactive).

Regards Andy

_______________________________________________ rtcweb mailing
list rtcweb@ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/l=
istinfo/rtcweb

_______________________________________________ rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org> https://www.ietf.org/mailman/listin=
fo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org<mailto:rtcweb@ietf.org>
https://www.ietf.org/mailman/listinfo/rtcweb

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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body bgcolor=3D"#FFFFFF">
<div>I agree with Justin that mute is a local action and should not stop th=
e RTP or change the state of the remote track.</div>
<div><br>
</div>
<div>Regards</div>
<div>Andy<br>
<br>
<br>
</div>
<div><br>
On 14 Jun 2012, at 20:37, &quot;Justin Uberti&quot; &lt;<a href=3D"mailto:j=
uberti@google.com">juberti@google.com</a>&gt; wrote:<br>
<br>
</div>
<div></div>
<blockquote type=3D"cite">
<div>
<div style=3D"font-family:arial,helvetica,sans-serif"><font>For &quot;mute&=
quot;, stopping sending of RTP may trigger concealment on the remote side. =
We currently send silent audio when muted to prevent this.
<div>For &quot;hold&quot;, you are notifying the remote side (e.g. a=3Dinac=
tive), so it makes sense to stop sending RTP.<br>
<br>
<div class=3D"gmail_quote">On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansso=
n LK <span dir=3D"ltr">
&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank">s=
tefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb">
<div class=3D"h5">On 06/14/2012 11:21 AM, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Changing the subject line again, since we're discussing how to do<br>
mute, not whether to do it.....<br>
<br>
On 06/14/2012 11:15 AM, Hutton, Andrew wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 13 June 2012 15:37 Stefan Hakansson wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
This is my personal understanding:<br>
<br>
- the API (and this part has been there for a long time) allows<br>
the application to disable/enable individual tracks of<br>
MediaStreams - So say you have a MediaStream (one audio track,<br>
one video track in this example) that is rendered using a video<br>
element; disabling audio would lead to silence, disabling video<br>
to that the video stops - The question is: what should happen if<br>
the MediaStream is connected to a PeerConnection and sent to a<br>
peer? - using recvonly does not seem appropriate as that would<br>
stop all ssrc:s belonging to the same m-line - and the app only<br>
disabled one - one option could be to simply stop sending the RTP<br>
packets for this track. I don't know how reliable that would be.<br>
</blockquote>
In the case of outgoing media then I think mute should simply<br>
disconnect the source (e.g. microphone) and there is no impact on<br>
RTP other than depending on the codec silence will be transmitted<br>
or SID frame etc. There is also no impact on RTCP which continues.<br>
<br>
This is different to setting the direction to be recvonly which<br>
would completely stop the outgoing RTP packets and would normally<br>
be signaled to the remote party. This should of course be achieved<br>
using setLocalDescription.<br>
</blockquote>
<br>
Just to add to the complexity of the &quot;mute&quot; case:<br>
<br>
When an audio stream is muted, should there be comfort noise sent? If<br>
yes, what should determine the noise level?<br>
</blockquote>
<br>
</div>
</div>
To me it feels as that ideally we should stop sending RTP (and make the oth=
er end know that the track is disabled). If we go to comfort noise state, t=
he track in the MediaStream at the other end would still be active (even th=
ough what is played could be very
 low level comfort noise), but to reflect the local state the remote track =
should go to &quot;muted&quot;, shouldn't it?
<div class=3D"HOEnZb">
<div class=3D"h5"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
(people who've never had to deal with comfort noise can safely tune<br>
out at this point)<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
- I think there has been proposals signaling this in SDP or in<br>
RTCP, but I don't think there was ever any conclusion<br>
</blockquote>
If we are talking about a local mute then I don't see any need to<br>
signal this but again this is different to changing the direction<br>
attribute in SDP.<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
For the other part, muting incoming media, this is simple: the<br>
app (at receiving browser) could just disable the tracks it wants<br>
in the incoming MediaStreams. A related question is if this<br>
should be signaled back to the sender (so it could stop sending).<br>
This is essentially a question of efficiency/optimization, and<br>
nothing stops the app from signaling that now using an app<br>
specific protocol (perhaps carried over the data channel), but<br>
perhaps this should be brought into SDP or RTCP.<br>
<br>
</blockquote>
If the application wants to ask the remote party to pause sending<br>
of RTP then it should surely do this in SDP by setting the<br>
direction attribute appropriately (i.e. sendonly or inactive).<br>
<br>
Regards Andy<br>
<br>
______________________________<u></u>_________________ rtcweb mailing<br>
list <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</=
a> <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blan=
k">
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________ rtcweb mailing list<=
br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a> <a=
 href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div>
</div>
</blockquote>
</div>
<br>
</div>
</font></div>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

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From jonathan@vidyo.com  Thu Jun 14 13:03:21 2012
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From: Jonathan Lennox <jonathan@vidyo.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 14 Jun 2012 16:03:17 -0400
Thread-Topic: [rtcweb] What I think we need to do about ICE
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Subject: Re: [rtcweb] What I think we need to do about ICE
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On Jun 12, 2012, at 10:32 AM, Eric Rescorla wrote:

> To recap what I was saying on the list:
>=20
> 1. Some method of determining when a candidate pair has failed.
>=20
> 2. Some method of failing over to another candidate pair after
> failure in as clean a fashion as possible, preferably exploiting
> the fact that we already know about other valid candidate pairs,
> if possible. Example: I had a direct and relayed pair that worked
> and I chose direct. Is there a way to fall back to relayed without
> doing a total re-ice.
>=20
> 3. When I discover a new candidate pair, some way of investigating
> it and maybe changing to it w/o interfering with my current active
> candidate pair.
>=20
> I think we need to do (1) and probably (2) now and (3) is a nice to have.

>From the point of view of RFC 5245, during an ICE Restart "media can contin=
ue to be sent to the previously validated pair." (9.1.1.1 and 11.1.1).  So =
3) should come for free, if implementations are written properly.

Here's the model of ICE as I understand it.  An endpoint is always free to =
gather as many or as few candidates as it wants, at any time before it star=
ts ICE or an ICE Restart.  This includes the currently-selected candidate, =
and any other previously-validated candidates that happen to still be live.

Thus, an ICE Restart doesn't have to be a "total re-ice", necessarily.  If =
you send (e.g.) your current selected candidate, and one other newly-discov=
ered candidate, as your candidate set for an ICE restart, the restart's con=
nectivity checking should be lighter-weight than one where you give the lis=
t of every possible candidate (which you presumably would do for the sessio=
n's initial ICE exchange).  But it's still a full and compliant ICE connect=
ivity check.

That said, ICE says that once you're in "ICE Completed" state, the only way=
 to change a component's selected candidate pair is by doing an ICE restart=
.

Trickle candidates are equivalent to candidates sent in an updated offer or=
 answer in the "ICE Running" state (9.1.2.1 / 9.2.2.1).  Thus, they're igno=
red once an endpoint enters "ICE Completed" state.


ICE leaves a lot of endpoint behavior up to endpoint implementors, and in t=
he webrtc context I think many of these behaviors will need to be controlla=
ble by the javascript applications.  I think the following APIs may be need=
ed.  (Some of them may exist already in the w3 draft, I haven't followed al=
l its details).

* Some way to know when something interesting has happened to network conne=
ctivity, above and beyond EKR's 1).  This includes a new network interface =
coming up, or even potentially (if possible) a something more interesting l=
ike a captive portal becoming unblocked.

* Some way to configure the set of interfaces on which address gathering sh=
ould be done, and the priority among them.  (This implies there also needs =
to be way to interrogate the set of interfaces.)

* Some way to start address gathering, both for the initial PeerConnection =
setup and for an ICE restart.  Note that this needs to be done both by the =
offerer and the answerer, and in many cases an answerer won't know it needs=
 to do so until it receives an offer with an ICE restart in it.

* If you're a controlling endpoint, whether to use regular or aggressive no=
mination.  For regular nomination, once one or more candidate pairs have be=
en validated, a way for the controlling endpoint to select the one to use. =
 (Note that the controlling endpoint may not necessarily be the offerer, if=
 you receive an ice-lite offer.)

--
Jonathan Lennox
jonathan@vidyo.com



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From: Martin Thomson <martin.thomson@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 14 June 2012 12:36, Justin Uberti <juberti@google.com> wrote:
> For "mute", stopping sending of RTP may trigger concealment on the remote
> side. We currently send silent audio when muted to prevent this.
> For "hold", you are notifying the remote side (e.g. a=inactive), so it makes
> sense to stop sending RTP.

Since signaling takes a little while to transit the network, it makes
sense for hold to act as mute (send silence) until it is accepted.  I
think that most users will expect hold to take immediate effect, at
least from the perspective of sending of media.

From matthew.kaufman@skype.net  Thu Jun 14 14:41:50 2012
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From: Matthew Kaufman <matthew.kaufman@skype.net>
To: Jonathan Lennox <jonathan@vidyo.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] What I think we need to do about ICE
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Subject: Re: [rtcweb] What I think we need to do about ICE
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Jonathan Lennox:

> That said, ICE says that once you're in "ICE Completed" state, the only w=
ay to change a component's selected candidate pair is by doing an ICE resta=
rt.
> Trickle candidates are equivalent to candidates sent in an updated offer =
or answer in the "ICE Running" state (9.1.2.1 / 9.2.2.1).  Thus, they're ig=
nored once an endpoint enters "ICE Completed" state.

These sound like great reasons to not bake a full ICE state machine into th=
e browser but instead provide primatives to Javascript which allow for the =
sending and receiving of STUN connectivity tests. That way an endpoint talk=
ing to another one that didn't care about this "ICE Completed" state could =
add innovation around rapid recovery from network changes.

Matthew Kaufman


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Thread-Topic: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 14 June 2012 22:14 Martin Thomson wrote:
>=20
> Since signaling takes a little while to transit the network, it makes
> sense for hold to act as mute (send silence) until it is accepted.  I
> think that most users will expect hold to take immediate effect, at
> least from the perspective of sending of media.

This is debatable and relates to the discussion about when to apply any cha=
nge to the session description really the change should probably not be app=
lied until the answer is received otherwise it might be necessary to restor=
e the previous session description depending on the result of the offer/ans=
wer exchange.  I would say don't apply the change until the answer is recei=
ved.

Regards
Andy



> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Fri Jun 15 03:34:36 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/14/2012 10:01 PM, Hutton, Andrew wrote:
> I agree with Justin that mute is a local action and should not stop the
> RTP or change the state of the remote track.
>
> Regards
> Andy

First, a short explanation of (my understanding of) what is in the API now:

* each track has a muted/unmuted and a enabled/disabled state
* enable state can be controlled by the application
* muted state can only be controlled by the browser
* muted overrides enabled (so a track that is enabled will still not 
play if it is muted)

The idea is that "mute" state is out of the control of the app (so the 
browser can override - and the user if the browser chrome allows). There 
has been some discussion indicating that the above is not sufficient and 
that we need to add more.

I think we're talking about different "mutes" in this thread.

One that is that you mute using e.g. a button on your headset. This 
would lead to audio samples with amplitude of zero being generated. This 
should not change any state on the track (locally or remotely) IMO, and 
silence/cn should be sent in the RTP.

Another is that the app has a "mute" button. Clicking this would 
(depending of the app) lead to the corresponding track being "disabled" 
in the local MediaStream. For the remote MediaStream, this is basically 
the same as "mute in the browser chrome" in the local case (since the 
source stops producing data), and I think the state there should change 
to "muted".

A third is that you mute the device via the browser chrome (I assume 
this possibility will exist). In this case the track state changes to 
muted, and I really think this should propagate to the other end of a 
PeerConnection.

Stefan


>
>
>
> On 14 Jun 2012, at 20:37, "Justin Uberti" <juberti@google.com
> <mailto:juberti@google.com>> wrote:
>
>> For "mute", stopping sending of RTP may trigger concealment on the
>> remote side. We currently send silent audio when muted to prevent this.
>> For "hold", you are notifying the remote side (e.g. a=inactive), so it
>> makes sense to stop sending RTP.
>>
>> On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansson LK
>> <stefan.lk.hakansson@ericsson.com
>> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>>
>>     On 06/14/2012 11:21 AM, Harald Alvestrand wrote:
>>
>>         Changing the subject line again, since we're discussing how to do
>>         mute, not whether to do it.....
>>
>>         On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
>>
>>             On 13 June 2012 15:37 Stefan Hakansson wrote:
>>
>>                 This is my personal understanding:
>>
>>                 - the API (and this part has been there for a long
>>                 time) allows
>>                 the application to disable/enable individual tracks of
>>                 MediaStreams - So say you have a MediaStream (one
>>                 audio track,
>>                 one video track in this example) that is rendered
>>                 using a video
>>                 element; disabling audio would lead to silence,
>>                 disabling video
>>                 to that the video stops - The question is: what should
>>                 happen if
>>                 the MediaStream is connected to a PeerConnection and
>>                 sent to a
>>                 peer? - using recvonly does not seem appropriate as
>>                 that would
>>                 stop all ssrc:s belonging to the same m-line - and the
>>                 app only
>>                 disabled one - one option could be to simply stop
>>                 sending the RTP
>>                 packets for this track. I don't know how reliable that
>>                 would be.
>>
>>             In the case of outgoing media then I think mute should simply
>>             disconnect the source (e.g. microphone) and there is no
>>             impact on
>>             RTP other than depending on the codec silence will be
>>             transmitted
>>             or SID frame etc. There is also no impact on RTCP which
>>             continues.
>>
>>             This is different to setting the direction to be recvonly
>>             which
>>             would completely stop the outgoing RTP packets and would
>>             normally
>>             be signaled to the remote party. This should of course be
>>             achieved
>>             using setLocalDescription.
>>
>>
>>         Just to add to the complexity of the "mute" case:
>>
>>         When an audio stream is muted, should there be comfort noise
>>         sent? If
>>         yes, what should determine the noise level?
>>
>>
>>     To me it feels as that ideally we should stop sending RTP (and
>>     make the other end know that the track is disabled). If we go to
>>     comfort noise state, the track in the MediaStream at the other end
>>     would still be active (even though what is played could be very
>>     low level comfort noise), but to reflect the local state the
>>     remote track should go to "muted", shouldn't it?
>>
>>
>>
>>
>>         (people who've never had to deal with comfort noise can safely
>>         tune
>>         out at this point)
>>
>>
>>
>>                 - I think there has been proposals signaling this in
>>                 SDP or in
>>                 RTCP, but I don't think there was ever any conclusion
>>
>>             If we are talking about a local mute then I don't see any
>>             need to
>>             signal this but again this is different to changing the
>>             direction
>>             attribute in SDP.
>>
>>                 For the other part, muting incoming media, this is
>>                 simple: the
>>                 app (at receiving browser) could just disable the
>>                 tracks it wants
>>                 in the incoming MediaStreams. A related question is if
>>                 this
>>                 should be signaled back to the sender (so it could
>>                 stop sending).
>>                 This is essentially a question of
>>                 efficiency/optimization, and
>>                 nothing stops the app from signaling that now using an app
>>                 specific protocol (perhaps carried over the data
>>                 channel), but
>>                 perhaps this should be brought into SDP or RTCP.
>>
>>             If the application wants to ask the remote party to pause
>>             sending
>>             of RTP then it should surely do this in SDP by setting the
>>             direction attribute appropriately (i.e. sendonly or inactive).
>>
>>             Regards Andy
>>
>>             _________________________________________________ rtcweb
>>             mailing
>>             list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>             https://www.ietf.org/mailman/__listinfo/rtcweb
>>             <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>>         _________________________________________________ rtcweb
>>         mailing list
>>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>         https://www.ietf.org/mailman/__listinfo/rtcweb
>>         <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>>     _________________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/__listinfo/rtcweb
>>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Fri Jun 15 03:34:50 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Martin Thomson <martin.thomson@gmail.com>, Justin Uberti <juberti@google.com>
Date: Fri, 15 Jun 2012 12:34:46 +0200
Thread-Topic: [rtcweb] Mute implementations (Re: More Comments	on draft-ietf-rtcweb-use-cases-and-requirements-07)
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mute implementations (Re: More Comments	on	draft-ietf-rtcweb-use-cases-and-requirements-07)
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I'd say it's an implementation issue. If I want to put a call on hold, I wi=
ll stop sending media no matter what the response to the updated offer is.

Also, as we discussed in SIPCORE when working on the re-INVITE handling spe=
c, clients may not even be able to fallback to the before-the-new-offer-was=
-sent state (probably not an issue for hold, but in general).

However,  I don't think it's up to us to specify how services work :) We sh=
ould focus on what happens when no media is sent - for whatever reason.

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Hutton, Andrew
Sent: 15. kes=E4kuuta 2012 11:18
To: Martin Thomson; Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf=
-rtcweb-use-cases-and-requirements-07)

On 14 June 2012 22:14 Martin Thomson wrote:
>=20
> Since signaling takes a little while to transit the network, it makes=20
> sense for hold to act as mute (send silence) until it is accepted.  I=20
> think that most users will expect hold to take immediate effect, at=20
> least from the perspective of sending of media.

This is debatable and relates to the discussion about when to apply any cha=
nge to the session description really the change should probably not be app=
lied until the answer is received otherwise it might be necessary to restor=
e the previous session description depending on the result of the offer/ans=
wer exchange.  I would say don't apply the change until the answer is recei=
ved.

Regards
Andy



> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From christer.holmberg@ericsson.com  Fri Jun 15 03:36:35 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Martin Thomson <martin.thomson@gmail.com>, Justin Uberti <juberti@google.com>
Date: Fri, 15 Jun 2012 12:36:32 +0200
Thread-Topic: [rtcweb] Mute implementations (Re: More	Comments	on draft-ietf-rtcweb-use-cases-and-requirements-07)
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mute implementations (Re: More	Comments	on	draft-ietf-rtcweb-use-cases-and-requirements-07)
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I think we DO, however, need to specify what happens if a new offer is reje=
cted, and if the browser is not able to fallback to the previous state.

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: 15. kes=E4kuuta 2012 13:35
To: Hutton, Andrew; Martin Thomson; Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf=
-rtcweb-use-cases-and-requirements-07)

I'd say it's an implementation issue. If I want to put a call on hold, I wi=
ll stop sending media no matter what the response to the updated offer is.

Also, as we discussed in SIPCORE when working on the re-INVITE handling spe=
c, clients may not even be able to fallback to the before-the-new-offer-was=
-sent state (probably not an issue for hold, but in general).

However,  I don't think it's up to us to specify how services work :) We sh=
ould focus on what happens when no media is sent - for whatever reason.

Regards,

Christer

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Hutton, Andrew
Sent: 15. kes=E4kuuta 2012 11:18
To: Martin Thomson; Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf=
-rtcweb-use-cases-and-requirements-07)

On 14 June 2012 22:14 Martin Thomson wrote:
>=20
> Since signaling takes a little while to transit the network, it makes=20
> sense for hold to act as mute (send silence) until it is accepted.  I=20
> think that most users will expect hold to take immediate effect, at=20
> least from the perspective of sending of media.

This is debatable and relates to the discussion about when to apply any cha=
nge to the session description really the change should probably not be app=
lied until the answer is received otherwise it might be necessary to restor=
e the previous session description depending on the result of the offer/ans=
wer exchange.  I would say don't apply the change until the answer is recei=
ved.

Regards
Andy



> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From keith.drage@alcatel-lucent.com  Fri Jun 15 03:50:45 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
Date: Fri, 15 Jun 2012 12:50:37 +0200
Thread-Topic: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on	draft-ietf-rtcweb-use-cases-and-requirements-07)
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I agree it is important to agree on what resources are released (and theref=
ore made available for some other purpose) as a result of the action. This =
confusion has existed ever since the PSTN invented a hold service.

Is the function merely to mute the transducers locally, or is the function =
to make resources (any of media, signaling, transducers, etc) available for=
 some other purpose.


Regards

Keith

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Stefan Hakansson LK
Sent: 15 June 2012 11:35
To: Hutton, Andrew
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf=
-rtcweb-use-cases-and-requirements-07)

On 06/14/2012 10:01 PM, Hutton, Andrew wrote:
> I agree with Justin that mute is a local action and should not stop the
> RTP or change the state of the remote track.
>
> Regards
> Andy

First, a short explanation of (my understanding of) what is in the API now:

* each track has a muted/unmuted and a enabled/disabled state
* enable state can be controlled by the application
* muted state can only be controlled by the browser
* muted overrides enabled (so a track that is enabled will still not=20
play if it is muted)

The idea is that "mute" state is out of the control of the app (so the=20
browser can override - and the user if the browser chrome allows). There=20
has been some discussion indicating that the above is not sufficient and=20
that we need to add more.

I think we're talking about different "mutes" in this thread.

One that is that you mute using e.g. a button on your headset. This=20
would lead to audio samples with amplitude of zero being generated. This=20
should not change any state on the track (locally or remotely) IMO, and=20
silence/cn should be sent in the RTP.

Another is that the app has a "mute" button. Clicking this would=20
(depending of the app) lead to the corresponding track being "disabled"=20
in the local MediaStream. For the remote MediaStream, this is basically=20
the same as "mute in the browser chrome" in the local case (since the=20
source stops producing data), and I think the state there should change=20
to "muted".

A third is that you mute the device via the browser chrome (I assume=20
this possibility will exist). In this case the track state changes to=20
muted, and I really think this should propagate to the other end of a=20
PeerConnection.

Stefan


>
>
>
> On 14 Jun 2012, at 20:37, "Justin Uberti" <juberti@google.com
> <mailto:juberti@google.com>> wrote:
>
>> For "mute", stopping sending of RTP may trigger concealment on the
>> remote side. We currently send silent audio when muted to prevent this.
>> For "hold", you are notifying the remote side (e.g. a=3Dinactive), so it
>> makes sense to stop sending RTP.
>>
>> On Thu, Jun 14, 2012 at 9:10 AM, Stefan Hakansson LK
>> <stefan.lk.hakansson@ericsson.com
>> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>>
>>     On 06/14/2012 11:21 AM, Harald Alvestrand wrote:
>>
>>         Changing the subject line again, since we're discussing how to d=
o
>>         mute, not whether to do it.....
>>
>>         On 06/14/2012 11:15 AM, Hutton, Andrew wrote:
>>
>>             On 13 June 2012 15:37 Stefan Hakansson wrote:
>>
>>                 This is my personal understanding:
>>
>>                 - the API (and this part has been there for a long
>>                 time) allows
>>                 the application to disable/enable individual tracks of
>>                 MediaStreams - So say you have a MediaStream (one
>>                 audio track,
>>                 one video track in this example) that is rendered
>>                 using a video
>>                 element; disabling audio would lead to silence,
>>                 disabling video
>>                 to that the video stops - The question is: what should
>>                 happen if
>>                 the MediaStream is connected to a PeerConnection and
>>                 sent to a
>>                 peer? - using recvonly does not seem appropriate as
>>                 that would
>>                 stop all ssrc:s belonging to the same m-line - and the
>>                 app only
>>                 disabled one - one option could be to simply stop
>>                 sending the RTP
>>                 packets for this track. I don't know how reliable that
>>                 would be.
>>
>>             In the case of outgoing media then I think mute should simpl=
y
>>             disconnect the source (e.g. microphone) and there is no
>>             impact on
>>             RTP other than depending on the codec silence will be
>>             transmitted
>>             or SID frame etc. There is also no impact on RTCP which
>>             continues.
>>
>>             This is different to setting the direction to be recvonly
>>             which
>>             would completely stop the outgoing RTP packets and would
>>             normally
>>             be signaled to the remote party. This should of course be
>>             achieved
>>             using setLocalDescription.
>>
>>
>>         Just to add to the complexity of the "mute" case:
>>
>>         When an audio stream is muted, should there be comfort noise
>>         sent? If
>>         yes, what should determine the noise level?
>>
>>
>>     To me it feels as that ideally we should stop sending RTP (and
>>     make the other end know that the track is disabled). If we go to
>>     comfort noise state, the track in the MediaStream at the other end
>>     would still be active (even though what is played could be very
>>     low level comfort noise), but to reflect the local state the
>>     remote track should go to "muted", shouldn't it?
>>
>>
>>
>>
>>         (people who've never had to deal with comfort noise can safely
>>         tune
>>         out at this point)
>>
>>
>>
>>                 - I think there has been proposals signaling this in
>>                 SDP or in
>>                 RTCP, but I don't think there was ever any conclusion
>>
>>             If we are talking about a local mute then I don't see any
>>             need to
>>             signal this but again this is different to changing the
>>             direction
>>             attribute in SDP.
>>
>>                 For the other part, muting incoming media, this is
>>                 simple: the
>>                 app (at receiving browser) could just disable the
>>                 tracks it wants
>>                 in the incoming MediaStreams. A related question is if
>>                 this
>>                 should be signaled back to the sender (so it could
>>                 stop sending).
>>                 This is essentially a question of
>>                 efficiency/optimization, and
>>                 nothing stops the app from signaling that now using an a=
pp
>>                 specific protocol (perhaps carried over the data
>>                 channel), but
>>                 perhaps this should be brought into SDP or RTCP.
>>
>>             If the application wants to ask the remote party to pause
>>             sending
>>             of RTP then it should surely do this in SDP by setting the
>>             direction attribute appropriately (i.e. sendonly or inactive=
).
>>
>>             Regards Andy
>>
>>             _________________________________________________ rtcweb
>>             mailing
>>             list rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>             https://www.ietf.org/mailman/__listinfo/rtcweb
>>             <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>>         _________________________________________________ rtcweb
>>         mailing list
>>         rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>         https://www.ietf.org/mailman/__listinfo/rtcweb
>>         <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>>     _________________________________________________
>>     rtcweb mailing list
>>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>     https://www.ietf.org/mailman/__listinfo/rtcweb
>>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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From: Martin Thomson <martin.thomson@gmail.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 15 June 2012 03:34, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
> I don't think it's up to us to specify how services work :) We should focus on what happens when no media is sent - for whatever reason.

Take that and frame it.

From randell-ietf@jesup.org  Fri Jun 15 13:42:36 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 6/14/2012 3:36 PM, Justin Uberti wrote:
> For "mute", stopping sending of RTP may trigger concealment on the
> remote side. We currently send silent audio when muted to prevent this.
> For "hold", you are notifying the remote side (e.g. a=inactive), so it
> makes sense to stop sending RTP.

I would also suggest that for video-mute that instead of stopping video 
or sending black, the application should send a "mute image".  This 
could be done by changing the video source to be a fixed image (or small 
looped animation from a <video> element.  This greatly reduces confusion 
for users ("you just went black!" "where did you go?", etc)

This is largely an application issue, so long as you can redirect the 
source of input MediaStream for the outgoing video.

See the message stream around 2/13/2012 titled "Mute and MediaStream 
repointing".  One message:
http://www.w3.org/mid/4F391CFB.6030806@jesup.org

A quote:
------------------------------------

Since Hold is a slightly different concept, I believe 'muting' an 
outgoing stream should generate silence for audio, and some type of 
'muted' video message.  We used a camera icon with a circle-slash. 
Bandwidth for a static image can be dropped almost as much as you can 
for black.  I dislike (strongly) using 'black' to signal Mute because of 
codecs and the like in the path (and things like insertions of video 
'bugs' or other manipulation after the MediaStream generation (such as 
through the MediaStream Processing API proposal) might mess it up).

** This brings up an important point about repointing a MediaStream **

However, exactly what should be sent for Mute is probably an application 
issue - for video mute, the application should probably replace one 
source with a static (or not) image.  This speaks more to the 
MediaStream Processing API, since you don't want to renegotiate in order 
to change the tracks - you want to change the source of an existing track.

So, basically you want a MediaStream from getUserMedia() (call it 
MSCamera), a MediaStream from a static image or pre-recorded video (call 
it MSMute), and a MediaStream that can be either (MSMerged).  We can't 
send a track, just a stream, and again we don't want to renegotiate, so 
we need to create MSMerged from either MSCamera or MSMute.  The 
MediaStream Processing API should be able to handle this, much as it can 
handle smooth cross-fades of audio and stitching video clips together - 
see demos at:

http://robert.ocallahan.org/2012/01/mediastreams-processing-demos.html

This could be a canned video (or audio), a fixed image, or something 
algorithmically generated by the application using Workers ("You are 
second in line" "Your wait time is 3 minutes" "Please see our new fall 
line at http://bigstore.com/"), and equivalently-generated audio.

This would allow you to change the outgoing video source without 
triggering re-negotiations, which is very important.

Alternatively we'd need a way to override the source for individual 
tracks (and restore the default value), and a way to create a track as a 
discrete object, etc.

---------------------------------
end quote

-- 
Randell Jesup
randell-ietf@jesup.org

From martin.thomson@gmail.com  Fri Jun 15 13:49:49 2012
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Date: Fri, 15 Jun 2012 13:49:47 -0700
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To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 15 June 2012 13:42, Randell Jesup <randell-ietf@jesup.org> wrote:
> I would also suggest that for video-mute that instead of stopping video or
> sending black, the application should send a "mute image".

The question I have is whether there is: a) a fixed image/video that
is serialized to RTP b) video "comfort noise", or c) local mute
image/video playback.  If it can't be c, then you might need to
explain yourself.

From randell-ietf@jesup.org  Fri Jun 15 13:51:30 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 6/14/2012 4:13 PM, Martin Thomson wrote:
> On 14 June 2012 12:36, Justin Uberti<juberti@google.com>  wrote:
>> For "mute", stopping sending of RTP may trigger concealment on the remote
>> side. We currently send silent audio when muted to prevent this.
>> For "hold", you are notifying the remote side (e.g. a=inactive), so it makes
>> sense to stop sending RTP.
>
> Since signaling takes a little while to transit the network, it makes
> sense for hold to act as mute (send silence) until it is accepted.  I
> think that most users will expect hold to take immediate effect, at
> least from the perspective of sending of media.

Absolutely any Hold operation should Mute audio and video until the 
actual offer/answer is complete.  Been there, done that. :-)

-- 
Randell Jesup
randell-ietf@jesup.org

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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 6/15/2012 4:49 PM, Martin Thomson wrote:
> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>  wrote:
>> I would also suggest that for video-mute that instead of stopping video or
>> sending black, the application should send a "mute image".
>
> The question I have is whether there is: a) a fixed image/video that
> is serialized to RTP b) video "comfort noise", or c) local mute
> image/video playback.  If it can't be c, then you might need to
> explain yourself.

Either we build it into the system (which I dislike - SetMuteImage() or 
whathaveyou, though we could), or we put it in the application domain 
and suggest or assist implementation.

The simple way to handle Mute is to revector the input to a "mute" or 
"hold" source, and they may not be the same (hold would have "Please 
Hold" and music, perhaps; mute would have silence and "Muted" or a 
camera with a circle/slash, etc).  You can also signal that out of band 
to let the other end take smart action (remove muted video image, 
display "muted" next to a name, etc).  This can be done via 
DataChannels, for example.  You may then reinvite for Hold to make it 
one-way, but that's mostly optional.

-- 
Randell Jesup
randell-ietf@jesup.org

From harald@alvestrand.no  Sat Jun 16 12:56:09 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments on draft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/15/2012 11:18 PM, Randell Jesup wrote:
> On 6/15/2012 4:49 PM, Martin Thomson wrote:
>> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>  wrote:
>>> I would also suggest that for video-mute that instead of stopping 
>>> video or
>>> sending black, the application should send a "mute image".
>>
>> The question I have is whether there is: a) a fixed image/video that
>> is serialized to RTP b) video "comfort noise", or c) local mute
>> image/video playback.  If it can't be c, then you might need to
>> explain yourself.
>
> Either we build it into the system (which I dislike - SetMuteImage() 
> or whathaveyou, though we could), or we put it in the application 
> domain and suggest or assist implementation.
>
> The simple way to handle Mute is to revector the input to a "mute" or 
> "hold" source, and they may not be the same (hold would have "Please 
> Hold" and music, perhaps; mute would have silence and "Muted" or a 
> camera with a circle/slash, etc).  You can also signal that out of 
> band to let the other end take smart action (remove muted video image, 
> display "muted" next to a name, etc).  This can be done via 
> DataChannels, for example.  You may then reinvite for Hold to make it 
> one-way, but that's mostly optional.
The resulting requirements seem to me to be:

1) To be able to use a still image as source for a video stream
2) To be able to change the source of a video stream after its creation

Both are requirements on the GetUserMedia API. Can we link these to a 
specific use case, or do we need to modify the use case document so that 
the requirements are clear?

              Harald





From Jim.Barnett@genesyslab.com  Sat Jun 16 13:08:08 2012
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Thread-Topic: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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References: <5A2F246E-9D77-4B72-B3F1-681B00FA99FD@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F11B52E@ESESSCMS0362.eemea.ericsson.se><AAB6A6B6-BEC7-436C-B1EB-2C117596C7C6@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F2631AC@ESESSCMS0362.eemea.ericsson.se><9F33F40F6F2CD847824537F3C4E37DDF021B53@MCHP04MSX.global-ad.net><CA+9kkMCL1k3x2hYi632nDU2g6CtOq3O7risDi38+7FTeWkxcGA@mail.gmail.com><9F33F40F6F2CD847824537F3C4E37DDF0223ED@MCHP04MSX.global-ad.net><4FD89773.3090506@ericsson.com><9F33F40F6F2CD847824537F3C4E37DDF02594D@MCHP04MSX.global-ad.net><4FD9AD1E.6060504@alvestrand.no> <4FD9E2BE.2080907@ericsson.com><CAOJ7v-18EynuscX3tLRryzqKxyXwYzES9K4zroE-5=FL4o8WzA@mail.gmail.com><4FDB9E20.8070408@jesup.org><CABkgnnVqeyuayFsiA7YD+OFnyCi8tqode6VdypeaHKT+Z73KWw@mail.gmail.com><4FDBA6AC.2080107@jesup.org> <4FDCE4D4.7070909@alvestrand.no>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Harald Alvestrand" <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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Harald,
  Your description below makes me nervous:  with getUserMedia, the user
grants permission to _specific_ media streams (for example audio, or
front camera, but not rear).  If the system wants " to change the source
of a video stream after its creation", shouldn't it be required to ask
permission again?  In this case,  common sense says that it's not
necessary, but that's because we're not changing to an arbitrary
alternative the source (e.g., suddenly switching on the rear camera) but
rather substituting some sort of 'null' or 'dummy' source.  Can we think
of a different way to phrase it?

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Harald Alvestrand
Sent: Saturday, June 16, 2012 3:56 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)

On 06/15/2012 11:18 PM, Randell Jesup wrote:
> On 6/15/2012 4:49 PM, Martin Thomson wrote:
>> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>  wrote:
>>> I would also suggest that for video-mute that instead of stopping=20
>>> video or sending black, the application should send a "mute image".
>>
>> The question I have is whether there is: a) a fixed image/video that=20
>> is serialized to RTP b) video "comfort noise", or c) local mute=20
>> image/video playback.  If it can't be c, then you might need to=20
>> explain yourself.
>
> Either we build it into the system (which I dislike - SetMuteImage()=20
> or whathaveyou, though we could), or we put it in the application=20
> domain and suggest or assist implementation.
>
> The simple way to handle Mute is to revector the input to a "mute" or=20
> "hold" source, and they may not be the same (hold would have "Please=20
> Hold" and music, perhaps; mute would have silence and "Muted" or a=20
> camera with a circle/slash, etc).  You can also signal that out of=20
> band to let the other end take smart action (remove muted video image,

> display "muted" next to a name, etc).  This can be done via=20
> DataChannels, for example.  You may then reinvite for Hold to make it=20
> one-way, but that's mostly optional.
The resulting requirements seem to me to be:

1) To be able to use a still image as source for a video stream
2) To be able to change the source of a video stream after its creation

Both are requirements on the GetUserMedia API. Can we link these to a
specific use case, or do we need to modify the use case document so that
the requirements are clear?

              Harald




_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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References: <5A2F246E-9D77-4B72-B3F1-681B00FA99FD@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F11B52E@ESESSCMS0362.eemea.ericsson.se><AAB6A6B6-BEC7-436C-B1EB-2C117596C7C6@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F2631AC@ESESSCMS0362.eemea.ericsson.se><9F33F40F6F2CD847824537F3C4E37DDF021B53@MCHP04MSX.global-ad.net><CA+9kkMCL1k3x2hYi632nDU2g6CtOq3O7risDi38+7FTeWkxcGA@mail.gmail.com><9F33F40F6F2CD847824537F3C4E37DDF0223ED@MCHP04MSX.global-ad.net><4FD89773.3090506@ericsson.com><9F33F40F6F2CD847824537F3C4E37DDF02594D@MCHP04MSX.global-ad.net><4FD9AD1E.6060504@alvestrand.no> <4FD9E2BE.2080907@ericsson.com><CAOJ7v-18EynuscX3tLRryzqKxyXwYzES9K4zroE-5=FL4o8WzA@mail.gmail.com><4FDB9E20.8070408@jesup.org><CABkgnnVqeyuayFsiA7YD+OFnyCi8tqode6VdypeaHKT+Z73KWw@mail.gmail.com><4FDBA6AC.2080107@jesup.org> <4FDCE4D4.7070909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8106581BE5@NAHALD.us.int.genesyslab.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/16/2012 10:07 PM, Jim Barnett wrote:
> Harald,
>    Your description below makes me nervous:  with getUserMedia, the user
> grants permission to _specific_ media streams (for example audio, or
> front camera, but not rear).
Yes, he grants permission to use of the camera. That permission persists 
until <we have to define this point in time>. There's some interesting 
language there now about "strong" and "weak" references to video streams.

>    If the system wants " to change the source
> of a video stream after its creation", shouldn't it be required to ask
> permission again?  In this case,  common sense says that it's not
> necessary, but that's because we're not changing to an arbitrary
> alternative the source (e.g., suddenly switching on the rear camera) but
> rather substituting some sort of 'null' or 'dummy' source.  Can we think
> of a different way to phrase it?
There are multiple ways to implement a requirement, as always.

One possibility is the simple one:

getUserMedia(..., function(strream) {
       outgoingstream = stream;
})
getStreamFromPicture(image, function(stream) {
       
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

whatever the "replace" call does.
There may be a requirement that we keep a reference around that keeps 
the video stream, so that we can restore it, in order that the camera 
not close and can be used later:

getUserMedia(..., function(strream) {
       copyOfCameraStream = stream;
       outgoingstream = MediaStream(stream.audioTracks, stream.videoTracks);

})

getStreamFromPicture(image, function(stream) {
       
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

restorePicture = function() {
      
outgoingtstream.videoStreams(0).replaceContent(copyOfCameraStream.video.Streams(0));
}

But there are surely multiple other ways to define it. What I want to 
make sure we have done first is to capture the requirement accurately.

>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Harald Alvestrand
> Sent: Saturday, June 16, 2012 3:56 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Mute implementations (Re: More Comments
> ondraft-ietf-rtcweb-use-cases-and-requirements-07)
>
> On 06/15/2012 11:18 PM, Randell Jesup wrote:
>> On 6/15/2012 4:49 PM, Martin Thomson wrote:
>>> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>   wrote:
>>>> I would also suggest that for video-mute that instead of stopping
>>>> video or sending black, the application should send a "mute image".
>>> The question I have is whether there is: a) a fixed image/video that
>>> is serialized to RTP b) video "comfort noise", or c) local mute
>>> image/video playback.  If it can't be c, then you might need to
>>> explain yourself.
>> Either we build it into the system (which I dislike - SetMuteImage()
>> or whathaveyou, though we could), or we put it in the application
>> domain and suggest or assist implementation.
>>
>> The simple way to handle Mute is to revector the input to a "mute" or
>> "hold" source, and they may not be the same (hold would have "Please
>> Hold" and music, perhaps; mute would have silence and "Muted" or a
>> camera with a circle/slash, etc).  You can also signal that out of
>> band to let the other end take smart action (remove muted video image,
>> display "muted" next to a name, etc).  This can be done via
>> DataChannels, for example.  You may then reinvite for Hold to make it
>> one-way, but that's mostly optional.
> The resulting requirements seem to me to be:
>
> 1) To be able to use a still image as source for a video stream
> 2) To be able to change the source of a video stream after its creation
>
> Both are requirements on the GetUserMedia API. Can we link these to a
> specific use case, or do we need to modify the use case document so that
> the requirements are clear?
>
>                Harald
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: [rtcweb] REMINDER: Call for Papers, IAB/IRTF Workshop on Congestion Control for Interactive Real-Time Communication, Vancouver, Canada, July 28, 2012
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Call for Papers

 

The IAB and IRTF will hold a workshop on "Congestion Control for Interactive
Real-Time Communication" in Vancouver, Canada on Saturday, July 28th, 2012
prior to the IETF-84 meeting.  More information on the workshop is available
at  http://www.iab.org/cc-workshop/ 
 
This is an invitation-only workshop. 
 
Every prospective workshop participant must submit a position paper. The
position paper reflects your views on a particular area of the workshop
theme. We are interested in your opinion as an expert rather than your
official company position.  Keep your submission short and focused.  If your
expertise allows you to write about a range of topics, pick the one topic
you think would bring the most value to the workshop. Authors of accepted
papers will be invited to the workshop.  Papers up to 3 pages, formatted in
HTML, PDF, or plain text (for example, as a submitted Internet-Draft) are
ideal.  Accepted position papers will be published. 
 
Please use the following Website for submitting your papers: 
http://iab.org/ccirtc/paper?p=new
 
Important Dates 
 
Position paper submission deadline: June 23, 2012 
Notification to paper authors: June 30, 2012 
Workshop date: July 28, 2012 
 
IPR Policy 
 
Due to the close relationship to ongoing IRTF and IETF work, the IPR 
policies described in RFC 5378 and RFC 3979 apply, both to submitted 
position papers and to discussions at the workshop. 
 
Privacy Notice 
 
Paper submissions have to contain a name and an email address. We use this
information to subscribe you to a mailing list for sharing workshop related
information prior to the workshop (e.g., updates regarding the meeting
venue, feedback on the position paper submissions). This specially created
mailing list is only used for the duration of the workshop and will be
deleted after the publication of the workshop report (which may take several
months). You may at any time decide to unsubscribe from the mailing list (at
your risk of missing information workshop related postings).  
 
Your name will be listed in the meeting minutes and the workshop report. We
will also publish all accepted position papers. 
 
There will be an audio recording of the workshop discussion. This workshop
recording will not be distributed to the workshop participants nor to the
public; the purpose of the recording is to ensure that the workshop report
and the meeting minutes accurately reflect the discussions.  
 
Meeting Venue 
 
The workshop will be held in Vancouver, Canada on Saturday, July 28th, 2012
prior to the IETF-84 meeting (see
http://www.ietf.org/meeting/84/index.html).  Additional details about the
meeting venue will be provided to authors of accepted papers.  

Sponsorship 
 
If you are interested to help us working towards better interactive media
congestion control mechanisms on the Internet (such as by making a
contribution towards catering costs and room rental), please contact us! 
 
Contact Information 

In case of questions please send email to mary.ietf.barnes at gmail.com 


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From Jim.Barnett@genesyslab.com  Sat Jun 16 14:58:58 2012
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Thread-Topic: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
Thread-Index: Ac1L/2XN6ybpngMORm2cIIOeRph0bQACwPsw
References: <5A2F246E-9D77-4B72-B3F1-681B00FA99FD@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F11B52E@ESESSCMS0362.eemea.ericsson.se><AAB6A6B6-BEC7-436C-B1EB-2C117596C7C6@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F2631AC@ESESSCMS0362.eemea.ericsson.se><9F33F40F6F2CD847824537F3C4E37DDF021B53@MCHP04MSX.global-ad.net><CA+9kkMCL1k3x2hYi632nDU2g6CtOq3O7risDi38+7FTeWkxcGA@mail.gmail.com><9F33F40F6F2CD847824537F3C4E37DDF0223ED@MCHP04MSX.global-ad.net><4FD89773.3090506@ericsson.com><9F33F40F6F2CD847824537F3C4E37DDF02594D@MCHP04MSX.global-ad.net><4FD9AD1E.6060504@alvestrand.no> <4FD9E2BE.2080907@ericsson.com><CAOJ7v-18EynuscX3tLRryzqKxyXwYzES9K4zroE-5=FL4o8WzA@mail.gmail.com><4FDB9E20.8070408@jesup.org><CABkgnnVqeyuayFsiA7YD+OFnyCi8tqode6VdypeaHKT+Z73KWw@mail.gmail.com><4FDBA6AC.2080107@jesup.org> <4FDCE4D4.7070909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8106581BE5@NAHALD.us.int.genesyslab.com> <4FDCEDC2.2040903@alvestrand.no>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Harald Alvestrand" <harald@alvestrand.no>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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It's capturing the requirement accurately that I'm concerned with.  I
think "To be able to change the source of a video stream after its
creation" is too broad, in the sense that someone might interpret it to
mean that the system could shift from the front to the rear camera
without asking permission.  For mute and hold, we want something like
"switch to a dummy source and back".  The source that we switch to in
cases like this is not a 'first-class' video source, but rather a
place-holder.  That's very different from switching to another live
source.

- Jim

-----Original Message-----
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Saturday, June 16, 2012 4:34 PM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)

On 06/16/2012 10:07 PM, Jim Barnett wrote:
> Harald,
>    Your description below makes me nervous:  with getUserMedia, the=20
> user grants permission to _specific_ media streams (for example audio,

> or front camera, but not rear).
Yes, he grants permission to use of the camera. That permission persists
until <we have to define this point in time>. There's some interesting
language there now about "strong" and "weak" references to video
streams.

>    If the system wants " to change the source of a video stream after=20
> its creation", shouldn't it be required to ask permission again?  In=20
> this case,  common sense says that it's not necessary, but that's=20
> because we're not changing to an arbitrary alternative the source=20
> (e.g., suddenly switching on the rear camera) but rather substituting=20
> some sort of 'null' or 'dummy' source.  Can we think of a different=20
> way to phrase it?
There are multiple ways to implement a requirement, as always.

One possibility is the simple one:

getUserMedia(..., function(strream) {
       outgoingstream =3D stream;
})
getStreamFromPicture(image, function(stream) {
      =20
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

whatever the "replace" call does.
There may be a requirement that we keep a reference around that keeps
the video stream, so that we can restore it, in order that the camera
not close and can be used later:

getUserMedia(..., function(strream) {
       copyOfCameraStream =3D stream;
       outgoingstream =3D MediaStream(stream.audioTracks,
stream.videoTracks);

})

getStreamFromPicture(image, function(stream) {
      =20
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

restorePicture =3D function() {
     =20
outgoingtstream.videoStreams(0).replaceContent(copyOfCameraStream.video.
Streams(0));
}

But there are surely multiple other ways to define it. What I want to
make sure we have done first is to capture the requirement accurately.

>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Harald Alvestrand
> Sent: Saturday, June 16, 2012 3:56 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Mute implementations (Re: More Comments
> ondraft-ietf-rtcweb-use-cases-and-requirements-07)
>
> On 06/15/2012 11:18 PM, Randell Jesup wrote:
>> On 6/15/2012 4:49 PM, Martin Thomson wrote:
>>> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>
wrote:
>>>> I would also suggest that for video-mute that instead of stopping=20
>>>> video or sending black, the application should send a "mute image".
>>> The question I have is whether there is: a) a fixed image/video that

>>> is serialized to RTP b) video "comfort noise", or c) local mute=20
>>> image/video playback.  If it can't be c, then you might need to=20
>>> explain yourself.
>> Either we build it into the system (which I dislike - SetMuteImage()=20
>> or whathaveyou, though we could), or we put it in the application=20
>> domain and suggest or assist implementation.
>>
>> The simple way to handle Mute is to revector the input to a "mute" or

>> "hold" source, and they may not be the same (hold would have "Please=20
>> Hold" and music, perhaps; mute would have silence and "Muted" or a=20
>> camera with a circle/slash, etc).  You can also signal that out of=20
>> band to let the other end take smart action (remove muted video=20
>> image, display "muted" next to a name, etc).  This can be done via=20
>> DataChannels, for example.  You may then reinvite for Hold to make it

>> one-way, but that's mostly optional.
> The resulting requirements seem to me to be:
>
> 1) To be able to use a still image as source for a video stream
> 2) To be able to change the source of a video stream after its=20
> creation
>
> Both are requirements on the GetUserMedia API. Can we link these to a=20
> specific use case, or do we need to modify the use case document so=20
> that the requirements are clear?
>
>                Harald
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From randell-ietf@jesup.org  Sat Jun 16 17:38:36 2012
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On 6/16/2012 5:58 PM, Jim Barnett wrote:
> It's capturing the requirement accurately that I'm concerned with.  I
> think "To be able to change the source of a video stream after its
> creation" is too broad, in the sense that someone might interpret it to
> mean that the system could shift from the front to the rear camera
> without asking permission.  For mute and hold, we want something like
> "switch to a dummy source and back".  The source that we switch to in
> cases like this is not a 'first-class' video source, but rather a
> place-holder.  That's very different from switching to another live
> source.

Which, by the way, we should be able to do.

MediaStreams need to be able to be repointed, switched, processed, etc, 
OR you need to be able to replace a MediaStream in the inputs to 
PeerConnection with another one transparently without renegotiation (and 
that misses a bunch of uses).

The MediaStream Processing provides a mechanism to do this for both 
audio and video, plus the ability to do more complex operations 
(cross-fades audio-and-video, frame-accurate source switches, etc).

If you don't have or use that, you need the ability to replace a track 
in a MediaStream with another track.

All we're talking about are mechanisms to shuffle existing or creatable 
tracks/streams; a new camera source would require a call to 
getUserMedia() and a dialog.  Once you had front and back, you could 
switch sources without effectively shutting down all media and restarting.

Simple one camera -> one video stream mechanisms are insufficient for 
application using these features.  If you force people, they'll roll 
their own using canvases, etc.

I want something generic and usable in lots of interesting ways, not a 
few special-purpose pre-coded mechanisms (for example, a 
track.setMuteImage() method).

-- 
Randell Jesup
randell-ietf@jesup.org


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From randell-ietf@jesup.org  Sun Jun 17 00:47:47 2012
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Subject: Re: [rtcweb] =?gb2312?b?tPC4tDogIE11dGUgaW1wbGVtZW50YXRpb25zIChSZTog?= =?gb2312?b?TW9yZSBDb21tZW50cyBvbmRyYWZ0LWlldGYtcnRjd2ViLXVzZS1jYXNlcy1h?= =?gb2312?b?bmQtcmVxdWlyZW1lbnRzLTA3KQ==?=
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On 6/17/2012 2:48 AM, Sunyang (Eric) wrote:
> You said:  "MediaStreams need to be able to be repointed, switched, processed, etc, OR you need to be able to replace a MediaStream in the inputs to PeerConnection with another one transparently without renegotiation (and that misses a bunch of uses)."
>
> But if the MediaStream  you choose to replace the existing one contain codecs that not supported by remote end, I think we still need renegotiation.
> Or you can make sure that the browser can choose the same codecs as default with the replaced one, I think browser can not do this.

It's possible that renegotiation might be required in *some* cases with
pre-encoded data from the camera, but not many (if any), since we need
to rate-control the data. We would need to decode the camera stream
(H.264 typically), then re-encode at the correct bitrate. Or we'll need
to change the camera bitrate targets on a frame-by-frame basis (unlikely).

-- 
Randell Jesup
randell-ietf@jesup.org


From harald@alvestrand.no  Sun Jun 17 01:01:08 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/16/2012 11:58 PM, Jim Barnett wrote:
> It's capturing the requirement accurately that I'm concerned with.  I
> think "To be able to change the source of a video stream after its
> creation" is too broad, in the sense that someone might interpret it to
> mean that the system could shift from the front to the rear camera
> without asking permission.
People certainly want to shift from the front to the rear camera, and 
people want to have the ability to get all that asking-for-permissions 
stuff out of the way before that.

I recommend the Media Task Force's "additional scenarios" for more 
usages of "local media".
Jjim, I know you know where this is, but for others:
http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scenarios.html

>    For mute and hold, we want something like
> "switch to a dummy source and back".  The source that we switch to in
> cases like this is not a 'first-class' video source, but rather a
> place-holder.  That's very different from switching to another live
> source.
ObLangDisagreement: Files, images and the output of video processors are 
very real sources, and should not be "second class".

Switching between live streams (that we have permission to access) is 
going to be required, but I think this belongs on the Media TF mailing list.

>
> - Jim
>
> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Saturday, June 16, 2012 4:34 PM
> To: Jim Barnett
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Mute implementations (Re: More Comments
> ondraft-ietf-rtcweb-use-cases-and-requirements-07)
>
> On 06/16/2012 10:07 PM, Jim Barnett wrote:
>> Harald,
>>     Your description below makes me nervous:  with getUserMedia, the
>> user grants permission to _specific_ media streams (for example audio,
>> or front camera, but not rear).
> Yes, he grants permission to use of the camera. That permission persists
> until<we have to define this point in time>. There's some interesting
> language there now about "strong" and "weak" references to video
> streams.
>
>>     If the system wants " to change the source of a video stream after
>> its creation", shouldn't it be required to ask permission again?  In
>> this case,  common sense says that it's not necessary, but that's
>> because we're not changing to an arbitrary alternative the source
>> (e.g., suddenly switching on the rear camera) but rather substituting
>> some sort of 'null' or 'dummy' source.  Can we think of a different
>> way to phrase it?
> There are multiple ways to implement a requirement, as always.
>
> One possibility is the simple one:
>
> getUserMedia(..., function(strream) {
>         outgoingstream = stream;
> })
> getStreamFromPicture(image, function(stream) {
>
> outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
> }..)
>
> whatever the "replace" call does.
> There may be a requirement that we keep a reference around that keeps
> the video stream, so that we can restore it, in order that the camera
> not close and can be used later:
>
> getUserMedia(..., function(strream) {
>         copyOfCameraStream = stream;
>         outgoingstream = MediaStream(stream.audioTracks,
> stream.videoTracks);
>
> })
>
> getStreamFromPicture(image, function(stream) {
>
> outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
> }..)
>
> restorePicture = function() {
>
> outgoingtstream.videoStreams(0).replaceContent(copyOfCameraStream.video.
> Streams(0));
> }
>
> But there are surely multiple other ways to define it. What I want to
> make sure we have done first is to capture the requirement accurately.
>
>> - Jim
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Harald Alvestrand
>> Sent: Saturday, June 16, 2012 3:56 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Mute implementations (Re: More Comments
>> ondraft-ietf-rtcweb-use-cases-and-requirements-07)
>>
>> On 06/15/2012 11:18 PM, Randell Jesup wrote:
>>> On 6/15/2012 4:49 PM, Martin Thomson wrote:
>>>> On 15 June 2012 13:42, Randell Jesup<randell-ietf@jesup.org>
> wrote:
>>>>> I would also suggest that for video-mute that instead of stopping
>>>>> video or sending black, the application should send a "mute image".
>>>> The question I have is whether there is: a) a fixed image/video that
>>>> is serialized to RTP b) video "comfort noise", or c) local mute
>>>> image/video playback.  If it can't be c, then you might need to
>>>> explain yourself.
>>> Either we build it into the system (which I dislike - SetMuteImage()
>>> or whathaveyou, though we could), or we put it in the application
>>> domain and suggest or assist implementation.
>>>
>>> The simple way to handle Mute is to revector the input to a "mute" or
>>> "hold" source, and they may not be the same (hold would have "Please
>>> Hold" and music, perhaps; mute would have silence and "Muted" or a
>>> camera with a circle/slash, etc).  You can also signal that out of
>>> band to let the other end take smart action (remove muted video
>>> image, display "muted" next to a name, etc).  This can be done via
>>> DataChannels, for example.  You may then reinvite for Hold to make it
>>> one-way, but that's mostly optional.
>> The resulting requirements seem to me to be:
>>
>> 1) To be able to use a still image as source for a video stream
>> 2) To be able to change the source of a video stream after its
>> creation
>>
>> Both are requirements on the GetUserMedia API. Can we link these to a
>> specific use case, or do we need to modify the use case document so
>> that the requirements are clear?
>>
>>                 Harald
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


--------------010701070805070108050808
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit

<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    On 06/16/2012 11:58 PM, Jim Barnett wrote:
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8106581BEA@NAHALD.us.int.genesyslab.com"
      type="cite">
      <pre wrap="">It's capturing the requirement accurately that I'm concerned with.  I
think "To be able to change the source of a video stream after its
creation" is too broad, in the sense that someone might interpret it to
mean that the system could shift from the front to the rear camera
without asking permission.</pre>
    </blockquote>
    People certainly want to shift from the front to the rear camera,
    and people want to have the ability to get all that
    asking-for-permissions stuff out of the way before that.<br>
    <br>
    I recommend the Media Task Force's "additional scenarios" for more
    usages of "local media".<br>
    Jjim, I know you know where this is, but for others:<br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scenarios.html">http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scenarios.html</a><br>
    <br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8106581BEA@NAHALD.us.int.genesyslab.com"
      type="cite">
      <pre wrap="">  For mute and hold, we want something like
"switch to a dummy source and back".  The source that we switch to in
cases like this is not a 'first-class' video source, but rather a
place-holder.  That's very different from switching to another live
source.</pre>
    </blockquote>
    ObLangDisagreement: Files, images and the output of video processors
    are very real sources, and should not be "second class".<br>
    <br>
    Switching between live streams (that we have permission to access)
    is going to be required, but I think this belongs on the Media TF
    mailing list.<br>
    <br>
    <blockquote
cite="mid:E17CAD772E76C742B645BD4DC602CD8106581BEA@NAHALD.us.int.genesyslab.com"
      type="cite">
      <pre wrap="">

- Jim

-----Original Message-----
From: Harald Alvestrand [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] 
Sent: Saturday, June 16, 2012 4:34 PM
To: Jim Barnett
Cc: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)

On 06/16/2012 10:07 PM, Jim Barnett wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">Harald,
   Your description below makes me nervous:  with getUserMedia, the 
user grants permission to _specific_ media streams (for example audio,
</pre>
      </blockquote>
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <pre wrap="">or front camera, but not rear).
</pre>
      </blockquote>
      <pre wrap="">Yes, he grants permission to use of the camera. That permission persists
until &lt;we have to define this point in time&gt;. There's some interesting
language there now about "strong" and "weak" references to video
streams.

</pre>
      <blockquote type="cite">
        <pre wrap="">   If the system wants " to change the source of a video stream after 
its creation", shouldn't it be required to ask permission again?  In 
this case,  common sense says that it's not necessary, but that's 
because we're not changing to an arbitrary alternative the source 
(e.g., suddenly switching on the rear camera) but rather substituting 
some sort of 'null' or 'dummy' source.  Can we think of a different 
way to phrase it?
</pre>
      </blockquote>
      <pre wrap="">There are multiple ways to implement a requirement, as always.

One possibility is the simple one:

getUserMedia(..., function(strream) {
       outgoingstream = stream;
})
getStreamFromPicture(image, function(stream) {
       
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

whatever the "replace" call does.
There may be a requirement that we keep a reference around that keeps
the video stream, so that we can restore it, in order that the camera
not close and can be used later:

getUserMedia(..., function(strream) {
       copyOfCameraStream = stream;
       outgoingstream = MediaStream(stream.audioTracks,
stream.videoTracks);

})

getStreamFromPicture(image, function(stream) {
       
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)

restorePicture = function() {
      
outgoingtstream.videoStreams(0).replaceContent(copyOfCameraStream.video.
Streams(0));
}

But there are surely multiple other ways to define it. What I want to
make sure we have done first is to capture the requirement accurately.

</pre>
      <blockquote type="cite">
        <pre wrap="">
- Jim

-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] On 
Behalf Of Harald Alvestrand
Sent: Saturday, June 16, 2012 3:56 PM
To: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)

On 06/15/2012 11:18 PM, Randell Jesup wrote:
</pre>
        <blockquote type="cite">
          <pre wrap="">On 6/15/2012 4:49 PM, Martin Thomson wrote:
</pre>
          <blockquote type="cite">
            <pre wrap="">On 15 June 2012 13:42, Randell Jesup<a class="moz-txt-link-rfc2396E" href="mailto:randell-ietf@jesup.org">&lt;randell-ietf@jesup.org&gt;</a>
</pre>
          </blockquote>
        </blockquote>
      </blockquote>
      <pre wrap="">wrote:
</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <blockquote type="cite">
            <blockquote type="cite">
              <pre wrap="">I would also suggest that for video-mute that instead of stopping 
video or sending black, the application should send a "mute image".
</pre>
            </blockquote>
            <pre wrap="">The question I have is whether there is: a) a fixed image/video that
</pre>
          </blockquote>
        </blockquote>
      </blockquote>
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <blockquote type="cite">
            <pre wrap="">is serialized to RTP b) video "comfort noise", or c) local mute 
image/video playback.  If it can't be c, then you might need to 
explain yourself.
</pre>
          </blockquote>
          <pre wrap="">Either we build it into the system (which I dislike - SetMuteImage() 
or whathaveyou, though we could), or we put it in the application 
domain and suggest or assist implementation.

The simple way to handle Mute is to revector the input to a "mute" or
</pre>
        </blockquote>
      </blockquote>
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <pre wrap="">"hold" source, and they may not be the same (hold would have "Please 
Hold" and music, perhaps; mute would have silence and "Muted" or a 
camera with a circle/slash, etc).  You can also signal that out of 
band to let the other end take smart action (remove muted video 
image, display "muted" next to a name, etc).  This can be done via 
DataChannels, for example.  You may then reinvite for Hold to make it
</pre>
        </blockquote>
      </blockquote>
      <pre wrap="">
</pre>
      <blockquote type="cite">
        <blockquote type="cite">
          <pre wrap="">one-way, but that's mostly optional.
</pre>
        </blockquote>
        <pre wrap="">The resulting requirements seem to me to be:

1) To be able to use a still image as source for a video stream
2) To be able to change the source of a video stream after its 
creation

Both are requirements on the GetUserMedia API. Can we link these to a 
specific use case, or do we need to modify the use case document so 
that the requirements are clear?

               Harald




_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
      <pre wrap="">
</pre>
    </blockquote>
    <br>
  </body>
</html>

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From: "Sunyang (Eric)" <eric.sun@huawei.com>
To: Harald Alvestrand <harald@alvestrand.no>, Jim Barnett <Jim.Barnett@genesyslab.com>
Thread-Topic: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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<body bgcolor=3D"white" lang=3D"ZH-CN" link=3D"blue" vlink=3D"purple">
<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Do we need=
 add one more scenario:<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The same v=
ideo B2B communication, and switch video stream between front and rear webc=
am?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The video =
streams displayed in the same video element, but source changed.<o:p></o:p>=
</span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">There is o=
n relative scenario in media capture scenarios, =A1=B0video diary=A1=AD.=A1=
=B1, but not not related with above mentioned scenario.<o:p></o:p></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Should we =
add it?<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;font-=
family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><o:p>&nbsp=
;</o:p></span></p>
<div>
<div style=3D"border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm">
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:SimSu=
n;color:windowtext">=B7=A2=BC=FE=C8=CB<span lang=3D"EN-US">:</span></span><=
/b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:SimSun;color:=
windowtext"> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
</span><b><span style=3D"font-size:10.0pt;font-family:SimSun;color:windowte=
xt">=B4=FA=B1=ED </span>
</b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-family:SimSun;color=
:windowtext">Harald Alvestrand<br>
</span><b><span style=3D"font-size:10.0pt;font-family:SimSun;color:windowte=
xt">=B7=A2=CB=CD=CA=B1=BC=E4<span lang=3D"EN-US">:</span></span></b><span l=
ang=3D"EN-US" style=3D"font-size:10.0pt;font-family:SimSun;color:windowtext=
"> 2012</span><span style=3D"font-size:10.0pt;font-family:SimSun;color:wind=
owtext">=C4=EA<span lang=3D"EN-US">6</span>=D4=C2<span lang=3D"EN-US">17</s=
pan>=C8=D5<span lang=3D"EN-US">
 16:01<br>
</span><b>=CA=D5=BC=FE=C8=CB<span lang=3D"EN-US">:</span></b><span lang=3D"=
EN-US"> Jim Barnett<br>
</span><b>=B3=AD=CB=CD<span lang=3D"EN-US">:</span></b><span lang=3D"EN-US"=
> rtcweb@ietf.org<br>
</span><b>=D6=F7=CC=E2<span lang=3D"EN-US">:</span></b><span lang=3D"EN-US"=
> Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-=
use-cases-and-requirements-07)<o:p></o:p></span></span></p>
</div>
</div>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">On 06/16/2012 11:58 PM, Jim Bar=
nett wrote:
<o:p></o:p></span></p>
<pre><span lang=3D"EN-US">It's capturing the requirement accurately that I'=
m concerned with.&nbsp; I<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">think &quot;To be able to change the source of a =
video stream after its<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">creation&quot; is too broad, in the sense that so=
meone might interpret it to<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">mean that the system could shift from the front t=
o the rear camera<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">without asking permission.<o:p></o:p></span></pre=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US">People certainly want to shift =
from the front to the rear camera, and people want to have the ability to g=
et all that asking-for-permissions stuff out of the way before that.<br>
<br>
I recommend the Media Task Force's &quot;additional scenarios&quot; for mor=
e usages of &quot;local media&quot;.<br>
Jjim, I know you know where this is, but for others:<br>
<a href=3D"http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scen=
arios.html">http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/sce=
narios.html</a><br>
<br>
<br>
<o:p></o:p></span></p>
<pre><span lang=3D"EN-US">&nbsp; For mute and hold, we want something like<=
o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">&quot;switch to a dummy source and back&quot;.&nb=
sp; The source that we switch to in<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">cases like this is not a 'first-class' video sour=
ce, but rather a<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">place-holder.&nbsp; That's very different from sw=
itching to another live<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">source.<o:p></o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-US">ObLangDisagreement: Files, imag=
es and the output of video processors are very real sources, and should not=
 be &quot;second class&quot;.<br>
<br>
Switching between live streams (that we have permission to access) is going=
 to be required, but I think this belongs on the Media TF mailing list.<br>
<br>
<br>
<o:p></o:p></span></p>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">- Jim<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">-----Original Message-----<o:p></o:p></span></pre=
>
<pre><span lang=3D"EN-US">From: Harald Alvestrand [<a href=3D"mailto:harald=
@alvestrand.no">mailto:harald@alvestrand.no</a>] <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Sent: Saturday, June 16, 2012 4:34 PM<o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">To: Jim Barnett<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a><o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Subject: Re: [rtcweb] Mute implementations (Re: M=
ore Comments<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">ondraft-ietf-rtcweb-use-cases-and-requirements-07=
)<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">On 06/16/2012 10:07 PM, Jim Barnett wrote:<o:p></=
o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">Harald,<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp; Your description below makes me nerv=
ous:&nbsp; with getUserMedia, the <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">user grants permission to _specific_ media stream=
s (for example audio,<o:p></o:p></span></pre>
</blockquote>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">or front camera, but not rear).<o:p></o:p></span>=
</pre>
</blockquote>
<pre><span lang=3D"EN-US">Yes, he grants permission to use of the camera. T=
hat permission persists<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">until &lt;we have to define this point in time&gt=
;. There's some interesting<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">language there now about &quot;strong&quot; and &=
quot;weak&quot; references to video<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">streams.<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">&nbsp;&nbsp; If the system wants &quot; to change=
 the source of a video stream after <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">its creation&quot;, shouldn't it be required to a=
sk permission again?&nbsp; In <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">this case,&nbsp; common sense says that it's not =
necessary, but that's <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">because we're not changing to an arbitrary altern=
ative the source <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">(e.g., suddenly switching on the rear camera) but=
 rather substituting <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">some sort of 'null' or 'dummy' source.&nbsp; Can =
we think of a different <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">way to phrase it?<o:p></o:p></span></pre>
</blockquote>
<pre><span lang=3D"EN-US">There are multiple ways to implement a requiremen=
t, as always.<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">One possibility is the simple one:<o:p></o:p></sp=
an></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">getUserMedia(..., function(strream) {<o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; outgoingstre=
am =3D stream;<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">})<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">getStreamFromPicture(image, function(stream) {<o:=
p></o:p></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">outgoingstream.videoStreams(0).replaceContent(str=
eam.videoStreams(0));<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">}..)<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">whatever the &quot;replace&quot; call does.<o:p><=
/o:p></span></pre>
<pre><span lang=3D"EN-US">There may be a requirement that we keep a referen=
ce around that keeps<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">the video stream, so that we can restore it, in o=
rder that the camera<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">not close and can be used later:<o:p></o:p></span=
></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">getUserMedia(..., function(strream) {<o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; copyOfCamera=
Stream =3D stream;<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; outgoingstre=
am =3D MediaStream(stream.audioTracks,<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">stream.videoTracks);<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">})<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">getStreamFromPicture(image, function(stream) {<o:=
p></o:p></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;<o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">outgoingstream.videoStreams(0).replaceContent(str=
eam.videoStreams(0));<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">}..)<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">restorePicture =3D function() {<o:p></o:p></span>=
</pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <o:p></o:p></span>=
</pre>
<pre><span lang=3D"EN-US">outgoingtstream.videoStreams(0).replaceContent(co=
pyOfCameraStream.video.<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Streams(0));<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">}<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">But there are surely multiple other ways to defin=
e it. What I want to<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">make sure we have done first is to capture the re=
quirement accurately.<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">- Jim<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">-----Original Message-----<o:p></o:p></span></pre=
>
<pre><span lang=3D"EN-US">From: <a href=3D"mailto:rtcweb-bounces@ietf.org">=
rtcweb-bounces@ietf.org</a> [<a href=3D"mailto:rtcweb-bounces@ietf.org">mai=
lto:rtcweb-bounces@ietf.org</a>] On <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Behalf Of Harald Alvestrand<o:p></o:p></span></pr=
e>
<pre><span lang=3D"EN-US">Sent: Saturday, June 16, 2012 3:56 PM<o:p></o:p><=
/span></pre>
<pre><span lang=3D"EN-US">To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a><o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Subject: Re: [rtcweb] Mute implementations (Re: M=
ore Comments<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">ondraft-ietf-rtcweb-use-cases-and-requirements-07=
)<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">On 06/15/2012 11:18 PM, Randell Jesup wrote:<o:p>=
</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">On 6/15/2012 4:49 PM, Martin Thomson wrote:<o:p><=
/o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">On 15 June 2012 13:42, Randell Jesup<a href=3D"ma=
ilto:randell-ietf@jesup.org">&lt;randell-ietf@jesup.org&gt;</a><o:p></o:p><=
/span></pre>
</blockquote>
</blockquote>
</blockquote>
<pre><span lang=3D"EN-US">wrote:<o:p></o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">I would also suggest that for video-mute that ins=
tead of stopping <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">video or sending black, the application should se=
nd a &quot;mute image&quot;.<o:p></o:p></span></pre>
</blockquote>
<pre><span lang=3D"EN-US">The question I have is whether there is: a) a fix=
ed image/video that<o:p></o:p></span></pre>
</blockquote>
</blockquote>
</blockquote>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">is serialized to RTP b) video &quot;comfort noise=
&quot;, or c) local mute <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">image/video playback.&nbsp; If it can't be c, the=
n you might need to <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">explain yourself.<o:p></o:p></span></pre>
</blockquote>
<pre><span lang=3D"EN-US">Either we build it into the system (which I disli=
ke - SetMuteImage() <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">or whathaveyou, though we could), or we put it in=
 the application <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">domain and suggest or assist implementation.<o:p>=
</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">The simple way to handle Mute is to revector the =
input to a &quot;mute&quot; or<o:p></o:p></span></pre>
</blockquote>
</blockquote>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">&quot;hold&quot; source, and they may not be the =
same (hold would have &quot;Please <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">Hold&quot; and music, perhaps; mute would have si=
lence and &quot;Muted&quot; or a <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">camera with a circle/slash, etc).&nbsp; You can a=
lso signal that out of <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">band to let the other end take smart action (remo=
ve muted video <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">image, display &quot;muted&quot; next to a name, =
etc).&nbsp; This can be done via <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">DataChannels, for example.&nbsp; You may then rei=
nvite for Hold to make it<o:p></o:p></span></pre>
</blockquote>
</blockquote>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<blockquote style=3D"margin-top:5.0pt;margin-bottom:5.0pt">
<pre><span lang=3D"EN-US">one-way, but that's mostly optional.<o:p></o:p></=
span></pre>
</blockquote>
<pre><span lang=3D"EN-US">The resulting requirements seem to me to be:<o:p>=
</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">1) To be able to use a still image as source for =
a video stream<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">2) To be able to change the source of a video str=
eam after its <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">creation<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">Both are requirements on the GetUserMedia API. Ca=
n we link these to a <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">specific use case, or do we need to modify the us=
e case document so <o:p></o:p></span></pre>
<pre><span lang=3D"EN-US">that the requirements are clear?<o:p></o:p></span=
></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<pre><span lang=3D"EN-US">_______________________________________________<o=
:p></o:p></span></pre>
<pre><span lang=3D"EN-US">rtcweb mailing list<o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.or=
g</a><o:p></o:p></span></pre>
<pre><span lang=3D"EN-US"><a href=3D"https://www.ietf.org/mailman/listinfo/=
rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span><=
/pre>
</blockquote>
<pre><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></pre>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

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On 6/17/2012 4:45 AM, Sunyang (Eric) wrote:
> Most pad and smarthphone, front and back camera has different quality.
> Not to metion the high resolution local video file (720p?)
> If we switch mediastream between them, and like what you said, you decoded-encoded-control-frame-by-frame,
> the system may be slow down, if you use codecs in browser, it will be more slower.
>
> The point is that we accept the renegoation rather then decode-encode operation to fit for previous SDP.
> Renegoation is not serious, we can accept it, but we can not accept the performance degradation.

I'm not certain what you're saying here. If the camera is encoded we
likely are having to decode and re-encode anyways (perhaps with a scale,
crop or composite operation as well, which require decoded video as well).

The point is that some operations (mute, hold, etc) *need* to be "on the
next frame" or close. This can be a temporary thing (mute video until
renegotiate as recv-only for hold), but it must not wait for a
renegotiation.

The system can and will scale the new video source down to the same
outgoing resolution already selected, and if it wants to it can scale it
up to the previous resolution if it's lower (that probably isn't a good
idea, but can be done).

-- 
Randell Jesup
randell-ietf@jesup.org


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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Harald Alvestrand" <harald@alvestrand.no>
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I still think that there is a significant difference between
files/images and live sources.  A live source provides a lot of
information about my environment - what's going on, whether I have
shaved, etc.  The file or image source that we use for hold is designed
to provide _no_ information about my environment.  In a user's mind,
showing "please wait" or a  images of a waterfall is very different from
showing anything _live_ from his environment, and we may want to reflect
that difference in the requirements.  For example, does the user have to
give explicit permission to show the "Please wait" source when the call
is put on hold, or is that configured as a default in the browser?  (The
latter, I'd  think, which makes it different from the front and rear
cameras.)

=20

-          Jim

P.S. I agree that the system can switch from front to rear cameras once
it has permission for both, though it should make it clear that it is
doing so.  =20

=20

From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Sunday, June 17, 2012 4:01 AM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)

=20

On 06/16/2012 11:58 PM, Jim Barnett wrote:=20

It's capturing the requirement accurately that I'm concerned with.  I
think "To be able to change the source of a video stream after its
creation" is too broad, in the sense that someone might interpret it to
mean that the system could shift from the front to the rear camera
without asking permission.

People certainly want to shift from the front to the rear camera, and
people want to have the ability to get all that asking-for-permissions
stuff out of the way before that.

I recommend the Media Task Force's "additional scenarios" for more
usages of "local media".
Jjim, I know you know where this is, but for others:
http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scenarios.ht
ml




  For mute and hold, we want something like
"switch to a dummy source and back".  The source that we switch to in
cases like this is not a 'first-class' video source, but rather a
place-holder.  That's very different from switching to another live
source.

ObLangDisagreement: Files, images and the output of video processors are
very real sources, and should not be "second class".

Switching between live streams (that we have permission to access) is
going to be required, but I think this belongs on the Media TF mailing
list.




=20
=20
- Jim
=20
-----Original Message-----
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Saturday, June 16, 2012 4:34 PM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments
ondraft-ietf-rtcweb-use-cases-and-requirements-07)
=20
On 06/16/2012 10:07 PM, Jim Barnett wrote:

	Harald,
	   Your description below makes me nervous:  with getUserMedia,
the=20
	user grants permission to _specific_ media streams (for example
audio,

=20

	or front camera, but not rear).

Yes, he grants permission to use of the camera. That permission persists
until <we have to define this point in time>. There's some interesting
language there now about "strong" and "weak" references to video
streams.
=20

	   If the system wants " to change the source of a video stream
after=20
	its creation", shouldn't it be required to ask permission again?
In=20
	this case,  common sense says that it's not necessary, but
that's=20
	because we're not changing to an arbitrary alternative the
source=20
	(e.g., suddenly switching on the rear camera) but rather
substituting=20
	some sort of 'null' or 'dummy' source.  Can we think of a
different=20
	way to phrase it?

There are multiple ways to implement a requirement, as always.
=20
One possibility is the simple one:
=20
getUserMedia(..., function(strream) {
       outgoingstream =3D stream;
})
getStreamFromPicture(image, function(stream) {
      =20
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)
=20
whatever the "replace" call does.
There may be a requirement that we keep a reference around that keeps
the video stream, so that we can restore it, in order that the camera
not close and can be used later:
=20
getUserMedia(..., function(strream) {
       copyOfCameraStream =3D stream;
       outgoingstream =3D MediaStream(stream.audioTracks,
stream.videoTracks);
=20
})
=20
getStreamFromPicture(image, function(stream) {
      =20
outgoingstream.videoStreams(0).replaceContent(stream.videoStreams(0));
}..)
=20
restorePicture =3D function() {
     =20
outgoingtstream.videoStreams(0).replaceContent(copyOfCameraStream.video.
Streams(0));
}
=20
But there are surely multiple other ways to define it. What I want to
make sure we have done first is to capture the requirement accurately.
=20

	=20
	- Jim
	=20
	-----Original Message-----
	From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
On=20
	Behalf Of Harald Alvestrand
	Sent: Saturday, June 16, 2012 3:56 PM
	To: rtcweb@ietf.org
	Subject: Re: [rtcweb] Mute implementations (Re: More Comments
	ondraft-ietf-rtcweb-use-cases-and-requirements-07)
	=20
	On 06/15/2012 11:18 PM, Randell Jesup wrote:

		On 6/15/2012 4:49 PM, Martin Thomson wrote:

			On 15 June 2012 13:42, Randell
Jesup<randell-ietf@jesup.org> <mailto:randell-ietf@jesup.org>=20

wrote:

				I would also suggest that for video-mute
that instead of stopping=20
				video or sending black, the application
should send a "mute image".

			The question I have is whether there is: a) a
fixed image/video that

=20

			is serialized to RTP b) video "comfort noise",
or c) local mute=20
			image/video playback.  If it can't be c, then
you might need to=20
			explain yourself.

		Either we build it into the system (which I dislike -
SetMuteImage()=20
		or whathaveyou, though we could), or we put it in the
application=20
		domain and suggest or assist implementation.
		=20
		The simple way to handle Mute is to revector the input
to a "mute" or

=20

		"hold" source, and they may not be the same (hold would
have "Please=20
		Hold" and music, perhaps; mute would have silence and
"Muted" or a=20
		camera with a circle/slash, etc).  You can also signal
that out of=20
		band to let the other end take smart action (remove
muted video=20
		image, display "muted" next to a name, etc).  This can
be done via=20
		DataChannels, for example.  You may then reinvite for
Hold to make it

=20

		one-way, but that's mostly optional.

	The resulting requirements seem to me to be:
	=20
	1) To be able to use a still image as source for a video stream
	2) To be able to change the source of a video stream after its=20
	creation
	=20
	Both are requirements on the GetUserMedia API. Can we link these
to a=20
	specific use case, or do we need to modify the use case document
so=20
	that the requirements are clear?
	=20
	               Harald
	=20
	=20
	=20
	=20
	_______________________________________________
	rtcweb mailing list
	rtcweb@ietf.org
	https://www.ietf.org/mailman/listinfo/rtcweb

=20

=20


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite =
lang=3DEN-US link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I still think that there is a significant difference between =
files/images and live sources.&nbsp; A live source provides a lot of =
information about my environment &#8211; what&#8217;s going on, whether =
I have shaved, etc.&nbsp; The file or image source that we use for hold =
is designed to provide _<i>no</i>_ information about my =
environment.&nbsp; In a user&#8217;s mind, showing &#8220;please =
wait&#8221; or a&nbsp; images of a waterfall is very different from =
showing anything _<i>live</i>_ from his environment, and we may want to =
reflect that difference in the requirements.&nbsp; For example, does the =
user have to give explicit permission to show the &#8220;Please =
wait&#8221; source when the call is put on hold, or is that configured =
as a default in the browser?&nbsp; (The latter, I&#8217;d&nbsp; think, =
which makes it different from the front and rear =
cameras.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoListParagraph =
style=3D'text-indent:-.25in;mso-list:l0 level1 lfo1'><![if =
!supportLists]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span style=3D'mso-list:Ignore'>-<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Jim<o:p></o:p></span></p><p class=3DMsoNormal =
style=3D'margin-left:.25in'><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>P.S. I agree that the system can switch from front to rear cameras =
once it has permission for both, though it should make it clear that it =
is doing so. &nbsp;&nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'> Harald Alvestrand [mailto:harald@alvestrand.no] <br><b>Sent:</b> =
Sunday, June 17, 2012 4:01 AM<br><b>To:</b> Jim Barnett<br><b>Cc:</b> =
rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] Mute implementations =
(Re: More Comments =
ondraft-ietf-rtcweb-use-cases-and-requirements-07)<o:p></o:p></span></p><=
/div></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal>On 06/16/2012 11:58 PM, Jim Barnett wrote: =
<o:p></o:p></p><pre>It's capturing the requirement accurately that I'm =
concerned with.&nbsp; I<o:p></o:p></pre><pre>think &quot;To be able to =
change the source of a video stream after =
its<o:p></o:p></pre><pre>creation&quot; is too broad, in the sense that =
someone might interpret it to<o:p></o:p></pre><pre>mean that the system =
could shift from the front to the rear =
camera<o:p></o:p></pre><pre>without asking =
permission.<o:p></o:p></pre><p class=3DMsoNormal>People certainly want =
to shift from the front to the rear camera, and people want to have the =
ability to get all that asking-for-permissions stuff out of the way =
before that.<br><br>I recommend the Media Task Force's &quot;additional =
scenarios&quot; for more usages of &quot;local media&quot;.<br>Jjim, I =
know you know where this is, but for others:<br><a =
href=3D"http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/scena=
rios.html">http://dvcs.w3.org/hg/dap/raw-file/tip/media-stream-capture/sc=
enarios.html</a><br><br><br><o:p></o:p></p><pre>&nbsp; For mute and =
hold, we want something like<o:p></o:p></pre><pre>&quot;switch to a =
dummy source and back&quot;.&nbsp; The source that we switch to =
in<o:p></o:p></pre><pre>cases like this is not a 'first-class' video =
source, but rather a<o:p></o:p></pre><pre>place-holder.&nbsp; That's =
very different from switching to another =
live<o:p></o:p></pre><pre>source.<o:p></o:p></pre><p =
class=3DMsoNormal>ObLangDisagreement: Files, images and the output of =
video processors are very real sources, and should not be &quot;second =
class&quot;.<br><br>Switching between live streams (that we have =
permission to access) is going to be required, but I think this belongs =
on the Media TF mailing =
list.<br><br><br><o:p></o:p></p><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nb=
sp;</o:p></pre><pre>- =
Jim<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>-----Original =
Message-----<o:p></o:p></pre><pre>From: Harald Alvestrand [<a =
href=3D"mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] =
<o:p></o:p></pre><pre>Sent: Saturday, June 16, 2012 4:34 =
PM<o:p></o:p></pre><pre>To: Jim Barnett<o:p></o:p></pre><pre>Cc: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pre>=
Subject: Re: [rtcweb] Mute implementations (Re: More =
Comments<o:p></o:p></pre><pre>ondraft-ietf-rtcweb-use-cases-and-requireme=
nts-07)<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>On 06/16/2012 =
10:07 PM, Jim Barnett wrote:<o:p></o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>Harald,<o:p></o:p></p=
re><pre>&nbsp;&nbsp; Your description below makes me nervous:&nbsp; with =
getUserMedia, the <o:p></o:p></pre><pre>user grants permission to =
_specific_ media streams (for example =
audio,<o:p></o:p></pre></blockquote><pre><o:p>&nbsp;</o:p></pre><blockquo=
te style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>or front camera, =
but not rear).<o:p></o:p></pre></blockquote><pre>Yes, he grants =
permission to use of the camera. That permission =
persists<o:p></o:p></pre><pre>until &lt;we have to define this point in =
time&gt;. There's some interesting<o:p></o:p></pre><pre>language there =
now about &quot;strong&quot; and &quot;weak&quot; references to =
video<o:p></o:p></pre><pre>streams.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p=
></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>&nbsp;&nbsp; If the =
system wants &quot; to change the source of a video stream after =
<o:p></o:p></pre><pre>its creation&quot;, shouldn't it be required to =
ask permission again?&nbsp; In <o:p></o:p></pre><pre>this case,&nbsp; =
common sense says that it's not necessary, but that's =
<o:p></o:p></pre><pre>because we're not changing to an arbitrary =
alternative the source <o:p></o:p></pre><pre>(e.g., suddenly switching =
on the rear camera) but rather substituting <o:p></o:p></pre><pre>some =
sort of 'null' or 'dummy' source.&nbsp; Can we think of a different =
<o:p></o:p></pre><pre>way to phrase =
it?<o:p></o:p></pre></blockquote><pre>There are multiple ways to =
implement a requirement, as =
always.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>One possibility =
is the simple =
one:<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>getUserMedia(..., =
function(strream) =
{<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
outgoingstream =3D =
stream;<o:p></o:p></pre><pre>})<o:p></o:p></pre><pre>getStreamFromPicture=
(image, function(stream) =
{<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
<o:p></o:p></pre><pre>outgoingstream.videoStreams(0).replaceContent(strea=
m.videoStreams(0));<o:p></o:p></pre><pre>}..)<o:p></o:p></pre><pre><o:p>&=
nbsp;</o:p></pre><pre>whatever the &quot;replace&quot; call =
does.<o:p></o:p></pre><pre>There may be a requirement that we keep a =
reference around that keeps<o:p></o:p></pre><pre>the video stream, so =
that we can restore it, in order that the =
camera<o:p></o:p></pre><pre>not close and can be used =
later:<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>getUserMedia(...,=
 function(strream) =
{<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
copyOfCameraStream =3D =
stream;<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
outgoingstream =3D =
MediaStream(stream.audioTracks,<o:p></o:p></pre><pre>stream.videoTracks);=
<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>})<o:p></o:p></pre><pre=
><o:p>&nbsp;</o:p></pre><pre>getStreamFromPicture(image, =
function(stream) =
{<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
<o:p></o:p></pre><pre>outgoingstream.videoStreams(0).replaceContent(strea=
m.videoStreams(0));<o:p></o:p></pre><pre>}..)<o:p></o:p></pre><pre><o:p>&=
nbsp;</o:p></pre><pre>restorePicture =3D function() =
{<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
<o:p></o:p></pre><pre>outgoingtstream.videoStreams(0).replaceContent(copy=
OfCameraStream.video.<o:p></o:p></pre><pre>Streams(0));<o:p></o:p></pre><=
pre>}<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>But there are =
surely multiple other ways to define it. What I want =
to<o:p></o:p></pre><pre>make sure we have done first is to capture the =
requirement =
accurately.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre><o:p>&nbsp;</o:p></pr=
e><pre>- =
Jim<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>-----Original =
Message-----<o:p></o:p></pre><pre>From: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a =
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] On <o:p></o:p></pre><pre>Behalf Of Harald =
Alvestrand<o:p></o:p></pre><pre>Sent: Saturday, June 16, 2012 3:56 =
PM<o:p></o:p></pre><pre>To: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pre>=
Subject: Re: [rtcweb] Mute implementations (Re: More =
Comments<o:p></o:p></pre><pre>ondraft-ietf-rtcweb-use-cases-and-requireme=
nts-07)<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>On 06/15/2012 =
11:18 PM, Randell Jesup wrote:<o:p></o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>On 6/15/2012 4:49 =
PM, Martin Thomson wrote:<o:p></o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>On 15 June 2012 =
13:42, Randell Jesup<a =
href=3D"mailto:randell-ietf@jesup.org">&lt;randell-ietf@jesup.org&gt;</a>=
<o:p></o:p></pre></blockquote></blockquote></blockquote><pre>wrote:<o:p><=
/o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>I would also suggest =
that for video-mute that instead of stopping <o:p></o:p></pre><pre>video =
or sending black, the application should send a &quot;mute =
image&quot;.<o:p></o:p></pre></blockquote><pre>The question I have is =
whether there is: a) a fixed image/video =
that<o:p></o:p></pre></blockquote></blockquote></blockquote><pre><o:p>&nb=
sp;</o:p></pre><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>is serialized to RTP =
b) video &quot;comfort noise&quot;, or c) local mute =
<o:p></o:p></pre><pre>image/video playback.&nbsp; If it can't be c, then =
you might need to <o:p></o:p></pre><pre>explain =
yourself.<o:p></o:p></pre></blockquote><pre>Either we build it into the =
system (which I dislike - SetMuteImage() <o:p></o:p></pre><pre>or =
whathaveyou, though we could), or we put it in the application =
<o:p></o:p></pre><pre>domain and suggest or assist =
implementation.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>The =
simple way to handle Mute is to revector the input to a &quot;mute&quot; =
or<o:p></o:p></pre></blockquote></blockquote><pre><o:p>&nbsp;</o:p></pre>=
<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>&quot;hold&quot; =
source, and they may not be the same (hold would have &quot;Please =
<o:p></o:p></pre><pre>Hold&quot; and music, perhaps; mute would have =
silence and &quot;Muted&quot; or a <o:p></o:p></pre><pre>camera with a =
circle/slash, etc).&nbsp; You can also signal that out of =
<o:p></o:p></pre><pre>band to let the other end take smart action =
(remove muted video <o:p></o:p></pre><pre>image, display =
&quot;muted&quot; next to a name, etc).&nbsp; This can be done via =
<o:p></o:p></pre><pre>DataChannels, for example.&nbsp; You may then =
reinvite for Hold to make =
it<o:p></o:p></pre></blockquote></blockquote><pre><o:p>&nbsp;</o:p></pre>=
<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><pre>one-way, but that's =
mostly optional.<o:p></o:p></pre></blockquote><pre>The resulting =
requirements seem to me to =
be:<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>1) To be able to =
use a still image as source for a video stream<o:p></o:p></pre><pre>2) =
To be able to change the source of a video stream after its =
<o:p></o:p></pre><pre>creation<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pr=
e><pre>Both are requirements on the GetUserMedia API. Can we link these =
to a <o:p></o:p></pre><pre>specific use case, or do we need to modify =
the use case document so <o:p></o:p></pre><pre>that the requirements are =
clear?<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Harald<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p>=
</pre><pre><o:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>______=
_________________________________________<o:p></o:p></pre><pre>rtcweb =
mailing list<o:p></o:p></pre><pre><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></pre><pre>=
<a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></pre></blockquote><pre><o:p>&nbs=
p;</o:p></pre><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></html>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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For the fun of it I thought a little while about what options that are 
currently available with the APIs that are in the W3C docs as is. The 
list is in no way exhaustive, there are other ways to do it.

Perhaps this is not sufficient for what people want to do, but at least 
there are some things you can do already:

1. Replace video with the display of an image
=============================================
One solution: Use the poster attribute of the video element. Quote 
from the html spec: The poster attribute gives the address of an image 
file that the user agent can show while no video data is available.. We 
can spec it such that if no video track of a MediaStreamis active, that 
counts as no data available. This way, as soon as the user mutes the 
video, an image will replace it at display. The site can supply the 
image (either a default by the site or one profile pic selected by the 
user), or it can be provided over the data channel (as a blob).

Another solution: A signal is sent on the data channel (possibly 
accompanied by the image) on the data channel, the peers app overlays 
the image on top of the video element as result (and switches back on 
another signal).

A third (similar) solution: The app at the receiving end listens to 
onunmute of the video track in question; once it fires an image is 
overlayed.

A fourth: The peers app analyzes the video using canvas, and overlays 
an image when there is no data in the video.

2. Switch between front and rear camera
========================================
One solution: create a MediaStream (e.g. by calling getUserMedia twice 
immediately after each other, or by combining existing tracks into a new 
MediaStream) containing two video tracks (front+back cam). The video 
shown is toggled by enabling/disabling video tracks (the video element 
plays the first active one in the case of several video tracks).

Another: use add/removeTrack to switch video track being sent.

3. Play music on hold
=====================
One solution (audio only case): The site provides an URL to a (music) 
file to play when on hold. The peers app registers an event listener 
that listens to onmute on the incoming audio track. When onmute 
fires, the app changes the source of the audio element to the URL of the 
music file.

One solution if the communication is audio+video, and the intent is to 
just replace the audio (but continue playing video): assume a 
MediaStream with one audio and one video track is used. Play the audio 
in an audio element, the video in a (muted) video element. At on hold, 
replace the source of the audio element with the music file.

A solution if the communication is audio+video, and the intent is to 
replace audio and video: either mute the tracks (at sending side); 
receiving end gets onmute event and replaces the .src of the video 
element with the URL to a video file, or signal on the data channel 
something that makes the other end replace the src of the video element 
with an URL to a file

Another solution is that the sending end does removeStream from 
PeerConnection. This fires onremovestream at the receiving side; that 
event can trigger the change of src to the video element. This solution 
frees up resources.

Note that once the recording function has been specced (in such a way 
that the recorded file can be the source of a MediaStream) more options 
will become available.

--Stefan

On 06/17/2012 02:38 AM, Randell Jesup wrote:
> On 6/16/2012 5:58 PM, Jim Barnett wrote:
>> It's capturing the requirement accurately that I'm concerned with.  I
>> think "To be able to change the source of a video stream after its
>> creation" is too broad, in the sense that someone might interpret it to
>> mean that the system could shift from the front to the rear camera
>> without asking permission.  For mute and hold, we want something like
>> "switch to a dummy source and back".  The source that we switch to in
>> cases like this is not a 'first-class' video source, but rather a
>> place-holder.  That's very different from switching to another live
>> source.
>
> Which, by the way, we should be able to do.
>
> MediaStreams need to be able to be repointed, switched, processed, etc,
> OR you need to be able to replace a MediaStream in the inputs to
> PeerConnection with another one transparently without renegotiation (and
> that misses a bunch of uses).
>
> The MediaStream Processing provides a mechanism to do this for both
> audio and video, plus the ability to do more complex operations
> (cross-fades audio-and-video, frame-accurate source switches, etc).
>
> If you don't have or use that, you need the ability to replace a track
> in a MediaStream with another track.
>
> All we're talking about are mechanisms to shuffle existing or creatable
> tracks/streams; a new camera source would require a call to
> getUserMedia() and a dialog.  Once you had front and back, you could
> switch sources without effectively shutting down all media and restarting.
>
> Simple one camera ->  one video stream mechanisms are insufficient for
> application using these features.  If you force people, they'll roll
> their own using canvases, etc.
>
> I want something generic and usable in lots of interesting ways, not a
> few special-purpose pre-coded mechanisms (for example, a
> track.setMuteImage() method).
>


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To: Jim Barnett <Jim.Barnett@genesyslab.com>
References: <5A2F246E-9D77-4B72-B3F1-681B00FA99FD@cisco.com><F3D4ABD6AB47084B84337CF4F3446A464B2F2631AC@ESESSCMS0362.eemea.ericsson.se><9F33F40F6F2CD847824537F3C4E37DDF021B53@MCHP04MSX.global-ad.net><CA+9kkMCL1k3x2hYi632nDU2g6CtOq3O7risDi38+7FTeWkxcGA@mail.gmail.com><9F33F40F6F2CD847824537F3C4E37DDF0223ED@MCHP04MSX.global-ad.net><4FD89773.3090506@ericsson.com><9F33F40F6F2CD847824537F3C4E37DDF02594D@MCHP04MSX.global-ad.net><4FD9AD1E.6060504@alvestrand.no> <4FD9E2BE.2080907@ericsson.com><CAOJ7v-18EynuscX3tLRryzqKxyXwYzES9K4zroE-5=FL4o8WzA@mail.gmail.com><4FDB9E20.8070408@jesup.org><CABkgnnVqeyuayFsiA7YD+OFnyCi8tqode6VdypeaHKT+Z73KWw@mail.gmail.com><4FDBA6AC.2080107@jesup.org> <4FDCE4D4.7070909@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8106581BE5@NAHALD.us.int.genesyslab.com> <4FDCEDC2.2040903@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8106581BEA@NAHALD.us.int.genesyslab.com> <4FDD8EBA.2040601@alvestrand.no> <E17CAD772E76C742B645BD4DC602CD8106581BF2@NAHALD.us.int.genesysl ab.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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On 06/17/2012 04:17 PM, Jim Barnett wrote:
>
> I still think that there is a significant difference between 
> files/images and live sources.  A live source provides a lot of 
> information about my environment -- what's going on, whether I have 
> shaved, etc.  The file or image source that we use for hold is 
> designed to provide _/no/_ information about my environment.  In a 
> user's mind, showing "please wait" or a  images of a waterfall is very 
> different from showing anything _/live/_ from his environment, and we 
> may want to reflect that difference in the requirements.  For example, 
> does the user have to give explicit permission to show the "Please 
> wait" source when the call is put on hold, or is that configured as a 
> default in the browser?  (The latter, I'd  think, which makes it 
> different from the front and rear cameras.)
>
>
I agree with respect to the needed permissions; this is mostly a 
language issue.

I think we will have some applications (like the shared YouTube viewer 
in hangouts) where a stored medium is the most important piece of the 
app - but I don't think that means we need special access procedures for 
it (the "file upload" dialog for getting permissions to access a file 
should be sufficient), and for resources that are already accessible to 
the Web app, like images loaded to the page - I see absolutely no reason 
to put extra requirements on it.

But as I said - Media task force.




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    On 06/17/2012 04:17 PM, Jim Barnett wrote:
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        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">I
            still think that there is a significant difference between
            files/images and live sources.&nbsp; A live source provides a lot
            of information about my environment &#8211; what&#8217;s going on,
            whether I have shaved, etc.&nbsp; The file or image source that
            we use for hold is designed to provide _<i>no</i>_
            information about my environment.&nbsp; In a user&#8217;s mind, showing
            &#8220;please wait&#8221; or a&nbsp; images of a waterfall is very different
            from showing anything _<i>live</i>_ from his environment,
            and we may want to reflect that difference in the
            requirements.&nbsp; For example, does the user have to give
            explicit permission to show the &#8220;Please wait&#8221; source when
            the call is put on hold, or is that configured as a default
            in the browser?&nbsp; (The latter, I&#8217;d&nbsp; think, which makes it
            different from the front and rear cameras.)<o:p></o:p></span></p>
        <br>
      </div>
    </blockquote>
    I agree with respect to the needed permissions; this is mostly a
    language issue.<br>
    <br>
    I think we will have some applications (like the shared YouTube
    viewer in hangouts) where a stored medium is the most important
    piece of the app - but I don't think that means we need special
    access procedures for it (the "file upload" dialog for getting
    permissions to access a file should be sufficient), and for
    resources that are already accessible to the Web app, like images
    loaded to the page - I see absolutely no reason to put extra
    requirements on it.<br>
    <br>
    But as I said - Media task force.<br>
    <br>
    <br>
    <br>
  </body>
</html>

--------------040809090306040100070909--

From harald@alvestrand.no  Sun Jun 17 08:57:47 2012
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Subject: [rtcweb] Review, RTP recommendations - draft-ietf-rtcweb-rtp-usage-03
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At the interim, I promised to make more precise my objections to some of 
the language of this I-D. I'm taking the opportunity to read through the 
whole document and give commentary.
I hope this is helpful.

- Section 2, rationale: Should add that an important driver for the 
profile is to rule things OUT of scope - that features not mentioned 
here are not required for successful RTCWEB interoperation.

- Section 3, terminology: This should reference the RTP document and the 
rtcweb-overview document. There are lots of terms that come from there.

In particular, the term "media stream" needs to be called out, since a 
WebRTC "MediaStream" is not the same as an RTP "media stream", and both 
are used in this document. I suggest consistently using the terms 
"WebRTC MediaStream" (prefix, no space) and "RTP media stream" (prefix, 
with space) for the two concepts, and explicitly never using the term 
"media stream" without a prefix.

- Section 4.7 Symmetric RTP/RTCP: Between -02 and -03, I see that 
support for symmetric RTP/RTCP has been changed from

    Using Symmetric RTP and RTCP [RFC4961] is REQUIRED.

to

    implementations are
    REQUIRED to implement Symmetric RTP [RFC4961]

Reading RFC 4961, I see that it states:

    A device supports symmetric RTP if it selects, communicates, and uses
    IP addresses and port numbers such that, when receiving a
    bidirectional RTP media stream on UDP port "A" and IP address "a", it
    also transmits RTP media for that stream from the same source UDP
    port "A" and IP address "a".

So it's impossible to support RFC 4961 without using it, and the 
formulations are equivalent. But that's such a novel concept among the 
usual universe of RTP extensions that it's worth calling out explicitly.

It might even be beneficial to say that it's OK to not interoperate with 
devices that don't do RFC 4961.

- Section 4.8 RTCP CNAME generation - as discussed at the interim, it 
might be good to state that support for receiving RFC 6222-style CNAMEs, 
as well as RFC 3550-style names, is required.

- Section 5.1 states that RTP Transport Translators "are expected to be 
supported". RTP Transport Translators require that call setup be 
coordinated across all endpoints in the session to avoid clashing 
payload types, as discussed in the multiplex draft. I'm OK with a 
formulation that says that "this RTP profile should be able to support 
all these topologies, if supported by appropriate signallling", but I'm 
not OK with a formulation that absolutely requires signalling support 
for RTP Transport Translators; I'm not at all confident that we can 
successfully negotiate RTP Transport Translator-based applications for 
all cases.

- Section 5.1.5 TMMBR - should we say explicitly that the sender is 
expected to honor the constraint?

- Section 7.4 Legacy Interop Limitations (for congestion) seems like a 
work in progress. We'll discuss this more in the congestion context, but 
the "it's been suggested ... send 64 Kbits/sec" at the bottom should 
definitely not be stated like this in the final document.

- Section 11.1 "However, one SSRC  will identify a media stream and its 
timing. " - I think in this case, "WebRTC MediaStream" is the intended 
usage.

- Section 12.1 and 12.2 - in this section, I think all the uses of 
"media stream" refer to "RTP media stream".

I can't parse the grammar of "In addition the above discussed
    criteria to support multiple media types in one single RTP session
    results that also an end-point that has one audio and one video
    source still need two transmit using two SSRCs concurrently" - I 
think it's trying to say that when audio and video go into the same RTP 
session, they use different SSRCs.

- Section 12.3 is not very clear to me. I'm not sure I would agree with 
it if it was clear; the "relay" described here is probably an RTP 
Transport Translator, but if it is, it should be called that.

This is a quick pass, where I have focused on sections 1-7. There are 
many formulation issues, and some of them may hide technical 
disagreements. But on the whole, I think this document is very close to 
saying most of the things that need to be said.

                          Harald



--------------040607040006060807040501
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    At the interim, I promised to make more precise my objections to
    some of the language of this I-D. I'm taking the opportunity to read
    through the whole document and give commentary.<br>
    I hope this is helpful.<br>
    <br>
    - Section 2, rationale: Should add that an important driver for the
    profile is to rule things OUT of scope - that features not mentioned
    here are not required for successful RTCWEB interoperation.<br>
    <br>
    - Section 3, terminology: This should reference the RTP document and
    the rtcweb-overview document. There are lots of terms that come from
    there.<br>
    <br>
    In particular, the term "media stream" needs to be called out, since
    a WebRTC "MediaStream" is not the same as an RTP "media stream", and
    both are used in this document. I suggest consistently using the
    terms "WebRTC MediaStream" (prefix, no space) and "RTP media stream"
    (prefix, with space) for the two concepts, and explicitly never
    using the term "media stream" without a prefix.<br>
    <br>
    - Section 4.7 Symmetric RTP/RTCP: Between -02 and -03, I see that
    support for symmetric RTP/RTCP has been changed from<br>
    <br>
    &nbsp;&nbsp; Using Symmetric RTP and RTCP [RFC4961] is REQUIRED.<br>
    <br>
    to<br>
    <br>
    &nbsp;&nbsp; implementations are<br>
    &nbsp;&nbsp; REQUIRED to implement Symmetric RTP [RFC4961]<br>
    <br>
    Reading RFC 4961, I see that it states:<br>
    <br>
    &nbsp;&nbsp; A device supports symmetric RTP if it selects, communicates, and
    uses<br>
    &nbsp;&nbsp; IP addresses and port numbers such that, when receiving a<br>
    &nbsp;&nbsp; bidirectional RTP media stream on UDP port "A" and IP address
    "a", it<br>
    &nbsp;&nbsp; also transmits RTP media for that stream from the same source UDP<br>
    &nbsp;&nbsp; port "A" and IP address "a".<br>
    <br>
    So it's impossible to support RFC 4961 without using it, and the
    formulations are equivalent. But that's such a novel concept among
    the usual universe of RTP extensions that it's worth calling out
    explicitly.<br>
    <br>
    It might even be beneficial to say that it's OK to not interoperate
    with devices that don't do RFC 4961.<br>
    <br>
    - Section 4.8 RTCP CNAME generation - as discussed at the interim,
    it might be good to state that support for receiving RFC 6222-style
    CNAMEs, as well as RFC 3550-style names, is required.<br>
    <br>
    - Section 5.1 states that RTP Transport Translators "are expected to
    be supported". RTP Transport Translators require that call setup be
    coordinated across all endpoints in the session to avoid clashing
    payload types, as discussed in the multiplex draft. I'm OK with a
    formulation that says that "this RTP profile should be able to
    support all these topologies, if supported by appropriate
    signallling", but I'm not OK with a formulation that absolutely
    requires signalling support for RTP Transport Translators; I'm not
    at all confident that we can successfully negotiate RTP Transport
    Translator-based applications for all cases.<br>
    <br>
    - Section 5.1.5 TMMBR - should we say explicitly that the sender is
    expected to honor the constraint?<br>
    <br>
    - Section 7.4 Legacy Interop Limitations (for congestion) seems like
    a work in progress. We'll discuss this more in the congestion
    context, but the "it's been suggested ... send 64 Kbits/sec" at the
    bottom should definitely not be stated like this in the final
    document.<br>
    <br>
    - Section 11.1 "However, one SSRC&nbsp; will identify a media stream and
    its timing.
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    " - I think in this case, "WebRTC MediaStream" is the intended
    usage.<br>
    <br>
    - Section 12.1 and 12.2 - in this section, I think all the uses of
    "media stream" refer to "RTP media stream".<br>
    <br>
    I can't parse the grammar of "In addition the above discussed<br>
    &nbsp;&nbsp; criteria to support multiple media types in one single RTP
    session<br>
    &nbsp;&nbsp; results that also an end-point that has one audio and one video<br>
    &nbsp;&nbsp; source still need two transmit using two SSRCs concurrently" - I
    think it's trying to say that when audio and video go into the same
    RTP session, they use different SSRCs. <br>
    <br>
    - Section 12.3 is not very clear to me. I'm not sure I would agree
    with it if it was clear; the "relay" described here is probably an
    RTP Transport Translator, but if it is, it should be called that.<br>
    <br>
    This is a quick pass, where I have focused on sections 1-7. There
    are many formulation issues, and some of them may hide technical
    disagreements. But on the whole, I think this document is very close
    to saying most of the things that need to be said.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    <br>
  </body>
</html>

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From randell-ietf@jesup.org  Sun Jun 17 09:02:49 2012
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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CCing Media-capture as much of this touches on issues there.  For 
context, you can read the recent rtcweb archives.  The general topic is 
"Mute" (and Hold) and both replacing video with an image or pre-recorded 
video clip (repeating if short), and related issue of replacing a 
front-camera video with a rear-camera video without operations that take 
significant time (renegotiation of the peerconnection).  Basically 
immediate source switches.

On 6/17/2012 10:17 AM, Jim Barnett wrote:
> I still think that there is a significant difference between
> files/images and live sources.  A live source provides a lot of
> information about my environment  whats going on, whether I have
> shaved, etc.  The file or image source that we use for hold is designed
> to provide _/no/_ information about my environment.  In a users mind,
> showing please wait or a  images of a waterfall is very different from
> showing anything _/live/_ from his environment, and we may want to
> reflect that difference in the requirements.  For example, does the user
> have to give explicit permission to show the Please wait source when
> the call is put on hold, or is that configured as a default in the
> browser?  (The latter, Id  think, which makes it different from the
> front and rear cameras.)

Selecting an image or video from the user's filesystem would require the 
user's direct permission (ala <input type="file">).  Sourcing a 
mediastream from a web source (i.e. part of the app) or from a 
app-generated canvas (with "please wait" in it) would not.

If the app requests both front and rear camera streams and gets them 
(two getUserMedia() calls), then permission is already given and the app 
should be able to do anything with those streams.

If the app hasn't requested the rear camera yet, requesting would have 
to be done first and would engender a request to the user to permit access.

Once the app has both streams, it should be able to switch between the 
streams on demand (front/back UI button in the app), with no delay, and 
certainly no renegotiation of the call.

If there's a reason it could renegotiate after switching.  For (a 
contrived) example, if it had negotiated a call with a low maximum 
resolution, it could switch (and downscale the new hires stream) until a 
renegotiation allowed sending the full resolution.  That's a contrived 
example, but there might be some real-world examples, and that's up to 
the app.

The same things apply to switching between video (live or recorded) and 
still images.

These speak to how MediaStream work and can be processed, and how a 
track in a MediaStream can be used to represent different things, and 
how that's different than a MediaStream with two tracks.  Some of this 
impacts the interface and assumptions on MediaStreams and tracks in 
PeerConnection.  These are issues because PeerConnection, <video> (and 
others) may play a particular track of a MediaStream.

I'll further respond to Stefan's detailed email.

-- 
Randell Jesup
randell-ietf@jesup.org


From harald@alvestrand.no  Sun Jun 17 09:18:01 2012
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On 06/17/2012 08:48 AM, Sunyang (Eric) wrote:
> You said:  "MediaStreams need to be able to be repointed, switched, processed, etc, OR you need to be able to replace a MediaStream in the inputs to PeerConnection with another one transparently without renegotiation (and that misses a bunch of uses)."
>
> But if the MediaStream  you choose to replace the existing one contain codecs that not supported by remote end, I think we still need renegotiation.
Alert.... a MediaStream doesn't contain a codec.

http://wiki.alvestrand.no/w/index.php/The_Byte_Stream_Fallacy

A MediaStream may be linked to a source that provides data in a given
codec format; it may be linked to a sink that accepts data in the given
codec format, and as a result the work that goes on to support the
moving of data from the source to the sink may be more or less complex.

But saying that a MediaStream "contains a codec" is (to my mind) not a
good idea.

> Or you can make sure that the browser can choose the same codecs as default with the replaced one, I think browser can not do this.
>
> -----ÓÊŒþÔ­Œþ-----
> ·¢ŒþÈË: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] Žú±í Randell Jesup
> ·¢ËÍÊ±Œä: 2012Äê6ÔÂ17ÈÕ 8:39
> ÊÕŒþÈË: rtcweb@ietf.org
> Ö÷Ìâ: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
>
> On 6/16/2012 5:58 PM, Jim Barnett wrote:
>> It's capturing the requirement accurately that I'm concerned with.  I
>> think "To be able to change the source of a video stream after its
>> creation" is too broad, in the sense that someone might interpret it to
>> mean that the system could shift from the front to the rear camera
>> without asking permission.  For mute and hold, we want something like
>> "switch to a dummy source and back".  The source that we switch to in
>> cases like this is not a 'first-class' video source, but rather a
>> place-holder.  That's very different from switching to another live
>> source.
> Which, by the way, we should be able to do.
>
> MediaStreams need to be able to be repointed, switched, processed, etc, 
> OR you need to be able to replace a MediaStream in the inputs to 
> PeerConnection with another one transparently without renegotiation (and 
> that misses a bunch of uses).
>
> The MediaStream Processing provides a mechanism to do this for both 
> audio and video, plus the ability to do more complex operations 
> (cross-fades audio-and-video, frame-accurate source switches, etc).
>
> If you don't have or use that, you need the ability to replace a track 
> in a MediaStream with another track.
>
> All we're talking about are mechanisms to shuffle existing or creatable 
> tracks/streams; a new camera source would require a call to 
> getUserMedia() and a dialog.  Once you had front and back, you could 
> switch sources without effectively shutting down all media and restarting.
>
> Simple one camera -> one video stream mechanisms are insufficient for 
> application using these features.  If you force people, they'll roll 
> their own using canvases, etc.
>
> I want something generic and usable in lots of interesting ways, not a 
> few special-purpose pre-coded mechanisms (for example, a 
> track.setMuteImage() method).
>


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    On 06/17/2012 08:48 AM, Sunyang (Eric) wrote:
    <blockquote
cite="mid:9254B5E6361B1648AFC00BA447E6E8C32AEA910F@szxeml545-mbx.china.huawei.com"
      type="cite">
      <pre wrap="">
You said:  "MediaStreams need to be able to be repointed, switched, processed, etc, OR you need to be able to replace a MediaStream in the inputs to PeerConnection with another one transparently without renegotiation (and that misses a bunch of uses)."

But if the MediaStream  you choose to replace the existing one contain codecs that not supported by remote end, I think we still need renegotiation.</pre>
    </blockquote>
    Alert.... a MediaStream doesn't contain a codec.<br>
    <br>
    <meta http-equiv="content-type" content="text/html; charset=GB2312">
    <a
      href="http://wiki.alvestrand.no/w/index.php/The_Byte_Stream_Fallacy">http://wiki.alvestrand.no/w/index.php/The_Byte_Stream_Fallacy</a><br>
    <br>
    A MediaStream may be linked to a source that provides data in a
    given codec format; it may be linked to a sink that accepts data in
    the given codec format, and as a result the work that goes on to
    support the moving of data from the source to the sink may be more
    or less complex.<br>
    <br>
    But saying that a MediaStream "contains a codec" is (to my mind) not
    a good idea.<br>
    <br>
    <blockquote
cite="mid:9254B5E6361B1648AFC00BA447E6E8C32AEA910F@szxeml545-mbx.china.huawei.com"
      type="cite">
      <pre wrap="">
Or you can make sure that the browser can choose the same codecs as default with the replaced one, I think browser can not do this.

-----ÓÊŒþÔ­Œþ-----
·¢ŒþÈË: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] Žú±í Randell Jesup
·¢ËÍÊ±Œä: 2012Äê6ÔÂ17ÈÕ 8:39
ÊÕŒþÈË: <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
Ö÷Ìâ: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)

On 6/16/2012 5:58 PM, Jim Barnett wrote:
</pre>
      <blockquote type="cite">
        <pre wrap="">It's capturing the requirement accurately that I'm concerned with.  I
think "To be able to change the source of a video stream after its
creation" is too broad, in the sense that someone might interpret it to
mean that the system could shift from the front to the rear camera
without asking permission.  For mute and hold, we want something like
"switch to a dummy source and back".  The source that we switch to in
cases like this is not a 'first-class' video source, but rather a
place-holder.  That's very different from switching to another live
source.
</pre>
      </blockquote>
      <pre wrap="">
Which, by the way, we should be able to do.

MediaStreams need to be able to be repointed, switched, processed, etc, 
OR you need to be able to replace a MediaStream in the inputs to 
PeerConnection with another one transparently without renegotiation (and 
that misses a bunch of uses).

The MediaStream Processing provides a mechanism to do this for both 
audio and video, plus the ability to do more complex operations 
(cross-fades audio-and-video, frame-accurate source switches, etc).

If you don't have or use that, you need the ability to replace a track 
in a MediaStream with another track.

All we're talking about are mechanisms to shuffle existing or creatable 
tracks/streams; a new camera source would require a call to 
getUserMedia() and a dialog.  Once you had front and back, you could 
switch sources without effectively shutting down all media and restarting.

Simple one camera -&gt; one video stream mechanisms are insufficient for 
application using these features.  If you force people, they'll roll 
their own using canvases, etc.

I want something generic and usable in lots of interesting ways, not a 
few special-purpose pre-coded mechanisms (for example, a 
track.setMuteImage() method).

</pre>
    </blockquote>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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CC'd to both lists, as this affects both Media Capture (MediaStreams) 
and rtcweb protocol issues (and PeerConnection API issues, but I'm not 
yet including the WebRTC list).

On 6/17/2012 10:33 AM, Stefan Hakansson LK wrote:
> For the fun of it I thought a little while about what options that are
> currently available with the APIs that are in the W3C docs as is. The
> list is in no way exhaustive, there are other ways to do it.
>
> Perhaps this is not sufficient for what people want to do, but at least
> there are some things you can do already:
>
> 1. Replace video with the display of an image
> =============================================
> One solution: Use the poster attribute of the video element. Quote
> from the html spec: The poster attribute gives the address of an image
> file that the user agent can show while no video data is available.. We
> can spec it such that if no video track of a MediaStreamis active, that
> counts as no data available. This way, as soon as the user mutes the
> video, an image will replace it at display. The site can supply the
> image (either a default by the site or one profile pic selected by the
> user), or it can be provided over the data channel (as a blob).

This is a receiver-side action and thus is not very useful for Mute and 
Hold operations.  This would depend on those stopping transmitting 
(which could cause other issues) and would infer that lack of reception 
means mute or hold - while it could also mean a temporary or permanent 
loss of connectivity, etc.

> Another solution: A signal is sent on the data channel (possibly
> accompanied by the image) on the data channel, the peers app overlays
> the image on top of the video element as result (and switches back on
> another signal).

I've done this (using RTCP APP as a data channel), and it works well 
*if* both sides are the same application, and it still requires that the 
sources immediately mute (though black would be ok in this case). 
However, if you ever need to talk to something that doesn't understand 
this out-of-band "I'm muted" signal, then you really want to Mute/Hold 
with an image (and sound for Hold).  (And I've had to do this as well.)

> A third (similar) solution: The app at the receiving end listens to
> onunmute of the video track in question; once it fires an image is
> overlayed.

How does "onmute" work?  This similarly presupposes agreement on the 
out-of-band message, and also has the same issue when in a heterogeneous 
environment.

> A fourth: The peers app analyzes the video using canvas, and overlays
> an image when there is no data in the video.

a) ugh, what a waste of CPU cycles
b) Has the same problem as the first - the receiver doesn't know why 
it's getting no video (or black video).  Detecting no incoming video and 
overlaying makes sense (though timeouts can be tricky to avoid false 
positives), but it's not a replacement for Mute and Hold from the source.

> 2. Switch between front and rear camera
> ========================================
> One solution: create a MediaStream (e.g. by calling getUserMedia twice
> immediately after each other, or by combining existing tracks into a new
> MediaStream) containing two video tracks (front+back cam). The video
> shown is toggled by enabling/disabling video tracks (the video element
> plays the first active one in the case of several video tracks).

This means toggling would require a renegotiation end-to-end before the 
switch occurs.  Massively annoying (delay, perhaps long) and means more 
complexity required for receiving - I should be able to switch cameras 
and your simple receiver app shouldn't need to do a bunch of footwork.

> Another: use add/removeTrack to switch video track being sent.

Ditto.

> 3. Play music on hold
> =====================
> One solution (audio only case): The site provides an URL to a (music)
> file to play when on hold. The peers app registers an event listener
> that listens to onmute on the incoming audio track. When onmute
> fires, the app changes the source of the audio element to the URL of the
> music file.

Again, this depends on the "onmute" notification somehow getting across, 
and there may be a silence gap until it does (unless it's sent inband in 
the RTP somehow (header extensions?), and then there are redundancy needs.

If one were developing a single, non-heterogeneous app one might do 
this, but relying on this in a heterogeneous world has problems.  (For 
example, connection to a PSTN gateway - you'd have to build this 
behavior into the WebRTC<->PSTN gateway and tightly specify it at the 
rtcweb level.)  This amounts to build a specification for an 
application, not for a set of tools and capabilities others can use to 
build apps.

> One solution if the communication is audio+video, and the intent is to
> just replace the audio (but continue playing video): assume a
> MediaStream with one audio and one video track is used. Play the audio
> in an audio element, the video in a (muted) video element. At on hold,
> replace the source of the audio element with the music file.

stream = getUserMedia();
audio.src = stream;
video.src = stream;
video.mute();
audiostream = audio.captureStreamUntilEnded();
videostream = video.captureStreamUntilEnded();
mergedstream = new MediaStream(audio track from audiostream, video track 
from videostream);
pc.addStream(mergedstream);

onmute: audio.src = music;

Problem (other than this is a bit painful and wasteful): the audio and 
video streams are now unsynchronized.

> A solution if the communication is audio+video, and the intent is to
> replace audio and video: either mute the tracks (at sending side);
> receiving end gets onmute event and replaces the .src of the video
> element with the URL to a video file, or signal on the data channel
> something that makes the other end replace the src of the video element
> with an URL to a file

There's no transmittal of these "onmute" events you're talking about 
across the RTP layer.

See the previous arguments about receiver-side display, timing, etc.

>
> Another solution is that the sending end does removeStream from
> PeerConnection. This fires onremovestream at the receiving side; that
> event can trigger the change of src to the video element. This solution
> frees up resources.

This requires a full renegotiation (which causes onremovestream to fire 
once complete).

>
> Note that once the recording function has been specced (in such a way
> that the recorded file can be the source of a MediaStream) more options
> will become available.

I'm already assuming something like that 
(video.captureStreamUntilEnded()) - see the MediaStream Processing API 
proposal.


-- 
Randell Jesup
randell-ietf@jesup.org


From stefan.lk.hakansson@ericsson.com  Mon Jun 18 00:21:10 2012
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 06/12/2012 05:26 PM, Harald Alvestrand wrote:
> On 06/12/2012 05:07 PM, Anant Narayanan wrote:
>> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>>> pc = new PeerConnection();
>>>
>>> GetUserMedia(..., function(stream) {
>>> stream1 = stream;
>>
>> As per my understanding of the gUM spec, this line:
>>
>>> stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);
>>
>> indicates that the audio and video tracks are *copied* into stream2,
>> and are not references.
> I have concluded from the discussions this week that we're not of one
> mind on this, and that various items, including this, make more sense if
> we see them as references.
>
> The actual language for the constructor is:
>
> "Addtracktostreamsaudio track list."
>
> There's nothing here about making a copy of /track/.
>
> Remember that the original specification (a year ago) had chained tracks
> feeding into other tracks; we simplified this model to a model where a
> track always links a source to a sink, not to another track.
>
> I think that our previous day's discussion has concluded that if we want
> to make a change to a track, and not have that change affect another
> track, we have to make a copy, and we have to make it explicit. For the
> (I think normal) case where we don't manipulate requirements on tracks,
> or are happy to have any changes affect all copies of the track, we
> don't need the copy.
>
>> Consider, for instance, that if a resolution change was made to a
>> track in stream2, the track from the same source in stream1 would
>> remain unchanged.
>>
>> Thus, I'd expect the following:
>>
>>> ..... stuff to start connecting ....
>>>
>>> pc.addStream(stream1);
>>> pc.addStream(stream2);
>>>
>>> offer = pc.createOffer(....);
>>>
>>> With MSID, the offer will look like this:
>>>
>>> ....
>>> m=audio
>>> a=ssrc:1234 msid=xyzzy a0
>>> a=ssrc:1234 msid=xyww a0
>>> ...
>>> m=video
>>> a=ssrc:1256 msid=xyzzy v0
>>> a=ssrc:1256 msid=xyww v0
>>
>> to look like:
>>
>> m=audio
>> a=ssrc:1234 cname:144a
>> a=ssrc:4321 cname:1e24
>> ...
>> m=video
>> a=ssrc:1256 cname:144a
>> a=ssrc:6521 cname:1e24
>>
>> instead.
>
> This markup (which is actually already standardized) says two things:
>
> - the sender has two synchronization contexts, 144a and 1e24
> - the recipient has no information about whether or not these are
> synchronized to the same clock

I prefer using the "msid=xyz a0" syntax for a couple of reasons:
* it does not overload the sync contexts (and lose some sync info in the 
process)
* it opens for optimization by sending a track only once even if 
included in more than one MediaStream

>
> This also means that the packets of each stream are transmitted twice,
> of course.
>> And, this:
>>
>>> and the expectation on the recipient side is that you will get
>>> pc.onaddstream called with:
>>>
>>> 1) stream with id = xyzzy, audio and video
>>> 2) stream with id = xyww, audio and video
>>
>> would remain the same, except with the IDs substituted by CNAMEs.
>
>
> That's a reasonable interpretation. Is it the one we want to make?
>
>
>


From stefan.lk.hakansson@ericsson.com  Mon Jun 18 05:08:03 2012
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Date: Mon, 18 Jun 2012 14:08:00 +0200
From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Subject: Re: [rtcweb] Mute implementations (Re: More Comments ondraft-ietf-rtcweb-use-cases-and-requirements-07)
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Hi Randell,

some feedback inline.

But overall, this was done more as a thought exercise. I'm not at all 
against being able to replace the source of a MediaStream(track) in 
operation. But I don't know when that should be added.

Br,
Stefan

On 06/17/2012 06:32 PM, Randell Jesup wrote:
> CC'd to both lists, as this affects both Media Capture (MediaStreams)
> and rtcweb protocol issues (and PeerConnection API issues, but I'm not
> yet including the WebRTC list).
>
> On 6/17/2012 10:33 AM, Stefan Hakansson LK wrote:
>> For the fun of it I thought a little while about what options that are
>> currently available with the APIs that are in the W3C docs as is. The
>> list is in no way exhaustive, there are other ways to do it.
>>
>> Perhaps this is not sufficient for what people want to do, but at least
>> there are some things you can do already:
>>
>> 1. Replace video with the display of an image
>> =============================================
>> One solution: Use the poster attribute of the video element. Quote
>> from the html spec: The poster attribute gives the address of an image
>> file that the user agent can show while no video data is available.. We
>> can spec it such that if no video track of a MediaStreamis active, that
>> counts as no data available. This way, as soon as the user mutes the
>> video, an image will replace it at display. The site can supply the
>> image (either a default by the site or one profile pic selected by the
>> user), or it can be provided over the data channel (as a blob).
>
> This is a receiver-side action and thus is not very useful for Mute and
> Hold operations.  This would depend on those stopping transmitting
> (which could cause other issues) and would infer that lack of reception
> means mute or hold - while it could also mean a temporary or permanent
> loss of connectivity, etc.

I think that a track should enter state "muted" (in JS) if no data is 
received, as this is in line with how "muted" is defined. If no data is 
received due to connection loss this will be detected on lack of 
ICE-messages (or RTCP?) I guess and become known to the app via the 
PeerConnection state changing from "open" to "connecting".

That said, I think the changed state of the track should be explicitly 
signaled. One proposal I have heard is to use the "sending" attribute in 
http://datatracker.ietf.org/doc/draft-lennox-mmusic-sdp-source-selection/?include_text=1, 
another (that I like better because timing should be better) is to use 
something like what is in 
http://datatracker.ietf.org/doc/draft-westerlund-avtext-rtp-stream-pause/ but 
for the send direction (basically "pause-indication" sent in RTCP). But 
perhaps in a first version the lack of RTP packets for a certain time 
could work as "mute" indication.

>
>> Another solution: A signal is sent on the data channel (possibly
>> accompanied by the image) on the data channel, the peers app overlays
>> the image on top of the video element as result (and switches back on
>> another signal).
>
> I've done this (using RTCP APP as a data channel), and it works well
> *if* both sides are the same application, and it still requires that the
> sources immediately mute (though black would be ok in this case).
> However, if you ever need to talk to something that doesn't understand
> this out-of-band "I'm muted" signal, then you really want to Mute/Hold
> with an image (and sound for Hold).  (And I've had to do this as well.)

Right, this requires the applications to understand each other. But that 
is already a prerequisite, isn't it? An arbitrary app would not 
understand that a PeerConnection object should be created just because 
an SDP offer was pushed to it from the server and so on.

>
>> A third (similar) solution: The app at the receiving end listens to
>> onunmute of the video track in question; once it fires an image is
>> overlayed.
>
> How does "onmute" work?  This similarly presupposes agreement on the
> out-of-band message, and also has the same issue when in a heterogeneous
> environment.

Re. "onmute", see above. Agree on that it presupposes agreement, but 
some kind of agreement is needed anyway.
>
>> A fourth: The peers app analyzes the video using canvas, and overlays
>> an image when there is no data in the video.
>
> a) ugh, what a waste of CPU cycles
> b) Has the same problem as the first - the receiver doesn't know why
> it's getting no video (or black video).  Detecting no incoming video and
> overlaying makes sense (though timeouts can be tricky to avoid false
> positives), but it's not a replacement for Mute and Hold from the source.

Agree to that it will consume CPU cycles.


>
>> 2. Switch between front and rear camera
>> ========================================
>> One solution: create a MediaStream (e.g. by calling getUserMedia twice
>> immediately after each other, or by combining existing tracks into a new
>> MediaStream) containing two video tracks (front+back cam). The video
>> shown is toggled by enabling/disabling video tracks (the video element
>> plays the first active one in the case of several video tracks).
>
> This means toggling would require a renegotiation end-to-end before the
> switch occurs.  Massively annoying (delay, perhaps long) and means more
> complexity required for receiving - I should be able to switch cameras
> and your simple receiver app shouldn't need to do a bunch of footwork.

I don't agree fully. The entire MediaStream (in this case consisting of 
one audio and two video tracks) is negotiated at the same time. But only 
one of the video tracks is actively transmitting RTP most of the time.

>
>> Another: use add/removeTrack to switch video track being sent.
>
> Ditto.

Here I agree that there would be a re-negotiation.

>
>> 3. Play music on hold
>> =====================
>> One solution (audio only case): The site provides an URL to a (music)
>> file to play when on hold. The peers app registers an event listener
>> that listens to onmute on the incoming audio track. When onmute
>> fires, the app changes the source of the audio element to the URL of the
>> music file.
>
> Again, this depends on the "onmute" notification somehow getting across,
> and there may be a silence gap until it does (unless it's sent inband in
> the RTP somehow (header extensions?), and then there are redundancy needs.

Yep, "onmute" must be trigged by something.

>
> If one were developing a single, non-heterogeneous app one might do
> this, but relying on this in a heterogeneous world has problems.  (For
> example, connection to a PSTN gateway - you'd have to build this
> behavior into the WebRTC<->PSTN gateway and tightly specify it at the
> rtcweb level.)  This amounts to build a specification for an
> application, not for a set of tools and capabilities others can use to
> build apps.

Again, I don't think you could develop two different clients in 
isolation anyway. Have not thought about the gateway issues.

>
>> One solution if the communication is audio+video, and the intent is to
>> just replace the audio (but continue playing video): assume a
>> MediaStream with one audio and one video track is used. Play the audio
>> in an audio element, the video in a (muted) video element. At on hold,
>> replace the source of the audio element with the music file.
>
> stream = getUserMedia();
> audio.src = stream;
> video.src = stream;
> video.mute();
> audiostream = audio.captureStreamUntilEnded();
> videostream = video.captureStreamUntilEnded();
> mergedstream = new MediaStream(audio track from audiostream, video track
> from videostream);
> pc.addStream(mergedstream);
>
> onmute: audio.src = music;
>
> Problem (other than this is a bit painful and wasteful): the audio and
> video streams are now unsynchronized.

I guess this part is a bit underspecified yet. It is said that tracks in 
a MediaStream should be synchronized, but not what happens when 
different tracks are played in different elements.

>
>> A solution if the communication is audio+video, and the intent is to
>> replace audio and video: either mute the tracks (at sending side);
>> receiving end gets onmute event and replaces the .src of the video
>> element with the URL to a video file, or signal on the data channel
>> something that makes the other end replace the src of the video element
>> with an URL to a file
>
> There's no transmittal of these "onmute" events you're talking about
> across the RTP layer.

I think we need something like that added (as discussed above) eventually.

>
> See the previous arguments about receiver-side display, timing, etc.
>
>>
>> Another solution is that the sending end does removeStream from
>> PeerConnection. This fires onremovestream at the receiving side; that
>> event can trigger the change of src to the video element. This solution
>> frees up resources.
>
> This requires a full renegotiation (which causes onremovestream to fire
> once complete).

Yes.

>
>>
>> Note that once the recording function has been specced (in such a way
>> that the recorded file can be the source of a MediaStream) more options
>> will become available.
>
> I'm already assuming something like that
> (video.captureStreamUntilEnded()) - see the MediaStream Processing API
> proposal.

It should get into the spec once designed.

>
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
Thread-Topic: [rtcweb] CNAME as a media stream identifier - what I don't think works
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Date: Tue, 19 Jun 2012 21:20:44 +0000
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On Jun 12, 2012, at 16:37 , Stefan Hakansson LK wrote:

> On 06/12/2012 05:07 PM, Anant Narayanan wrote:
>> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>>> pc =3D new PeerConnection();
>>>=20
>>> GetUserMedia(..., function(stream) {
>>>       stream1 =3D stream;
>>=20
>> As per my understanding of the gUM spec, this line:
>>=20
>>>       stream2 =3D new MediaStream(stream1.audioTracks, stream1.videoTra=
cks);
>>=20
>> indicates that the audio and video tracks are *copied* into stream2, and
>> are not references. Consider, for instance, that if a resolution change
>> was made to a track in stream2, the track from the same source in
>> stream1 would remain unchanged.
>=20
> This is a bit unclear to me. What would a resolution change really mean i=
n this case? To me the resolution (when a camera is used as source) would b=
e the resolution delivered by the camera. Can it deliver two (or more) reso=
lutions at the same time (perhaps also with different frame rates, aspect r=
ations and so on)?
>=20
> To me it makes most sense (if the application is going to explicitly ask =
for specific resolution, frame rate, aspec ration) to ask for it at getUser=
Media time. If the app does not ask for this explicitly, at least the suita=
ble resolution would be known by the browser whenever the video is played i=
n a video element (from the size of it).
>=20

I'm not sure but it looks like implementations are assuming that if getuser=
media asks fro the same camera twice, the second time it will be busy. I wa=
s executing the second stream to be cloned from the first one.=20

So lets say you had a 4k by 4k camera and wanted an HD stream for Alice and=
 SD for bob. You would get the first stream with constrains set to HD, then=
 clone that to second stream with constraints set to SD.

Sorry if two versions of this email arrived



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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Jonathan Lennox <jonathan@vidyo.com>
Thread-Topic: [rtcweb] What I think we need to do about ICE
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Date: Tue, 19 Jun 2012 21:20:44 +0000
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I think the word "restart" is causing some confusion. Better words might be=
 "reboot" meaning you want throw away everything you have, get new local po=
rts, and start over. And "continue" meaning you have new interfaces, or rel=
ays, or want more components or something but want to do some more gatherin=
g and/or checking.=20


On Jun 14, 2012, at 16:03 , Jonathan Lennox wrote:

> On Jun 12, 2012, at 10:32 AM, Eric Rescorla wrote:
>=20
>> To recap what I was saying on the list:
>>=20
>> 1. Some method of determining when a candidate pair has failed.
>>=20
>> 2. Some method of failing over to another candidate pair after
>> failure in as clean a fashion as possible, preferably exploiting
>> the fact that we already know about other valid candidate pairs,
>> if possible. Example: I had a direct and relayed pair that worked
>> and I chose direct. Is there a way to fall back to relayed without
>> doing a total re-ice.
>>=20
>> 3. When I discover a new candidate pair, some way of investigating
>> it and maybe changing to it w/o interfering with my current active
>> candidate pair.
>>=20
>> I think we need to do (1) and probably (2) now and (3) is a nice to have=
.
>=20
> From the point of view of RFC 5245, during an ICE Restart "media can cont=
inue to be sent to the previously validated pair." (9.1.1.1 and 11.1.1).  S=
o 3) should come for free, if implementations are written properly.
>=20
> Here's the model of ICE as I understand it.  An endpoint is always free t=
o gather as many or as few candidates as it wants, at any time before it st=
arts ICE or an ICE Restart.  This includes the currently-selected candidate=
, and any other previously-validated candidates that happen to still be liv=
e.
>=20
> Thus, an ICE Restart doesn't have to be a "total re-ice", necessarily.  I=
f you send (e.g.) your current selected candidate, and one other newly-disc=
overed candidate, as your candidate set for an ICE restart, the restart's c=
onnectivity checking should be lighter-weight than one where you give the l=
ist of every possible candidate (which you presumably would do for the sess=
ion's initial ICE exchange).  But it's still a full and compliant ICE conne=
ctivity check.
>=20
> That said, ICE says that once you're in "ICE Completed" state, the only w=
ay to change a component's selected candidate pair is by doing an ICE resta=
rt.
>=20
> Trickle candidates are equivalent to candidates sent in an updated offer =
or answer in the "ICE Running" state (9.1.2.1 / 9.2.2.1).  Thus, they're ig=
nored once an endpoint enters "ICE Completed" state.
>=20
>=20
> ICE leaves a lot of endpoint behavior up to endpoint implementors, and in=
 the webrtc context I think many of these behaviors will need to be control=
lable by the javascript applications.  I think the following APIs may be ne=
eded.  (Some of them may exist already in the w3 draft, I haven't followed =
all its details).
>=20
> * Some way to know when something interesting has happened to network con=
nectivity, above and beyond EKR's 1).  This includes a new network interfac=
e coming up, or even potentially (if possible) a something more interesting=
 like a captive portal becoming unblocked.
>=20
> * Some way to configure the set of interfaces on which address gathering =
should be done, and the priority among them.  (This implies there also need=
s to be way to interrogate the set of interfaces.)
>=20
> * Some way to start address gathering, both for the initial PeerConnectio=
n setup and for an ICE restart.  Note that this needs to be done both by th=
e offerer and the answerer, and in many cases an answerer won't know it nee=
ds to do so until it receives an offer with an ICE restart in it.
>=20
> * If you're a controlling endpoint, whether to use regular or aggressive =
nomination.  For regular nomination, once one or more candidate pairs have =
been validated, a way for the controlling endpoint to select the one to use=
.  (Note that the controlling endpoint may not necessarily be the offerer, =
if you receive an ice-lite offer.)
>=20
> --
> Jonathan Lennox
> jonathan@vidyo.com
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] What I think we need to do about ICE
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On Jun 12, 2012, at 4:53 , Bernard Aboba wrote:

> 3. It seems sensible to be able to cache recent tests and not rerun them.=
 If that were done then full offer/answers might only generate delta tests.

Depending on what sort of test you are thinking of, I don't think this will=
 work.=20

draft-jennings-behave-test-results-04 showed that for many nats, the primar=
y and secondary behavior is different. This makes caching any recent tests =
fraught with disaster. (I'll note some major voip products had really nasty=
 hard to debug voice connectivity failures due to this)

=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Eric Rescorla <ekr@rtfm.com>
Thread-Topic: [rtcweb] Relaying for privacy
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On May 16, 2012, at 18:09 , Eric Rescorla wrote:

> On Wed, May 16, 2012 at 2:54 PM, Justin Uberti <juberti@google.com> wrote=
:
>> My view was that we need to make it _possible_ for sites to respect priv=
acy
>> (via the API), and provide documentation that encourages sites to do so.=
 As
>> you point out, I think it's difficult for the browser to automatically
>> figure out what to do here, and when to do it.
>>=20
>> The identity of the "called" user may not be a factor in certain
>> applications (e.g. some p2p game), making it unclear as to whether forci=
ng
>> use of relay is even the right thing to do.
>=20
> I concur with all this.
>=20
> I think it's also important to reiterate that this is about the site coop=
erating
> to protect your privacy. If you want to protect the user's IP address fro=
m the
> site, then the user needs to use Torbutton or some such. And I would want
> Torbutton to arrange relay my RTCWEB traffic as well (though likely that
> can't go through Tor for performance reasons).
>=20
> -Ekr
>=20

Agree this is about the JS cooperating and the JS can decide when to do thi=
s but one side comment


If my browser is behind a NAT today, I don't think the JS can find out the =
private address. I could be wrong about this but I don't know how to do it.=
 Obviously the web server will see the IP of the outermost NAT. Once we can=
 create an offer, this will change, the JS will be able to see the private =
address.=20



From fluffy@cisco.com  Tue Jun 19 14:20:56 2012
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
Thread-Topic: [rtcweb] Combining consent freshness and liveness
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On Jun 14, 2012, at 14:58 , Bernard Aboba wrote:

> Agree that introducing a new method is not a good idea.  ICE has been imp=
lemented
> and deployed, and so we should not be breaking interoperability without a=
 very good
> reason. =20
>=20
> The cost of SHA-1 computation is not a good enough justification for the =
pain a=20
> new method would cause.=20

+1


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From: "Sunyang (Eric)" <eric.sun@huawei.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Eric Rescorla <ekr@rtfm.com>
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From randell-ietf@jesup.org  Tue Jun 19 22:08:50 2012
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On 6/19/2012 5:20 PM, Cullen Jennings (fluffy) wrote:
> On Jun 12, 2012, at 16:37 , Stefan Hakansson LK wrote:
>
>> On 06/12/2012 05:07 PM, Anant Narayanan wrote:
>>> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>>>> pc = new PeerConnection();
>>>>
>>>> GetUserMedia(..., function(stream) {
>>>>        stream1 = stream;
>>> As per my understanding of the gUM spec, this line:
>>>
>>>>        stream2 = new MediaStream(stream1.audioTracks, stream1.videoTracks);
>>> indicates that the audio and video tracks are *copied* into stream2, and
>>> are not references. Consider, for instance, that if a resolution change
>>> was made to a track in stream2, the track from the same source in
>>> stream1 would remain unchanged.
>> This is a bit unclear to me. What would a resolution change really mean in this case? To me the resolution (when a camera is used as source) would be the resolution delivered by the camera. Can it deliver two (or more) resolutions at the same time (perhaps also with different frame rates, aspect rations and so on)?
>>
>> To me it makes most sense (if the application is going to explicitly ask for specific resolution, frame rate, aspec ration) to ask for it at getUserMedia time. If the app does not ask for this explicitly, at least the suitable resolution would be known by the browser whenever the video is played in a video element (from the size of it).
>>
> I'm not sure but it looks like implementations are assuming that if getusermedia asks fro the same camera twice, the second time it will be busy. I was executing the second stream to be cloned from the first one.

Right, and that's what's described up higher (new MediaStream()). I 
don't think getUserMedia() is *obligated* to return busy, but I would 
expect it to.

>
> So lets say you had a 4k by 4k camera and wanted an HD stream for Alice and SD for bob. You would get the first stream with constrains set to HD, then clone that to second stream with constraints set to SD.

Ah, but here we disagree - there is no "clone with constraints", and 
even if there were you're implying a set of processing occur "somewhere" 
to make the cloned MediaStream.

MediaStream Processing could handle this (though I'd want to use a 
pre-defined scaler node, not do it in raw JS for perf and hardware-use 
reasons).

Cloning a 4k MediaStream and feeding them to the same (or different) 
PeerConnections would allow the PeerConnections to scale the video for 
encoding and transport.

-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Tue Jun 19 22:11:56 2012
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Subject: Re: [rtcweb] What I think we need to do about ICE
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On 6/19/2012 5:20 PM, Cullen Jennings (fluffy) wrote:
> On Jun 12, 2012, at 4:53 , Bernard Aboba wrote:
>
>> 3. It seems sensible to be able to cache recent tests and not rerun them. If that were done then full offer/answers might only generate delta tests.
> Depending on what sort of test you are thinking of, I don't think this will work.
>
> draft-jennings-behave-test-results-04 showed that for many nats, the primary and secondary behavior is different. This makes caching any recent tests fraught with disaster. (I'll note some major voip products had really nasty hard to debug voice connectivity failures due to this)

Agreed.  Especially since some other app or port use may push you into 
"secondary" without warning.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 19 June 2012 19:45, Sunyang (Eric) <eric.sun@huawei.com> wrote:
> I think js can get private ip address and corresponding port number, which I test with google chrome and apprtc, also some private demo.

That might be true if you have installed certain plugins, but not for
an unmodified installation.  At least until this
(http://www.w3.org/TR/netinfo-api/) gets deployed.

Proxies and NAT tend to hide addresses unless you add something like
what we are proposing to add.  Getting the address of a NAT is
actually worse for most residential users. RFC 1918 addresses are
usually what you get from the host candidates, which doesn't do much
in terms of privacy leakage, whereas the public address on the NAT is
usually only shared by a handful of people.

--Martin

From bernard_aboba@hotmail.com  Wed Jun 20 00:04:25 2012
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> > Depending on what sort of test you are thinking of=2C I don't think thi=
s will work.

[BA] I was referring to RFC 5245 Section 9.1.1.1:

   An agent sets the rest of the fields in the SDP for this media stream
   as it would in an initial offer of this media stream (see
   Section 4.3).  Consequently=2C the set of candidates MAY include some=2C
   none=2C or all of the previous candidates for that stream and MAY
   include a totally new set of candidates gathered as described in
   Section 4.1.1.=20
This gives the implementation considerable flexibility in determining what =
candidates to include in an ICE re-start.
 		 	   		  =

--_d04b656c-9c07-46b1-b9b2-7c4f6ebd9661_
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<html>
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{
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{
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
&gt=3B &gt=3B Depending on what sort of test you are thinking of=2C I don't=
 think this will work.<br><br>[BA] I was referring to RFC 5245 Section 9.1.=
1.1:<br><br><pre>   An agent sets the rest of the fields in the SDP for thi=
s media stream
   as it would in an initial offer of this media stream (see
   Section 4.3).  Consequently=2C the set of candidates MAY include some=2C
   none=2C or all of the previous candidates for that stream and MAY
   include a totally new set of candidates gathered as described in
   Section 4.1.1.</pre>&nbsp=3B<br>This gives the implementation considerab=
le flexibility in determining what candidates to include in an ICE re-start=
.<br> 		 	   		  </div></body>
</html>=

--_d04b656c-9c07-46b1-b9b2-7c4f6ebd9661_--

From pkyzivat@alum.mit.edu  Wed Jun 20 10:00:10 2012
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Subject: [rtcweb] PRANSWER and 3264
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ISTM that the introduction of JSEP with PRANSWER introduces behavior 
that is not addressed by the existing o/a spec RFC 3264.

3264 assumes that session management consists of a sequence of 
offer/answer pairs. It provides rules for how an answer is to be 
constructed in compliance with the matching offer, and rules for how a 
new offer is to be constructed in compliance with the prior o/a 
exchange. It does *not* define in any way the processing of two or more 
answers for a single offer.

3261 specifies how offers and answers are carried in SIP messages. It 
provides mechanisms that may seem to result in multiple answers to the 
same offer, but it puts rules on how those are constructed so that valid 
uses of SIP won't violate the assumptions of 3264. Specifically:

- when an sdp "answer" is carried in an unreliable response, that
   answer must be identical to one laster sent reliably. So its
   possible to treat the first one received as the answer, and not
   deal with changes in subsequent responses.

- when an INVITE is forked, the responses are identified by a
   unique dialog id for each responding UAS. The entire o/a protocol
   of 3264 is handled independently for each such dialog. Within each
   of the dialogs the assumptions of 3264 apply.

IIUC, PRANSWER is intended to be used in comparable situations to *both* 
of the above:

- a single dialog with a single endpoint, but where information about
   the desired session evolves over time. (E.g. new ICE candidates
   are discovered.)

- serial forking, where one candidate answer is received and processed,
   and then replaced by an alternative that has no relationship to
   the prior one.

(Note that a number of people have wanted such capability in SIP too. 
But so far it has been easier to tell them to use the existing 
facilities. In particular, cases that want an evolving answer can be 
handled by representing as a new dialog.)

So, I think one of two things is needed from rtcweb to resolve this:

1) a way to map a sequence of JSEP OFFER/*PRANSWER/ANSWER (consistent 
with Figure 2 of JSEP) to and from a sequence of offer/answer consistent 
with 3264.

2) a modification of 3264 that defines the rules for a sequence of 
answers to a single offer.

Of these, (1) is simpler. The simplistic way is to say that each 
PRANSWER and ANSWER is treated as *the* answer to the prior OFFER, 
independent of any prior answer.

But it might be useful to recognize there is a difference between 
evolving an answer and switching to a different fork. In the evolving 
case it would be useful to say what sorts of evolution are permitted and 
which are not.

	Thanks,
	Paul

From internet-drafts@ietf.org  Wed Jun 20 10:39:55 2012
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Overview: Real Time Protocols for Brower-based Applicati=
ons
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-04.txt
	Pages           : 22
	Date            : 2012-06-20

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - "real time communication on the Web".

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This document is a work item of the RTCWEB working group.



The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-overview

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-overview-04

A diff from previous version is available at:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-overview-04


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From christer.holmberg@ericsson.com  Wed Jun 20 11:04:20 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 20 Jun 2012 20:04:17 +0200
Thread-Topic: [rtcweb] PRANSWER and 3264
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Subject: Re: [rtcweb] PRANSWER and 3264
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Hi,

>ISTM that the introduction of JSEP with PRANSWER introduces behavior
>that is not addressed by the existing o/a spec RFC 3264.
>
>3264 assumes that session management consists of a sequence of
>offer/answer pairs. It provides rules for how an answer is to be
>constructed in compliance with the matching offer, and rules for how a
>new offer is to be constructed in compliance with the prior o/a
>exchange. It does *not* define in any way the processing of two or more
>answers for a single offer.
>
>3261 specifies how offers and answers are carried in SIP messages. It
>provides mechanisms that may seem to result in multiple answers to the
>same offer, but it puts rules on how those are constructed so that valid
>uses of SIP won't violate the assumptions of 3264. Specifically:
>
>- when an sdp "answer" is carried in an unreliable response, that
>   answer must be identical to one laster sent reliably. So its
>   possible to treat the first one received as the answer, and not
>   deal with changes in subsequent responses.
>
>- when an INVITE is forked, the responses are identified by a
>   unique dialog id for each responding UAS. The entire o/a protocol
>   of 3264 is handled independently for each such dialog. Within each
>   of the dialogs the assumptions of 3264 apply.
>
>IIUC, PRANSWER is intended to be used in comparable situations to *both*
>of the above:
>
>- a single dialog with a single endpoint, but where information about
>   the desired session evolves over time. (E.g. new ICE candidates
>   are discovered.)
>
>- serial forking, where one candidate answer is received and processed,
>   and then replaced by an alternative that has no relationship to
>   the prior one.
>
>(Note that a number of people have wanted such capability in SIP too.
>But so far it has been easier to tell them to use the existing
>facilities. In particular, cases that want an evolving answer can be
>handled by representing as a new dialog.)
>
>So, I think one of two things is needed from rtcweb to resolve this:
>
>1) a way to map a sequence of JSEP OFFER/*PRANSWER/ANSWER (consistent
>with Figure 2 of JSEP) to and from a sequence of offer/answer consistent
>with 3264.
>
>2) a modification of 3264 that defines the rules for a sequence of
>answers to a single offer.
>
>Of these, (1) is simpler. The simplistic way is to say that each
>PRANSWER and ANSWER is treated as *the* answer to the prior OFFER,
>independent of any prior answer.
>
>But it might be useful to recognize there is a difference between
>evolving an answer and switching to a different fork. In the evolving
>case it would be useful to say what sorts of evolution are permitted and
>which are not.

I don't see sending a *PRANSWER/ANSWER sequence to the browser as a big pro=
blem.

It's about updating the previous PRANSWER, and I think we already say that =
any (PR)ANSWER must follow 3264, and be a subset of the OFFER. It is suppor=
t of serial forking.

In my opinion, a bigger problem is if the browser, after it has received a =
JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In the case of SIP =
interworking, we can't simply map those to sequential different SDP answers=
 - at least not with using the same To tag value.

Regards,

Christer=

From dwing@cisco.com  Wed Jun 20 13:58:30 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Martin Thomson'" <martin.thomson@gmail.com>, "'Sunyang \(Eric\)'" <eric.sun@huawei.com>
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Subject: Re: [rtcweb] =?gb2312?b?tPC4tDogUmVsYXlpbmcgZm9yIHByaXZhY3k=?=
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Martin Thomson
> Sent: Tuesday, June 19, 2012 10:24 PM
> To: Sunyang (Eric)
> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
> Subject: Re: [rtcweb] =B4=F0=B8=B4: Relaying for privacy
>=20
> On 19 June 2012 19:45, Sunyang (Eric) <eric.sun@huawei.com> wrote:
> > I think js can get private ip address and corresponding port number,
> which I test with google chrome and apprtc, also some private demo.
>=20
> That might be true if you have installed certain plugins, but not for
> an unmodified installation.  At least until this
> (http://www.w3.org/TR/netinfo-api/) gets deployed.
>=20
> Proxies and NAT tend to hide addresses unless you add something like
> what we are proposing to add.  Getting the address of a NAT is
> actually worse for most residential users. RFC 1918 addresses are
> usually what you get from the host candidates, which doesn't do much
> in terms of privacy leakage, whereas the public address on the NAT is
> usually only shared by a handful of people.

RTCWEB lacks a precise problem statement on what we mean by "privacy".

Your view seems valid for a typical home user where just about everyone
is using 192.168.1.0/24.  A typical enterprise IT administrator,=20
however, does not want their internal network address assignments and
topology (host candidates) mapped from the outside.  Some IT=20
administrators strip SMTP "Received:" lines to prevent that=20
disclosure.

-d



From pkyzivat@alum.mit.edu  Wed Jun 20 15:45:05 2012
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Subject: Re: [rtcweb] PRANSWER and 3264
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On 6/20/12 8:04 PM, Christer Holmberg wrote:
> Hi,
>
>> ISTM that the introduction of JSEP with PRANSWER introduces behavior
>> that is not addressed by the existing o/a spec RFC 3264.
>>
>> 3264 assumes that session management consists of a sequence of
>> offer/answer pairs. It provides rules for how an answer is to be
>> constructed in compliance with the matching offer, and rules for how a
>> new offer is to be constructed in compliance with the prior o/a
>> exchange. It does *not* define in any way the processing of two or more
>> answers for a single offer.
>>
>> 3261 specifies how offers and answers are carried in SIP messages. It
>> provides mechanisms that may seem to result in multiple answers to the
>> same offer, but it puts rules on how those are constructed so that valid
>> uses of SIP won't violate the assumptions of 3264. Specifically:
>>
>> - when an sdp "answer" is carried in an unreliable response, that
>>    answer must be identical to one laster sent reliably. So its
>>    possible to treat the first one received as the answer, and not
>>    deal with changes in subsequent responses.
>>
>> - when an INVITE is forked, the responses are identified by a
>>    unique dialog id for each responding UAS. The entire o/a protocol
>>    of 3264 is handled independently for each such dialog. Within each
>>    of the dialogs the assumptions of 3264 apply.
>>
>> IIUC, PRANSWER is intended to be used in comparable situations to *both*
>> of the above:
>>
>> - a single dialog with a single endpoint, but where information about
>>    the desired session evolves over time. (E.g. new ICE candidates
>>    are discovered.)
>>
>> - serial forking, where one candidate answer is received and processed,
>>    and then replaced by an alternative that has no relationship to
>>    the prior one.
>>
>> (Note that a number of people have wanted such capability in SIP too.
>> But so far it has been easier to tell them to use the existing
>> facilities. In particular, cases that want an evolving answer can be
>> handled by representing as a new dialog.)
>>
>> So, I think one of two things is needed from rtcweb to resolve this:
>>
>> 1) a way to map a sequence of JSEP OFFER/*PRANSWER/ANSWER (consistent
>> with Figure 2 of JSEP) to and from a sequence of offer/answer consistent
>> with 3264.
>>
>> 2) a modification of 3264 that defines the rules for a sequence of
>> answers to a single offer.
>>
>> Of these, (1) is simpler. The simplistic way is to say that each
>> PRANSWER and ANSWER is treated as *the* answer to the prior OFFER,
>> independent of any prior answer.
>>
>> But it might be useful to recognize there is a difference between
>> evolving an answer and switching to a different fork. In the evolving
>> case it would be useful to say what sorts of evolution are permitted and
>> which are not.
>
> I don't see sending a *PRANSWER/ANSWER sequence to the browser as a big problem.
>
> It's about updating the previous PRANSWER, and I think we already say that any (PR)ANSWER must follow 3264, and be a subset of the OFFER. It is support of serial forking.

I don't know that it is so clear. At the end of the day, what is legal, 
and what is not, and what it means, must be written down somewhere in a 
normative way.

And I realized that my "simplistic" way above doesn't quite work, 
because of the requirements on o-lines.

> In my opinion, a bigger problem is if the browser, after it has received a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In the case of SIP interworking, we can't simply map those to sequential different SDP answers - at least not with using the same To tag value.

Yes, how to handle things in that direction must also be specified.

	Thanks,
	Paul

> Regards,
>
> Christer


From bernard_aboba@hotmail.com  Wed Jun 20 15:53:48 2012
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The Program Committee has received inquiries about remote participation at =
the IAB/IRTF Congestion Control Workshop to be held on June 28=2C 2012 in V=
ancouver=2C Canada.  While participants are strongly encouraged to attend t=
he meeting in person=2C arrangements
 for remote participation are planned (for those with accepted papers).   T=
herefore if you were considering submitting a paper but could not make the =
workshop in person=2C we would encourage you to go ahead and submit=3B  ple=
ase indicate in your submission if you
 are only able to participate remotely.
=20

Call for Papers
=20
The IAB and IRTF will hold a workshop on "Congestion Control for Interactiv=
e Real-Time Communication" in Vancouver=2C Canada on Saturday=2C July 28th=
=2C 2012 prior to the IETF-84 meeting.  More information on the workshop is=
 available at  http://www.iab.org/cc-workshop/

=20
This is an invitation-only workshop.=20
=20
Every prospective workshop participant must submit a position paper. The po=
sition paper reflects your views on a particular area of the workshop theme=
. We are interested in your opinion as an expert rather than your official =
company position.=20
 Keep your submission short and focused.  The focus should be on the proble=
ms and challenges that exist=2C rather than a detailed solution.  If your e=
xpertise allows you to write about a range of topics=2C please pick the one=
 topic you think would bring the most
 value to the workshop.=20
=20
Papers up to 3 pages=2C formatted in HTML=2C PDF=2C or plain text (for exam=
ple=2C as a submitted Internet-Draft) are ideal.  Authors of accepted paper=
s will be invited to the workshop.  Accepted position papers will be publis=
hed.=20

=20
Please use the following Website for submitting your papers:

http://iab.org/ccirtc/paper?p=3Dnew


Important Dates=20
=20
Position paper submission deadline: June 23=2C 2012=20
Notification to paper authors: June 30=2C 2012=20
Workshop date: July 28=2C 2012=20
=20
Note: please contact us if you are interested in submitting a paper but hav=
e an issue with the submission deadline.=20

=20
IPR Policy=20
=20


Due to the close relationship to ongoing IRTF and IETF work=2C the IPR poli=
cies described in RFC 5378 and RFC 3979 apply=2C both to submitted position=
 papers and to discussions at the workshop.=20
=20
Privacy Notice=20
=20
Paper submissions have to contain a name and an email address. We use this =
information to subscribe you to a mailing list for sharing workshop related=
 information prior to the workshop (e.g.=2C updates regarding the meeting v=
enue=2C feedback
 on the position paper submissions). This specially created mailing list is=
 only used for the duration of the workshop and will be deleted after the p=
ublication of the workshop report (which may take several months). You may =
at any time decide to unsubscribe
 from the mailing list (at your risk of missing information workshop relate=
d postings).=20

=20
Your name will be listed in the meeting minutes and the workshop report. We=
 will also publish all accepted position papers.

=20
There will be an audio recording of the workshop discussion. This workshop =
recording will not be distributed to the workshop participants nor to the p=
ublic=3B the purpose of the recording is to ensure that the workshop report=
 and the meeting
 minutes accurately reflect the discussions. =20
=20
Meeting Venue=20
=20
The workshop will be held in Vancouver=2C Canada on Saturday=2C July 28th=
=2C 2012 prior to the IETF-84 meeting (see http://www.ietf.org/meeting/84/i=
ndex.html).  Additional details about the meeting venue will be provided to=
 authors of accepted
 papers. =20
=20
While participants are strongly encouraged to attend the meeting in person=
=2C arrangements for remote participation are planned (for those with accep=
ted papers).   Please indicate in your submission if you are only able to p=
articipate remotely.

=20
Sponsorship=20
=20
If you are interested to help us working towards better interactive media c=
ongestion control mechanisms on the Internet (such as by making a contribut=
ion towards catering costs and room rental)=2C please contact us!

=20
Contact Information=20
=20
In case of questions please send email to mary.ietf.barnes at gmail.com
 		 	   		  =

--_54b9ab29-76c1-4856-8f64-10d4ea80f896_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
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<body class=3D'hmmessage'><div dir=3D'ltr'>
The Program Committee has received inquiries about remote participation at =
the IAB/IRTF Congestion Control Workshop to be held on June 28=2C 2012 in V=
ancouver=2C Canada.&nbsp=3B While participants are strongly encouraged to a=
ttend the meeting in person=2C arrangements
 for remote participation are planned (for those with accepted papers).&nbs=
p=3B&nbsp=3B Therefore if you were considering submitting a paper but could=
 not make the workshop in person=2C we would encourage you to go ahead and =
submit=3B&nbsp=3B please indicate in your submission if you
 are only able to participate remotely.<div><div class=3D"ecxWordSection1">
<p class=3D"ecxMsoNormal">&nbsp=3B</p>

<p class=3D"ecxMsoNormal"><b>Call for Papers</b></p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">The IAB and IRTF will hold a workshop on "Congest=
ion Control for Interactive Real-Time Communication" in Vancouver=2C Canada=
 on Saturday=2C July 28th=2C 2012 prior to the IETF-84 meeting.&nbsp=3B Mor=
e information on the workshop is available at&nbsp=3B http://www.iab.org/cc=
-workshop/
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">This is an invitation-only workshop. </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Every prospective workshop participant must submi=
t a position paper. The position paper reflects your views on a particular =
area of the workshop theme. We are interested in your opinion as an expert =
rather than your official company position.&nbsp=3B
 Keep your submission short and focused.&nbsp=3B The focus should be on the=
 problems and challenges that exist=2C rather than a detailed solution.&nbs=
p=3B If your expertise allows you to write about a range of topics=2C pleas=
e pick the one topic you think would bring the most
 value to the workshop. </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Papers up to 3 pages=2C formatted in HTML=2C PDF=
=2C or plain text (for example=2C as a submitted Internet-Draft) are ideal.=
&nbsp=3B Authors of accepted papers will be invited to the workshop.&nbsp=
=3B Accepted position papers will be published.&nbsp=3B
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Please use the following Website for submitting y=
our papers:
</p>
<p class=3D"ecxMsoNormal">http://iab.org/ccirtc/paper?p=3Dnew</p><p class=
=3D"ecxMsoNormal"><br></p>
<p class=3D"ecxMsoNormal"></p>
<p class=3D"ecxMsoNormal"><b>Important Dates </b></p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Position paper submission deadline: June 23=2C 20=
12 </p>
<p class=3D"ecxMsoNormal">Notification to paper authors: June 30=2C 2012 </=
p>
<p class=3D"ecxMsoNormal">Workshop date: July 28=2C 2012 </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Note: please contact us if you are interested in =
submitting a paper but have an issue with the submission deadline.&nbsp=3B
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal"><b>IPR Policy </b></p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>


<p class=3D"ecxMsoNormal">Due to the close relationship to ongoing IRTF and=
 IETF work=2C the IPR policies described in RFC 5378 and RFC 3979 apply=2C =
both to submitted position papers and to discussions at the workshop. </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal"><b>Privacy Notice </b></p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Paper submissions have to contain a name and an e=
mail address. We use this information to subscribe you to a mailing list fo=
r sharing workshop related information prior to the workshop (e.g.=2C updat=
es regarding the meeting venue=2C feedback
 on the position paper submissions). This specially created mailing list is=
 only used for the duration of the workshop and will be deleted after the p=
ublication of the workshop report (which may take several months). You may =
at any time decide to unsubscribe
 from the mailing list (at your risk of missing information workshop relate=
d postings).&nbsp=3B
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">Your name will be listed in the meeting minutes a=
nd the workshop report. We will also publish all accepted position papers.
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">There will be an audio recording of the workshop =
discussion. This workshop recording will not be distributed to the workshop=
 participants nor to the public=3B the purpose of the recording is to ensur=
e that the workshop report and the meeting
 minutes accurately reflect the discussions.&nbsp=3B </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal"><b>Meeting Venue</b> </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">The workshop will be held in Vancouver=2C Canada =
on Saturday=2C July 28th=2C 2012 prior to the IETF-84 meeting (see http://w=
ww.ietf.org/meeting/84/index.html).&nbsp=3B Additional details about the me=
eting venue will be provided to authors of accepted
 papers.&nbsp=3B </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">While participants are strongly encouraged to att=
end the meeting in person=2C arrangements for remote participation are plan=
ned (for those with accepted papers).&nbsp=3B&nbsp=3B Please indicate in yo=
ur submission if you are only able to participate remotely.
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal"><b>Sponsorship</b> </p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">If you are interested to help us working towards =
better interactive media congestion control mechanisms on the Internet (suc=
h as by making a contribution towards catering costs and room rental)=2C pl=
ease contact us!
</p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal"><b>Contact Information </b></p>
<p class=3D"ecxMsoNormal">&nbsp=3B</p>
<p class=3D"ecxMsoNormal">In case of questions please send email to mary.ie=
tf.barnes at gmail.com</p>
</div></div> 		 	   		  </div></body>
</html>=

--_54b9ab29-76c1-4856-8f64-10d4ea80f896_--

From ekr@rtfm.com  Wed Jun 20 16:43:00 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 20 Jun 2012 16:42:18 -0700
Message-ID: <CABcZeBO0pRhLa-rg0gOC-bWw0b-YfZcM60P-0TWjCVP_=ZQWQw@mail.gmail.com>
To: Dan Wing <dwing@cisco.com>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, rtcweb@ietf.org
Subject: Re: [rtcweb] =?gb2312?b?tPC4tDogUmVsYXlpbmcgZm9yIHByaXZhY3k=?=
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On Wed, Jun 20, 2012 at 1:58 PM, Dan Wing <dwing@cisco.com> wrote:
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Martin Thomson
>> Sent: Tuesday, June 19, 2012 10:24 PM
>> To: Sunyang (Eric)
>> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
>> Subject: Re: [rtcweb] =B4=F0=B8=B4: Relaying for privacy
>>
>> On 19 June 2012 19:45, Sunyang (Eric) <eric.sun@huawei.com> wrote:
>> > I think js can get private ip address and corresponding port number,
>> which I test with google chrome and apprtc, also some private demo.
>>
>> That might be true if you have installed certain plugins, but not for
>> an unmodified installation.  At least until this
>> (http://www.w3.org/TR/netinfo-api/) gets deployed.
>>
>> Proxies and NAT tend to hide addresses unless you add something like
>> what we are proposing to add.  Getting the address of a NAT is
>> actually worse for most residential users. RFC 1918 addresses are
>> usually what you get from the host candidates, which doesn't do much
>> in terms of privacy leakage, whereas the public address on the NAT is
>> usually only shared by a handful of people.
>
> RTCWEB lacks a precise problem statement on what we mean by "privacy".
>
> Your view seems valid for a typical home user where just about everyone
> is using 192.168.1.0/24.  A typical enterprise IT administrator,
> however, does not want their internal network address assignments and
> topology (host candidates) mapped from the outside.  Some IT
> administrators strip SMTP "Received:" lines to prevent that
> disclosure.

Dan,

When you say "internal" are you thinking of host candidates for NATed or
non-NATed machines? Obviously, for non-NATed, any Web site will see
that.

-Ekr

the

From dwing@cisco.com  Wed Jun 20 17:34:52 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Eric Rescorla'" <ekr@rtfm.com>
References: <CABkgnnWEqqAD0rPrFK3mj5jBCoqmyPTY_PFgVyg8t84egN0Baw@mail.gmail.com> <CAOJ7v-2K9VS0YKo=RJSPLcmT=X-kTQ3ynyp9APin92eW=LceTg@mail.gmail.com> <CABcZeBPDJ6MQgQFBW4+Gnq9V-Oo-Gb9d_7nuCnySzbFBzDoXVg@mail.gmail.com> <8F3DC910-5D40-4D8E-A44B-91686ABD930E@cisco.com> <9254B5E6361B1648AFC00BA447E6E8C32AEA9B00@szxeml545-mbx.china.huawei.com> <CABkgnnXadEzmYUNBHD6X986eZiGYmcOZ-pJqM83fx07DHpjCeQ@mail.gmail.com> <01b701cd4f27$70406680$50c13380$@com> <CABcZeBO0pRhLa-rg0gOC-bWw0b-YfZcM60P-0TWjCVP_=ZQWQw@mail.gmail.com>
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> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@rtfm.com]
> Sent: Wednesday, June 20, 2012 4:42 PM
> To: Dan Wing
> Cc: Martin Thomson; Sunyang (Eric); Cullen Jennings (fluffy);
> rtcweb@ietf.org
> Subject: Re: [rtcweb] =B4=F0=B8=B4: Relaying for privacy
>=20
> On Wed, Jun 20, 2012 at 1:58 PM, Dan Wing <dwing@cisco.com> wrote:
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >> Behalf Of Martin Thomson
> >> Sent: Tuesday, June 19, 2012 10:24 PM
> >> To: Sunyang (Eric)
> >> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org
> >> Subject: Re: [rtcweb] =B4=F0=B8=B4: Relaying for privacy
> >>
> >> On 19 June 2012 19:45, Sunyang (Eric) <eric.sun@huawei.com> wrote:
> >> > I think js can get private ip address and corresponding port
> number,
> >> which I test with google chrome and apprtc, also some private demo.
> >>
> >> That might be true if you have installed certain plugins, but not
> for
> >> an unmodified installation.  At least until this
> >> (http://www.w3.org/TR/netinfo-api/) gets deployed.
> >>
> >> Proxies and NAT tend to hide addresses unless you add something =
like
> >> what we are proposing to add.  Getting the address of a NAT is
> >> actually worse for most residential users. RFC 1918 addresses are
> >> usually what you get from the host candidates, which doesn't do =
much
> >> in terms of privacy leakage, whereas the public address on the NAT
> is
> >> usually only shared by a handful of people.
> >
> > RTCWEB lacks a precise problem statement on what we mean by
> "privacy".
> >
> > Your view seems valid for a typical home user where just about
> everyone
> > is using 192.168.1.0/24.  A typical enterprise IT administrator,
> > however, does not want their internal network address assignments =
and
> > topology (host candidates) mapped from the outside.  Some IT
> > administrators strip SMTP "Received:" lines to prevent that
> > disclosure.
>=20
> Dan,
>=20
> When you say "internal" are you thinking of host candidates for NATed
> or non-NATed machines? Obviously, for non-NATed, any Web site will see
> that.

I'm thinking of hosts behind a NAPT44 device, where the network
administrator feels the IP address sharing and IP address rewriting,
provided by that NAPT44, provides concealment of activities of
individual hosts.  The same feeling exists with IPv6 with IPv6
privacy addresses.  And perhaps that's what WEBRTC should do with
IPv6 is require a new privacy address for each "call".  (For
others' benefit, see last sentence of=20
http://tools.ietf.org/html/rfc4941#section-3.1)

-d



From ekr@rtfm.com  Wed Jun 20 18:35:15 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 20 Jun 2012 18:34:31 -0700
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Subject: Re: [rtcweb] =?utf-8?b?562U5aSNOiBSZWxheWluZyBmb3IgcHJpdmFjeQ==?=
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On Wed, Jun 20, 2012 at 5:34 PM, Dan Wing <dwing@cisco.com> wrote:
>
> I'm thinking of hosts behind a NAPT44 device, where the network
> administrator feels the IP address sharing and IP address rewriting,
> provided by that NAPT44, provides concealment of activities of
> individual hosts.

Hmm... I'm not sure this really applies to the Web threat model,
since it's easy to use cookies and/or fingerprinting techniques
to correlate activity between multiple sessions regardless of
the IP.

-Ekr


=A0>The same feeling exists with IPv6 with IPv6
> privacy addresses. =A0And perhaps that's what WEBRTC should do with
> IPv6 is require a new privacy address for each "call". =A0(For
> others' benefit, see last sentence of
> http://tools.ietf.org/html/rfc4941#section-3.1)

From ekr@rtfm.com  Wed Jun 20 19:04:59 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 20 Jun 2012 19:04:17 -0700
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Subject: [rtcweb] Randomly-generated CNAMEs
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RFC 6222 specifies two CNAME generation algorithms:

- Persistent CNAMEs based on UUIDs
- Short-term CNAMEs based on a digest of the MAC address and time of day.

However, reading this document and the security considerations it seems like
both methods produce CNAMEs which aren't unlinkable. Obviously the
persistent CNAMEs are linkable, and it doesn't look like short-term CNAMEs
have enough entropy to be unlinkable either. Specifically, the inputs are
(Section 5):

1. The MAC address (or EUI-64)
2. The time of day (NTP-style)
3. Initial SSRC, and IP/port quartet

(3) is known to the other side, (2) is at most 48-bits (and
practically much less)
and (2) is known to within a fairly narrow margin (and, it appears, communicated
in the RTP timestamp).

The upshot is that it seems likely that I can link two RTP sessions
from the same
person. Note that RFC 6222 doesn't say otherwise. Rather, it says:

 This document uses a hash function to ensure the uniqueness of RTCP
   CNAMEs.  A cryptographic hash function is used because such functions
   provide the randomness properties that are needed.  However, no
   security assumptions are made on the hash function.  The hash
   function is not assumed to be collision resistant, preimage
   resistant, or second preimage resistant in an adversarial setting; as
   described above, an attacker attempting an impersonation attack could
   merely set the RTCP CNAME directly rather than attacking the hash
   function.  Similarly, the hash function is not assumed to be a one-
   way function or pseudorandom in a cryptographic sense.

For privacy reasons, it would be nice to be able to generate CNAMEs randomly
so that they're not linkable. Unfortunately, 6222 seems to require
that short-term
credentials be generated via the above mechanism. Is there any reason why
that shouldn't be allowed?

-Ekr

From dwing@cisco.com  Wed Jun 20 19:51:12 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Eric Rescorla'" <ekr@rtfm.com>
References: <CABkgnnWEqqAD0rPrFK3mj5jBCoqmyPTY_PFgVyg8t84egN0Baw@mail.gmail.com> <CAOJ7v-2K9VS0YKo=RJSPLcmT=X-kTQ3ynyp9APin92eW=LceTg@mail.gmail.com> <CABcZeBPDJ6MQgQFBW4+Gnq9V-Oo-Gb9d_7nuCnySzbFBzDoXVg@mail.gmail.com> <8F3DC910-5D40-4D8E-A44B-91686ABD930E@cisco.com> <9254B5E6361B1648AFC00BA447E6E8C32AEA9B00@szxeml545-mbx.china.huawei.com> <CABkgnnXadEzmYUNBHD6X986eZiGYmcOZ-pJqM83fx07DHpjCeQ@mail.gmail.com> <01b701cd4f27$70406680$50c13380$@com> <CABcZeBO0pRhLa-rg0gOC-bWw0b-YfZcM60P-0TWjCVP_=ZQWQw@mail.gmail.com> <02e301cd4f45$ac06d8c0$04148a40$@com> <CABcZeBPPxUDxYOQujsxrJ2v_9xetoFhLYVH35LtPXLOSJv_5bA@mail.gmail.com>
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> -----Original Message-----
> From: Eric Rescorla [mailto:ekr@rtfm.com]
> Sent: Wednesday, June 20, 2012 6:35 PM
> To: Dan Wing
> Cc: Martin Thomson; Sunyang (Eric); Cullen Jennings (fluffy);
> rtcweb@ietf.org
> Subject: Re: [rtcweb] =E7=AD=94=E5=A4=8D: Relaying for privacy
>=20
> On Wed, Jun 20, 2012 at 5:34 PM, Dan Wing <dwing@cisco.com> wrote:
> >
> > I'm thinking of hosts behind a NAPT44 device, where the network
> > administrator feels the IP address sharing and IP address rewriting,
> > provided by that NAPT44, provides concealment of activities of
> > individual hosts.
>=20
> Hmm... I'm not sure this really applies to the Web threat model,
> since it's easy to use cookies and/or fingerprinting techniques
> to correlate activity between multiple sessions regardless of
> the IP.

Are you saying that ecause security can be compromised at layer=20
N, we should not provide security at layer Y?

-d


> -Ekr
>=20
>=20
>  >The same feeling exists with IPv6 with IPv6
> > privacy addresses.  And perhaps that's what WEBRTC should do with
> > IPv6 is require a new privacy address for each "call".  (For
> > others' benefit, see last sentence of
> > http://tools.ietf.org/html/rfc4941#section-3.1)


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Subject: Re: [rtcweb] PRANSWER and 3264
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On 06/21/2012 12:45 AM, Paul Kyzivat wrote:
>> In my opinion, a bigger problem is if the browser, after it has 
>> received a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In 
>> the case of SIP interworking, we can't simply map those to sequential 
>> different SDP answers - at least not with using the same To tag value.
>
> Yes, how to handle things in that direction must also be specified.

Isn't that up to the application, which presumably knows it's talking to 
a SIP gateway, and can just decide to "not do that", just as it decides 
to take no action on ICE candidate callbacks, since SIP doesn't support 
trickle candidates?

            Harald


From csp@csperkins.org  Thu Jun 21 03:22:07 2012
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To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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Ekr,

I'm not sure I understand what problem you're trying to solve: the point =
of the CNAME is to allow different RTP sessions to be linked.=20

Colin


On 21 Jun 2012, at 03:04, Eric Rescorla wrote:
> RFC 6222 specifies two CNAME generation algorithms:
>=20
> - Persistent CNAMEs based on UUIDs
> - Short-term CNAMEs based on a digest of the MAC address and time of =
day.
>=20
> However, reading this document and the security considerations it =
seems like both methods produce CNAMEs which aren't unlinkable. =
Obviously the persistent CNAMEs are linkable, and it doesn't look like =
short-term CNAMEs have enough entropy to be unlinkable either. =
Specifically, the inputs are (Section 5):
>=20
> 1. The MAC address (or EUI-64)
> 2. The time of day (NTP-style)
> 3. Initial SSRC, and IP/port quartet
>=20
> (3) is known to the other side, (2) is at most 48-bits (and =
practically much less) and (2) is known to within a fairly narrow margin =
(and, it appears, communicated in the RTP timestamp).
>=20
> The upshot is that it seems likely that I can link two RTP sessions =
from the same person. Note that RFC 6222 doesn't say otherwise. Rather, =
it says:
>=20
> This document uses a hash function to ensure the uniqueness of RTCP
>   CNAMEs.  A cryptographic hash function is used because such =
functions
>   provide the randomness properties that are needed.  However, no
>   security assumptions are made on the hash function.  The hash
>   function is not assumed to be collision resistant, preimage
>   resistant, or second preimage resistant in an adversarial setting; =
as
>   described above, an attacker attempting an impersonation attack =
could
>   merely set the RTCP CNAME directly rather than attacking the hash
>   function.  Similarly, the hash function is not assumed to be a one-
>   way function or pseudorandom in a cryptographic sense.
>=20
> For privacy reasons, it would be nice to be able to generate CNAMEs =
randomly so that they're not linkable. Unfortunately, 6222 seems to =
require that short-term credentials be generated via the above =
mechanism. Is there any reason why that shouldn't be allowed?
>=20
> -Ekr
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



--=20
Colin Perkins
http://csperkins.org/




From fluffy@iii.ca  Thu Jun 21 05:39:45 2012
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One again, I think we have complete failure on the meaning of the word =
session :-)=20

I think what EKR was getting at is that if A call B in one phone call, =
then day later A wants to make an anonymous call to B, B should not be =
able to tell the second call is coming from same devices. I think that =
was part of the goal of 6222 but from EKR's email it looks like it fails =
to provide that.=20



On Jun 21, 2012, at 6:22 , Colin Perkins wrote:

> Ekr,
>=20
> I'm not sure I understand what problem you're trying to solve: the =
point of the CNAME is to allow different RTP sessions to be linked.=20
>=20
> Colin
>=20
>=20
> On 21 Jun 2012, at 03:04, Eric Rescorla wrote:
>> RFC 6222 specifies two CNAME generation algorithms:
>>=20
>> - Persistent CNAMEs based on UUIDs
>> - Short-term CNAMEs based on a digest of the MAC address and time of =
day.
>>=20
>> However, reading this document and the security considerations it =
seems like both methods produce CNAMEs which aren't unlinkable. =
Obviously the persistent CNAMEs are linkable, and it doesn't look like =
short-term CNAMEs have enough entropy to be unlinkable either. =
Specifically, the inputs are (Section 5):
>>=20
>> 1. The MAC address (or EUI-64)
>> 2. The time of day (NTP-style)
>> 3. Initial SSRC, and IP/port quartet
>>=20
>> (3) is known to the other side, (2) is at most 48-bits (and =
practically much less) and (2) is known to within a fairly narrow margin =
(and, it appears, communicated in the RTP timestamp).
>>=20
>> The upshot is that it seems likely that I can link two RTP sessions =
from the same person. Note that RFC 6222 doesn't say otherwise. Rather, =
it says:
>>=20
>> This document uses a hash function to ensure the uniqueness of RTCP
>>  CNAMEs.  A cryptographic hash function is used because such =
functions
>>  provide the randomness properties that are needed.  However, no
>>  security assumptions are made on the hash function.  The hash
>>  function is not assumed to be collision resistant, preimage
>>  resistant, or second preimage resistant in an adversarial setting; =
as
>>  described above, an attacker attempting an impersonation attack =
could
>>  merely set the RTCP CNAME directly rather than attacking the hash
>>  function.  Similarly, the hash function is not assumed to be a one-
>>  way function or pseudorandom in a cryptographic sense.
>>=20
>> For privacy reasons, it would be nice to be able to generate CNAMEs =
randomly so that they're not linkable. Unfortunately, 6222 seems to =
require that short-term credentials be generated via the above =
mechanism. Is there any reason why that shouldn't be allowed?
>>=20
>> -Ekr
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt


From fluffy@iii.ca  Thu Jun 21 05:47:23 2012
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Subject: Re: [rtcweb] CNAME as a media stream identifier - what I don't think works
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On Jun 20, 2012, at 1:08 , Randell Jesup wrote:

> On 6/19/2012 5:20 PM, Cullen Jennings (fluffy) wrote:
>> On Jun 12, 2012, at 16:37 , Stefan Hakansson LK wrote:
>>=20
>>> On 06/12/2012 05:07 PM, Anant Narayanan wrote:
>>>> On 6/12/12 12:14 PM, Harald Alvestrand wrote:
>>>>> pc =3D new PeerConnection();
>>>>>=20
>>>>> GetUserMedia(..., function(stream) {
>>>>>       stream1 =3D stream;
>>>> As per my understanding of the gUM spec, this line:
>>>>=20
>>>>>       stream2 =3D new MediaStream(stream1.audioTracks, =
stream1.videoTracks);
>>>> indicates that the audio and video tracks are *copied* into =
stream2, and
>>>> are not references. Consider, for instance, that if a resolution =
change
>>>> was made to a track in stream2, the track from the same source in
>>>> stream1 would remain unchanged.
>>> This is a bit unclear to me. What would a resolution change really =
mean in this case? To me the resolution (when a camera is used as =
source) would be the resolution delivered by the camera. Can it deliver =
two (or more) resolutions at the same time (perhaps also with different =
frame rates, aspect rations and so on)?
>>>=20
>>> To me it makes most sense (if the application is going to explicitly =
ask for specific resolution, frame rate, aspec ration) to ask for it at =
getUserMedia time. If the app does not ask for this explicitly, at least =
the suitable resolution would be known by the browser whenever the video =
is played in a video element (from the size of it).
>>>=20
>> I'm not sure but it looks like implementations are assuming that if =
getusermedia asks fro the same camera twice, the second time it will be =
busy. I was executing the second stream to be cloned from the first one.
>=20
> Right, and that's what's described up higher (new MediaStream()). I =
don't think getUserMedia() is *obligated* to return busy, but I would =
expect it to.
>=20
>>=20
>> So lets say you had a 4k by 4k camera and wanted an HD stream for =
Alice and SD for bob. You would get the first stream with constrains set =
to HD, then clone that to second stream with constraints set to SD.
>=20
> Ah, but here we disagree - there is no "clone with constraints", and =
even if there were you're implying a set of processing occur "somewhere" =
to make the cloned MediaStream.
>=20
> MediaStream Processing could handle this (though I'd want to use a =
pre-defined scaler node, not do it in raw JS for perf and hardware-use =
reasons).
>=20
> Cloning a 4k MediaStream and feeding them to the same (or different) =
PeerConnections would allow the PeerConnections to scale the video for =
encoding and transport.
>=20
>=20

Sure - that sounds like reasonable approach to me.=20



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On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>
> One again, I think we have complete failure on the meaning of the word se=
ssion :-)
>
> I think what EKR was getting at is that if A call B in one phone call, th=
en day later A wants to make an anonymous call to B, B should not be able t=
o tell the second call is coming from same devices. I think that was part o=
f the goal of 6222 but from EKR's email it looks like it fails to provide t=
hat.


Exactly.

-Ekr

From g.kiranreddy4u@gmail.com  Thu Jun 21 06:27:02 2012
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From: kiran kumar <g.kiranreddy4u@gmail.com>
Date: Thu, 21 Jun 2012 18:56:40 +0530
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Subject: Re: [rtcweb] PRANSWER and 3264
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--14dae9340f3978eebb04c2fb783c
Content-Type: text/plain; charset=ISO-8859-1

>>In my opinion, a bigger problem is if the browser, after it has received
a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In the case of
SIP interworking, we can't >>simply map those to sequential different SDP
answers - at least not with using the same To tag value.


>>Yes, how to handle things in that direction must also be specified.


>Isn't that up to the application, which presumably knows it's talking to a
SIP gateway, and can just decide to "not do that", just as it decides to
take no action on ICE candidate >callbacks, since SIP doesn't support
trickle candidates?
>          Harald
Hi,

In cases where the signalling uses the existing SIP signalling nodes,
Application may not know about this.
Also application, if uses JSON for example, should be independent of the
signalling protocols (SIP/JINGLE) that arises after the gateway.
I am thinking of to write a draft for JSON-SIP mapping gateway by modifying
ROAP-SIP mapping gateway draft ( Since ROAP may not be used in future),
where the solution for this problem is very much useful.

But if the application is using XMPP/JINGLE and supposed to pass through a
XMPP-SIP gateway, then the solution is more critical.

Thanks,
Kiran.

--14dae9340f3978eebb04c2fb783c
Content-Type: text/html; charset=ISO-8859-1
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<div>=A0</div>
<div>=A0</div>
<p>&gt;&gt;In my opinion, a bigger problem is if the browser, after it has =
received a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In the c=
ase of SIP interworking, we can&#39;t &gt;&gt;simply map those to sequentia=
l different SDP answers - at least not with using the same To tag value.</p=
>


<p><br>&gt;&gt;Yes, how to handle things in that direction must also be spe=
cified.</p>
<p><br>&gt;Isn&#39;t that up to the application, which presumably knows it&=
#39;s talking to a SIP gateway, and can just decide to &quot;not do that&qu=
ot;, just as it decides to take no action on ICE candidate &gt;callbacks, s=
ince SIP doesn&#39;t support trickle candidates?</p>


<div>&gt;=A0=A0=A0=A0=A0=A0=A0=A0=A0 Harald</div>
<div>Hi,</div>
<div>=A0</div>
<div>In cases where the signalling uses the existing SIP signalling nodes, =
Application may not know about this. </div>
<div>Also application, if uses JSON for example, should be independent of t=
he signalling protocols (SIP/JINGLE) that arises after the gateway.</div>
<div>I am thinking of to write a draft for JSON-SIP mapping=A0gateway by mo=
difying ROAP-SIP mapping gateway draft ( Since ROAP may not be used in futu=
re), where the solution for this problem is very much useful.</div>
<div>=A0</div>
<div>But if the application is using XMPP/JINGLE and supposed to pass throu=
gh a XMPP-SIP gateway, then the solution is more critical.</div>
<div>=A0</div>
<div>Thanks,</div>
<div>Kiran.</div>

--14dae9340f3978eebb04c2fb783c--

From martin.thomson@gmail.com  Thu Jun 21 11:59:50 2012
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On 20 June 2012 13:58, Dan Wing <dwing@cisco.com> wrote:
> Your view seems valid for a typical home user where just about everyone
> is using 192.168.1.0/24. =C2=A0A typical enterprise IT administrator,
> however, does not want their internal network address assignments and
> topology (host candidates) mapped from the outside. =C2=A0Some IT
> administrators strip SMTP "Received:" lines to prevent that
> disclosure.

I am not saying that _browsers_ shouldn't be able to provide a means
of hiding host (or even server reflexive) candidates from sites and
remote peers.  However, if the browser makes it possible to obtain a
host or server reflexive candidate, then the site will be able to
acquire that information.  At that point, we have to rely on the site
to respect the privacy of the user.

I assume that in the cases you cite, the idea is to prevent _anyone_
from getting the topology information.

In either case, I don't think that there is anything significant that
we can say in an IETF document other to indicate that these features
(browser-fixed-relaying, and site-relaying) are highly desirable.

--Martin

From martin.thomson@gmail.com  Thu Jun 21 12:09:12 2012
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>> I think what EKR was getting at is that if A call B in one phone call, t=
hen day later A wants to make an anonymous call to B, B should not be able =
to tell the second call is coming from same devices. I think that was part =
of the goal of 6222 but from EKR's email it looks like it fails to provide =
that.
>
>
> Exactly.

I agree with the analysis.  The idea that the same browser could be
correlated across two independent sessions bothers me.  Within the one
"session", fine - on the contrary, mandatory.  Outside of that, I'd
like at least an option for anonymity.

From pkyzivat@alum.mit.edu  Thu Jun 21 14:53:43 2012
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Subject: Re: [rtcweb] PRANSWER and 3264
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On 6/21/12 10:46 AM, Harald Alvestrand wrote:
> On 06/21/2012 12:45 AM, Paul Kyzivat wrote:
>>> In my opinion, a bigger problem is if the browser, after it has
>>> received a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In
>>> the case of SIP interworking, we can't simply map those to sequential
>>> different SDP answers - at least not with using the same To tag value.
>>
>> Yes, how to handle things in that direction must also be specified.
>
> Isn't that up to the application, which presumably knows it's talking to
> a SIP gateway, and can just decide to "not do that", just as it decides
> to take no action on ICE candidate callbacks, since SIP doesn't support
> trickle candidates?

Perhaps. I'm still trying to get my head around this.

I suppose, if the application has sip on the other side, it could send 
javascript that doesn't use PRANSWER - that uses only pure 3264 
compliant o/a.

Note that this wasn't my main point. My main point was simply to 
completely define the offer/answer rules for JSEP. AFAICT they aren't 
currently defined, to the level that 3264 does.

I guess again you could say that it doesn't matter - that the javascript 
and the server just have to agree among themselves what is ok. But if 
so, then why use SDP at all?

	Thanks,
	Paul


From csp@csperkins.org  Thu Jun 21 14:54:41 2012
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To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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On 21 Jun 2012, at 20:09, Martin Thomson wrote:
> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> =
wrote:
>>> I think what EKR was getting at is that if A call B in one phone =
call, then day later A wants to make an anonymous call to B, B should =
not be able to tell the second call is coming from same devices. I think =
that was part of the goal of 6222 but from EKR's email it looks like it =
fails to provide that.
>>=20
>> Exactly.
>=20
> I agree with the analysis.  The idea that the same browser could be
> correlated across two independent sessions bothers me.  Within the one
> "session", fine - on the contrary, mandatory.  Outside of that, I'd
> like at least an option for anonymity.

Using SRTP to encrypt the traffic will stop third-parties correlating =
the sessions, and RTCWeb is mandates SRTP and encryption.

RFC 6222 does define per-session RTCP CNAME values, if you're concerned =
about the called party being able to correlate sessions based on the =
RTCP CNAME. We could mandate those instead of short-term persistent RTCP =
CNAME values. If that's not sufficient, then someone will need to write =
a draft that defines a new RTCP CNAME generation algorithm, which this =
draft can refer to.=20

--=20
Colin Perkins
http://csperkins.org/




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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 21 Jun 2012 15:10:20 -0700
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org> wrote:
> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>>>> I think what EKR was getting at is that if A call B in one phone call,=
 then day later A wants to make an anonymous call to B, B should not be abl=
e to tell the second call is coming from same devices. I think that was par=
t of the goal of 6222 but from EKR's email it looks like it fails to provid=
e that.
>>>
>>> Exactly.
>>
>> I agree with the analysis. =A0The idea that the same browser could be
>> correlated across two independent sessions bothers me. =A0Within the one
>> "session", fine - on the contrary, mandatory. =A0Outside of that, I'd
>> like at least an option for anonymity.
>
> Using SRTP to encrypt the traffic will stop third-parties correlating the=
 sessions, and RTCWeb is mandates SRTP and encryption.

I'm concerned about tracking by people who are second parties.

Consider the case where I call you from an anonymous phone at a domestic
violence shelter. I record the CNAME and then call all the DV shelters unti=
l
I determine which one has a CNAME from the same device.

> RFC 6222 does define per-session RTCP CNAME values, if you're concerned a=
bout the called party being able to correlate sessions based on the RTCP CN=
AME. We could mandate those instead of short-term persistent RTCP CNAME val=
ues. If that's not sufficient, then someone will need to write a draft that=
 defines a new RTCP CNAME generation algorithm, which this draft can refer =
to.

The problem is that those per-session CNAME values do not appear to be
unlinkable.
I.e., they are distinct but they are generated from the same
underlying data with
insufficient entropy to prevent someone who knows two CNAMEs from determini=
ng
if they are from the same device.

As far as new algorithm goes, is there some reason "Random" isn't good enog=
uh?

-Ekr

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To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
> On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org> =
wrote:
>> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
>>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
>>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> =
wrote:
>>>>> I think what EKR was getting at is that if A call B in one phone =
call, then day later A wants to make an anonymous call to B, B should =
not be able to tell the second call is coming from same devices. I think =
that was part of the goal of 6222 but from EKR's email it looks like it =
fails to provide that.
>>>>=20
>>>> Exactly.
>>>=20
>>> I agree with the analysis.  The idea that the same browser could be
>>> correlated across two independent sessions bothers me.  Within the =
one
>>> "session", fine - on the contrary, mandatory.  Outside of that, I'd
>>> like at least an option for anonymity.
>>=20
>> Using SRTP to encrypt the traffic will stop third-parties correlating =
the sessions, and RTCWeb is mandates SRTP and encryption.
>=20
> I'm concerned about tracking by people who are second parties.
>=20
> Consider the case where I call you from an anonymous phone at a =
domestic violence shelter. I record the CNAME and then call all the DV =
shelters until I determine which one has a CNAME from the same device.
>=20
>> RFC 6222 does define per-session RTCP CNAME values, if you're =
concerned about the called party being able to correlate sessions based =
on the RTCP CNAME. We could mandate those instead of short-term =
persistent RTCP CNAME values. If that's not sufficient, then someone =
will need to write a draft that defines a new RTCP CNAME generation =
algorithm, which this draft can refer to.
>=20
> The problem is that those per-session CNAME values do not appear to be
> unlinkable. I.e., they are distinct but they are generated from the =
same underlying data with insufficient entropy to prevent someone who =
knows two CNAMEs from determining if they are from the same device.
>=20
> As far as new algorithm goes, is there some reason "Random" isn't good =
enoguh?


There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It might =
be possible to use a random choice, provided the same random value is =
used across all sessions that need to be correlated, and the value is =
chosen to be unique with high probability. I haven't thought about it in =
detail. In any case, it's an update to RFC 3550 and RFC 6222, so someone =
will have to write up the draft defining this.=20

--=20
Colin Perkins
http://csperkins.org/




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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 21 Jun 2012 16:12:27 -0700
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Cc: rtcweb@ietf.org, Colin Perkins <csp@csperkins.org>, avt@ietf.org
Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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This matches my analysis.

-Ekr


On Thu, Jun 21, 2012 at 4:09 PM, Kevin Gross <kevin.gross@avanw.com> wrote:
> It seems like, pure random CNAMEs would have the same or better probabili=
ty
> of uniqueness compared to the hashed values specified in 6222. I don't th=
ink
> it is difficult to prove this if you assume the hash function does what i=
t's
> designed to and makes data white. 48 is not enough bits to ensure uniquen=
ess
> of a large population of random numbers. If I remember correctly, 64 bits=
 is
> where the statistics start to get comfortable. See
> http://en.wikipedia.org/wiki/Birthday_problem. Potentially there's a bigg=
er
> problem here than linkability :(
>
> Kevin Gross
>
> On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <csp@csperkins.org> wrote:
>>
>> On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
>> > On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org>
>> > wrote:
>> >> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
>> >>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
>> >>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca>
>> >>>> wrote:
>> >>>>> I think what EKR was getting at is that if A call B in one phone
>> >>>>> call, then day later A wants to make an anonymous call to B, B sho=
uld not be
>> >>>>> able to tell the second call is coming from same devices. I think =
that was
>> >>>>> part of the goal of 6222 but from EKR's email it looks like it fai=
ls to
>> >>>>> provide that.
>> >>>>
>> >>>> Exactly.
>> >>>
>> >>> I agree with the analysis. =A0The idea that the same browser could b=
e
>> >>> correlated across two independent sessions bothers me. =A0Within the=
 one
>> >>> "session", fine - on the contrary, mandatory. =A0Outside of that, I'=
d
>> >>> like at least an option for anonymity.
>> >>
>> >> Using SRTP to encrypt the traffic will stop third-parties correlating
>> >> the sessions, and RTCWeb is mandates SRTP and encryption.
>> >
>> > I'm concerned about tracking by people who are second parties.
>> >
>> > Consider the case where I call you from an anonymous phone at a domest=
ic
>> > violence shelter. I record the CNAME and then call all the DV shelters=
 until
>> > I determine which one has a CNAME from the same device.
>> >
>> >> RFC 6222 does define per-session RTCP CNAME values, if you're concern=
ed
>> >> about the called party being able to correlate sessions based on the =
RTCP
>> >> CNAME. We could mandate those instead of short-term persistent RTCP C=
NAME
>> >> values. If that's not sufficient, then someone will need to write a d=
raft
>> >> that defines a new RTCP CNAME generation algorithm, which this draft =
can
>> >> refer to.
>> >
>> > The problem is that those per-session CNAME values do not appear to be
>> > unlinkable. I.e., they are distinct but they are generated from the sa=
me
>> > underlying data with insufficient entropy to prevent someone who knows=
 two
>> > CNAMEs from determining if they are from the same device.
>> >
>> > As far as new algorithm goes, is there some reason "Random" isn't good
>> > enoguh?
>>
>>
>> There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It might
>> be possible to use a random choice, provided the same random value is us=
ed
>> across all sessions that need to be correlated, and the value is chosen =
to
>> be unique with high probability. I haven't thought about it in detail. I=
n
>> any case, it's an update to RFC 3550 and RFC 6222, so someone will have =
to
>> write up the draft defining this.
>>
>> --
>> Colin Perkins
>> http://csperkins.org/
>>
>>
>>
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>
>

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From: Eric Rescorla <ekr@rtfm.com>
Date: Thu, 21 Jun 2012 18:02:27 -0700
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On Thu, Jun 21, 2012 at 5:46 PM, Ali C. Begen (abegen) <abegen@cisco.com> wrote:
> Many implementors just assume "0" for any random number they need to come up with. So, saying to choose a random number may not cut it.

I would require it to be cryptographically random. That's not an issue
since this system needs such numbers all over the place.

From juberti@google.com  Thu Jun 21 18:49:20 2012
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From: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
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--485b397dcd392d72b004c305d73d
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On Thu, Jun 21, 2012 at 9:02 PM, Eric Rescorla <ekr@rtfm.com> wrote:

> On Thu, Jun 21, 2012 at 5:46 PM, Ali C. Begen (abegen) <abegen@cisco.com>
> wrote:
> > Many implementors just assume "0" for any random number they need to
> come up with. So, saying to choose a random number may not cut it.
>
> I would require it to be cryptographically random. That's not an issue
> since this system needs such numbers all over the place.
>
>
Right. In our implementation, we use random data for CNAMEs; it's simpler
than performing the RFC 6222 steps and indistinguishable on the wire.

--485b397dcd392d72b004c305d73d
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<div style="font-family:arial,helvetica,sans-serif"><font><br><br><div class="gmail_quote">On Thu, Jun 21, 2012 at 9:02 PM, Eric Rescorla <span dir="ltr">&lt;<a href="mailto:ekr@rtfm.com" target="_blank">ekr@rtfm.com</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">On Thu, Jun 21, 2012 at 5:46 PM, Ali C. Begen (abegen) &lt;<a href="mailto:abegen@cisco.com">abegen@cisco.com</a>&gt; wrote:<br>


&gt; Many implementors just assume &quot;0&quot; for any random number they need to come up with. So, saying to choose a random number may not cut it.<br>
<br>
I would require it to be cryptographically random. That&#39;s not an issue<br>
since this system needs such numbers all over the place.<br>
<div class="HOEnZb"><div class="h5"><br></div></div></blockquote><div><br></div><div>Right. In our implementation, we use random data for CNAMEs; it&#39;s simpler than performing the RFC 6222 steps and indistinguishable on the wire.</div>

</div></font></div>

--485b397dcd392d72b004c305d73d--

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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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On 6/21/2012 7:12 PM, Eric Rescorla wrote:
> This matches my analysis.

Mine also, and I agree a 64-bit cryptographically random number is safer 
against collision than the RFC 6222 value, unless you *want* to be able 
to reverse the operation to find the inputs.  I agree with Justin that 
there's no reason we can see to implement RFC6222 instead of using a 
64-bit random number per above, and plenty of reasons to use the 64-bit 
number.

> On Thu, Jun 21, 2012 at 4:09 PM, Kevin Gross <kevin.gross@avanw.com> wrote:
>> It seems like, pure random CNAMEs would have the same or better probability
>> of uniqueness compared to the hashed values specified in 6222. I don't think
>> it is difficult to prove this if you assume the hash function does what it's
>> designed to and makes data white. 48 is not enough bits to ensure uniqueness
>> of a large population of random numbers. If I remember correctly, 64 bits is
>> where the statistics start to get comfortable. See
>> http://en.wikipedia.org/wiki/Birthday_problem. Potentially there's a bigger
>> problem here than linkability :(
>>
>> Kevin Gross
>>
>> On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <csp@csperkins.org> wrote:
>>>
>>> On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
>>>> On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org>
>>>> wrote:
>>>>> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
>>>>>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
>>>>>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca>
>>>>>>> wrote:
>>>>>>>> I think what EKR was getting at is that if A call B in one phone
>>>>>>>> call, then day later A wants to make an anonymous call to B, B should not be
>>>>>>>> able to tell the second call is coming from same devices. I think that was
>>>>>>>> part of the goal of 6222 but from EKR's email it looks like it fails to
>>>>>>>> provide that.
>>>>>>>
>>>>>>> Exactly.
>>>>>>
>>>>>> I agree with the analysis.  The idea that the same browser could be
>>>>>> correlated across two independent sessions bothers me.  Within the one
>>>>>> "session", fine - on the contrary, mandatory.  Outside of that, I'd
>>>>>> like at least an option for anonymity.
>>>>>
>>>>> Using SRTP to encrypt the traffic will stop third-parties correlating
>>>>> the sessions, and RTCWeb is mandates SRTP and encryption.
>>>>
>>>> I'm concerned about tracking by people who are second parties.
>>>>
>>>> Consider the case where I call you from an anonymous phone at a domestic
>>>> violence shelter. I record the CNAME and then call all the DV shelters until
>>>> I determine which one has a CNAME from the same device.
>>>>
>>>>> RFC 6222 does define per-session RTCP CNAME values, if you're concerned
>>>>> about the called party being able to correlate sessions based on the RTCP
>>>>> CNAME. We could mandate those instead of short-term persistent RTCP CNAME
>>>>> values. If that's not sufficient, then someone will need to write a draft
>>>>> that defines a new RTCP CNAME generation algorithm, which this draft can
>>>>> refer to.
>>>>
>>>> The problem is that those per-session CNAME values do not appear to be
>>>> unlinkable. I.e., they are distinct but they are generated from the same
>>>> underlying data with insufficient entropy to prevent someone who knows two
>>>> CNAMEs from determining if they are from the same device.
>>>>
>>>> As far as new algorithm goes, is there some reason "Random" isn't good
>>>> enoguh?
>>>
>>>
>>> There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It might
>>> be possible to use a random choice, provided the same random value is used
>>> across all sessions that need to be correlated, and the value is chosen to
>>> be unique with high probability. I haven't thought about it in detail. In
>>> any case, it's an update to RFC 3550 and RFC 6222, so someone will have to
>>> write up the draft defining this.
>>>
>>> --
>>> Colin Perkins
>>> http://csperkins.org/


-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Thu Jun 21 20:11:09 2012
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On 6/21/2012 2:59 PM, Martin Thomson wrote:
> On 20 June 2012 13:58, Dan Wing <dwing@cisco.com> wrote:
>> Your view seems valid for a typical home user where just about everyone
>> is using 192.168.1.0/24.  A typical enterprise IT administrator,
>> however, does not want their internal network address assignments and
>> topology (host candidates) mapped from the outside.  Some IT
>> administrators strip SMTP "Received:" lines to prevent that
>> disclosure.
>
> I am not saying that _browsers_ shouldn't be able to provide a means
> of hiding host (or even server reflexive) candidates from sites and
> remote peers.  However, if the browser makes it possible to obtain a
> host or server reflexive candidate, then the site will be able to
> acquire that information.  At that point, we have to rely on the site
> to respect the privacy of the user.
>
> I assume that in the cases you cite, the idea is to prevent _anyone_
> from getting the topology information.
>
> In either case, I don't think that there is anything significant that
> we can say in an IETF document other to indicate that these features
> (browser-fixed-relaying, and site-relaying) are highly desirable.

This seems reasonable; browsers could implement configuration parameters 
that restrict this information from WebRTC applications or force use of 
a fixed relay server.  (Something not unheard of in the SIP world either.)

-- 
Randell Jesup
randell-ietf@jesup.org


From mary.ietf.barnes@gmail.com  Fri Jun 22 12:40:17 2012
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Subject: [rtcweb] "Telepresence Tutorial" @ IETF-84 (Please RSVP)
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Hi all,

We are planning a lunchtime tutorial on Monday, July 30th. =A0The
primary objective is to provide more background on telepresence as
well as an update of the work in CLUE to the broader RAI and IETF
community. =A0This will hopefully facilitate the process of completing
the CLUE protocol work as it gets broader community review.

Here's the proposed outline:
1) Introduction to telepresence
2) Example telepresence scenarios
3) Introduction to the CLUE work
4) High level introduction of framework
5) One or two use cases as examples of how the CLUE framework is
realized - showing how existing protocols (e.g., SDP, RTP) are
integral to the solution along
with additional signaling for CLUE.

We have reserved a room for around 50 and attendance will be FCFS, so
please RSVP ASAP:
http://www.doodle.com/mzu3nmdeehkxkiy

The deadline to reply is Friday, July 13th, 2012 (5pm Pacific), as we
are trying to find a sponsor for lunch (i.e., box lunches). Note, if
anyone is willing to sponsor all or even part of the lunch, please let
me know ASAP. =A0Also, if you have any dietary restrictions (e.g.,
vegetarian, etc.), please let me know offline or include a comment in
the doodle.
http://www.doodle.com/mzu3nmdeehkxkiys

Regards,
Mary
CLUE WG co-chair

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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Randell Jesup
> Sent: Thursday, June 21, 2012 8:02 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
> 
> On 6/21/2012 7:12 PM, Eric Rescorla wrote:
> > This matches my analysis.
> 
> Mine also, and I agree a 64-bit cryptographically random number is
> safer
> against collision than the RFC 6222 value, unless you *want* to be able
> to reverse the operation to find the inputs.  I agree with Justin that
> there's no reason we can see to implement RFC6222 instead of using a
> 64-bit random number per above, and plenty of reasons to use the 64-bit
> number.

Yeah, that's what I wrote when I became a co-author of
draft-begen-avt-rtp-cnames-02, although was 96 bits (instead of 64),
http://tools.ietf.org/rfcdiff?url2=draft-begen-avt-rtp-cnames-02.txt and
concatenated some other cruft beyond just an RFC4086 random number,

  o  To generate a unique CNAME for each RTP session, an endpoint MAY
      perform SHA1-HMAC [RFC2104] on the concatenated values of the RTP
      endpoint's initial SSRC, the source and destination IP addresses
      and ports, and a randomly-generated value [RFC4086], and then
      truncate the 160-bit output to 96 bits and finally convert the 96
      bits to ASCII using Base64 encoding [RFC4648].  This results in a
      128-bit printable CNAME.  Note that the CNAME MUST NOT change if
      an SSRC collision occurs, hence only the initial SSRC value chosen
      by the endpoint is used.

That text evolved as the document progressed, and veered farther away
from being "just a big random number".  I don't recall why it evolved.

-d




> > On Thu, Jun 21, 2012 at 4:09 PM, Kevin Gross <kevin.gross@avanw.com>
> wrote:
> >> It seems like, pure random CNAMEs would have the same or better
> probability
> >> of uniqueness compared to the hashed values specified in 6222. I
> don't think
> >> it is difficult to prove this if you assume the hash function does
> what it's
> >> designed to and makes data white. 48 is not enough bits to ensure
> uniqueness
> >> of a large population of random numbers. If I remember correctly, 64
> bits is
> >> where the statistics start to get comfortable. See
> >> http://en.wikipedia.org/wiki/Birthday_problem. Potentially there's a
> bigger
> >> problem here than linkability :(
> >>
> >> Kevin Gross
> >>
> >> On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <csp@csperkins.org>
> wrote:
> >>>
> >>> On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
> >>>> On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org>
> >>>> wrote:
> >>>>> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
> >>>>>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
> >>>>>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings
> <fluffy@iii.ca>
> >>>>>>> wrote:
> >>>>>>>> I think what EKR was getting at is that if A call B in one
> phone
> >>>>>>>> call, then day later A wants to make an anonymous call to B, B
> should not be
> >>>>>>>> able to tell the second call is coming from same devices. I
> think that was
> >>>>>>>> part of the goal of 6222 but from EKR's email it looks like it
> fails to
> >>>>>>>> provide that.
> >>>>>>>
> >>>>>>> Exactly.
> >>>>>>
> >>>>>> I agree with the analysis.  The idea that the same browser could
> be
> >>>>>> correlated across two independent sessions bothers me.  Within
> the one
> >>>>>> "session", fine - on the contrary, mandatory.  Outside of that,
> I'd
> >>>>>> like at least an option for anonymity.
> >>>>>
> >>>>> Using SRTP to encrypt the traffic will stop third-parties
> correlating
> >>>>> the sessions, and RTCWeb is mandates SRTP and encryption.
> >>>>
> >>>> I'm concerned about tracking by people who are second parties.
> >>>>
> >>>> Consider the case where I call you from an anonymous phone at a
> domestic
> >>>> violence shelter. I record the CNAME and then call all the DV
> shelters until
> >>>> I determine which one has a CNAME from the same device.
> >>>>
> >>>>> RFC 6222 does define per-session RTCP CNAME values, if you're
> concerned
> >>>>> about the called party being able to correlate sessions based on
> the RTCP
> >>>>> CNAME. We could mandate those instead of short-term persistent
> RTCP CNAME
> >>>>> values. If that's not sufficient, then someone will need to write
> a draft
> >>>>> that defines a new RTCP CNAME generation algorithm, which this
> draft can
> >>>>> refer to.
> >>>>
> >>>> The problem is that those per-session CNAME values do not appear
> to be
> >>>> unlinkable. I.e., they are distinct but they are generated from
> the same
> >>>> underlying data with insufficient entropy to prevent someone who
> knows two
> >>>> CNAMEs from determining if they are from the same device.
> >>>>
> >>>> As far as new algorithm goes, is there some reason "Random" isn't
> good
> >>>> enoguh?
> >>>
> >>>
> >>> There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It
> might
> >>> be possible to use a random choice, provided the same random value
> is used
> >>> across all sessions that need to be correlated, and the value is
> chosen to
> >>> be unique with high probability. I haven't thought about it in
> detail. In
> >>> any case, it's an update to RFC 3550 and RFC 6222, so someone will
> have to
> >>> write up the draft defining this.
> >>>
> >>> --
> >>> Colin Perkins
> >>> http://csperkins.org/
> 
> 
> --
> Randell Jesup
> randell-ietf@jesup.org
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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It seems to me that we have two issues here, and it might be good to 
separate them:

- I believe RTCWEB implementations should accept a very large set of 
CNAMEs - essentially anything that isn't hopelessly long or has 
"bizarre" characters in them.

- I believe RTCWEB implementations should be allowed to generate CNAMEs 
that satisfy EKR's requirements for unlinkability.

Those two are probably worthy of separate text, and perhaps even 
separate documents.

                 Harald


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On Sat, 2012-06-23 at 15:56 +0200, Harald Alvestrand wrote:

> It seems to me that we have two issues here, and it might be good to 
> separate them:
> 
> - I believe RTCWEB implementations should accept a very large set of 
> CNAMEs - essentially anything that isn't hopelessly long or has 
> "bizarre" characters in them.
> 
> - I believe RTCWEB implementations should be allowed to generate CNAMEs 
> that satisfy EKR's requirements for unlinkability.


s/allowed/strongly encouraged/ above & I agree.


> 
> Those two are probably worthy of separate text, and perhaps even 
> separate documents.
> 
>                  Harald
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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On Sat, 2012-06-23 at 15:56 +0200, Harald Alvestrand wrote:
<BLOCKQUOTE TYPE=CITE>
<PRE>
It seems to me that we have two issues here, and it might be good to 
separate them:

- I believe RTCWEB implementations should accept a very large set of 
CNAMEs - essentially anything that isn't hopelessly long or has 
&quot;bizarre&quot; characters in them.

- I believe RTCWEB implementations should be allowed to generate CNAMEs 
that satisfy EKR's requirements for unlinkability.
</PRE>
</BLOCKQUOTE>
<BR>
s/allowed/strongly encouraged/ above &amp; I agree.<BR>
<BR>
<BLOCKQUOTE TYPE=CITE>
<PRE>

Those two are probably worthy of separate text, and perhaps even 
separate documents.

                 Harald

_______________________________________________
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<A HREF="mailto:rtcweb@ietf.org">rtcweb@ietf.org</A>
<A HREF="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</A>
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From kevin.gross@avanw.com  Thu Jun 21 16:09:08 2012
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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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It seems like, pure random CNAMEs would have the same or better probability
of uniqueness compared to the hashed values specified in 6222. I don't
think it is difficult to prove this if you assume the hash function does
what it's designed to and makes data white. 48 is not enough bits to ensure
uniqueness of a large population of random numbers. If I remember
correctly, 64 bits is where the statistics start to get comfortable. See
http://en.wikipedia.org/wiki/Birthday_problem. Potentially there's a bigger
problem here than linkability :(

Kevin Gross

On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <csp@csperkins.org> wrote:

> On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
> > On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org>
> wrote:
> >> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
> >>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
> >>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca>
> wrote:
> >>>>> I think what EKR was getting at is that if A call B in one phone
> call, then day later A wants to make an anonymous call to B, B should not
> be able to tell the second call is coming from same devices. I think that
> was part of the goal of 6222 but from EKR's email it looks like it fails to
> provide that.
> >>>>
> >>>> Exactly.
> >>>
> >>> I agree with the analysis.  The idea that the same browser could be
> >>> correlated across two independent sessions bothers me.  Within the one
> >>> "session", fine - on the contrary, mandatory.  Outside of that, I'd
> >>> like at least an option for anonymity.
> >>
> >> Using SRTP to encrypt the traffic will stop third-parties correlating
> the sessions, and RTCWeb is mandates SRTP and encryption.
> >
> > I'm concerned about tracking by people who are second parties.
> >
> > Consider the case where I call you from an anonymous phone at a domestic
> violence shelter. I record the CNAME and then call all the DV shelters
> until I determine which one has a CNAME from the same device.
> >
> >> RFC 6222 does define per-session RTCP CNAME values, if you're concerned
> about the called party being able to correlate sessions based on the RTCP
> CNAME. We could mandate those instead of short-term persistent RTCP CNAME
> values. If that's not sufficient, then someone will need to write a draft
> that defines a new RTCP CNAME generation algorithm, which this draft can
> refer to.
> >
> > The problem is that those per-session CNAME values do not appear to be
> > unlinkable. I.e., they are distinct but they are generated from the same
> underlying data with insufficient entropy to prevent someone who knows two
> CNAMEs from determining if they are from the same device.
> >
> > As far as new algorithm goes, is there some reason "Random" isn't good
> enoguh?
>
>
> There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It might
> be possible to use a random choice, provided the same random value is used
> across all sessions that need to be correlated, and the value is chosen to
> be unique with high probability. I haven't thought about it in detail. In
> any case, it's an update to RFC 3550 and RFC 6222, so someone will have to
> write up the draft defining this.
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>

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It seems like, pure random CNAMEs would have the same or better probability=
 of uniqueness compared to the hashed values specified in 6222. I don&#39;t=
 think it is difficult to prove this if you assume the hash function does w=
hat it&#39;s designed to and makes data white. 48 is not enough bits to ens=
ure uniqueness of a large population of random numbers. If I remember corre=
ctly, 64 bits is where the statistics start to get comfortable. See=A0
<a href=3D"http://en.wikipedia.org/wiki/Birthday_problem">http://en.wikiped=
ia.org/wiki/Birthday_problem</a>. Potentially there&#39;s a bigger problem =
here than linkability :(<div><br clear=3D"all">Kevin Gross<br><div><br></di=
v>
<div class=3D"gmail_quote">On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <=
span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperkins.org" target=3D"_blank"=
>csp@csperkins.org</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">On 21 Jun 2012, at 23:10, Eric Resc=
orla wrote:<br>
&gt; On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins &lt;<a href=3D"mailto:c=
sp@csperkins.org">csp@csperkins.org</a>&gt; wrote:<br>
&gt;&gt; On 21 Jun 2012, at 20:09, Martin Thomson wrote:<br>
&gt;&gt;&gt; On 21 June 2012 05:48, Eric Rescorla &lt;<a href=3D"mailto:ekr=
@rtfm.com">ekr@rtfm.com</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt; On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings &lt;<a hr=
ef=3D"mailto:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;&gt;&gt;&gt;&gt; I think what EKR was getting at is that if A call B in=
 one phone call, then day later A wants to make an anonymous call to B, B s=
hould not be able to tell the second call is coming from same devices. I th=
ink that was part of the goal of 6222 but from EKR&#39;s email it looks lik=
e it fails to provide that.<br>

&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Exactly.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; I agree with the analysis. =A0The idea that the same browser c=
ould be<br>
&gt;&gt;&gt; correlated across two independent sessions bothers me. =A0With=
in the one<br>
&gt;&gt;&gt; &quot;session&quot;, fine - on the contrary, mandatory. =A0Out=
side of that, I&#39;d<br>
&gt;&gt;&gt; like at least an option for anonymity.<br>
&gt;&gt;<br>
&gt;&gt; Using SRTP to encrypt the traffic will stop third-parties correlat=
ing the sessions, and RTCWeb is mandates SRTP and encryption.<br>
&gt;<br>
&gt; I&#39;m concerned about tracking by people who are second parties.<br>
&gt;<br>
&gt; Consider the case where I call you from an anonymous phone at a domest=
ic violence shelter. I record the CNAME and then call all the DV shelters u=
ntil I determine which one has a CNAME from the same device.<br>
&gt;<br>
&gt;&gt; RFC 6222 does define per-session RTCP CNAME values, if you&#39;re =
concerned about the called party being able to correlate sessions based on =
the RTCP CNAME. We could mandate those instead of short-term persistent RTC=
P CNAME values. If that&#39;s not sufficient, then someone will need to wri=
te a draft that defines a new RTCP CNAME generation algorithm, which this d=
raft can refer to.<br>

&gt;<br>
&gt; The problem is that those per-session CNAME values do not appear to be=
<br>
&gt; unlinkable. I.e., they are distinct but they are generated from the sa=
me underlying data with insufficient entropy to prevent someone who knows t=
wo CNAMEs from determining if they are from the same device.<br>
&gt;<br>
&gt; As far as new algorithm goes, is there some reason &quot;Random&quot; =
isn&#39;t good enoguh?<br>
<br>
<br>
</div></div>There&#39;s some discussion in RFC 3550 Section 6.5.1 and RFC 6=
222. It might be possible to use a random choice, provided the same random =
value is used across all sessions that need to be correlated, and the value=
 is chosen to be unique with high probability. I haven&#39;t thought about =
it in detail. In any case, it&#39;s an update to RFC 3550 and RFC 6222, so =
someone will have to write up the draft defining this.<br>

<div class=3D"im HOEnZb"><br>
--<br>
Colin Perkins<br>
<a href=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</=
a><br>
<br>
<br>
<br>
</div><div class=3D"HOEnZb"><div class=3D"h5">_____________________________=
__________________<br>
Audio/Video Transport Core Maintenance<br>
<a href=3D"mailto:avt@ietf.org">avt@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/avt" target=3D"_blank">htt=
ps://www.ietf.org/mailman/listinfo/avt</a><br>
</div></div></blockquote></div><br></div>

--0016e6de17f3184a0604c3039ad2--

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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Kevin Gross <kevin.gross@avanw.com>, Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] [rtcweb] Randomly-generated CNAMEs
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "avt@ietf.org" <avt@ietf.org>
Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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Many implementors just assume "0" for any random number they need to come u=
p with. So, saying to choose a random number may not cut it.

-acbegen

> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf Of Kev=
in Gross
> Sent: Thursday, June 21, 2012 7:09 PM
> To: Colin Perkins
> Cc: Martin Thomson; rtcweb@ietf.org; avt@ietf.org; Cullen Jennings
> Subject: Re: [AVTCORE] [rtcweb] Randomly-generated CNAMEs
>=20
> It seems like, pure random CNAMEs would have the same or better probabili=
ty of uniqueness compared to the hashed values
> specified in 6222. I don't think it is difficult to prove this if you ass=
ume the hash function does what it's designed to and makes
> data white. 48 is not enough bits to ensure uniqueness of a large populat=
ion of random numbers. If I remember correctly, 64
> bits is where the statistics start to get comfortable. See  http://en.wik=
ipedia.org/wiki/Birthday_problem. Potentially there's a
> bigger problem here than linkability :(
>=20
> Kevin Gross
>=20
>=20
> On Thu, Jun 21, 2012 at 4:28 PM, Colin Perkins <csp@csperkins.org> wrote:
>=20
>=20
> 	On 21 Jun 2012, at 23:10, Eric Rescorla wrote:
> 	> On Thu, Jun 21, 2012 at 2:54 PM, Colin Perkins <csp@csperkins.org> wro=
te:
> 	>> On 21 Jun 2012, at 20:09, Martin Thomson wrote:
> 	>>> On 21 June 2012 05:48, Eric Rescorla <ekr@rtfm.com> wrote:
> 	>>>> On Thu, Jun 21, 2012 at 5:39 AM, Cullen Jennings <fluffy@iii.ca> wr=
ote:
> 	>>>>> I think what EKR was getting at is that if A call B in one phone c=
all, then day later A wants to make an
> anonymous call to B, B should not be able to tell the second call is comi=
ng from same devices. I think that was part of the
> goal of 6222 but from EKR's email it looks like it fails to provide that.
> 	>>>>
> 	>>>> Exactly.
> 	>>>
> 	>>> I agree with the analysis.  The idea that the same browser could be
> 	>>> correlated across two independent sessions bothers me.  Within the o=
ne
> 	>>> "session", fine - on the contrary, mandatory.  Outside of that, I'd
> 	>>> like at least an option for anonymity.
> 	>>
> 	>> Using SRTP to encrypt the traffic will stop third-parties correlating=
 the sessions, and RTCWeb is mandates SRTP
> and encryption.
> 	>
> 	> I'm concerned about tracking by people who are second parties.
> 	>
> 	> Consider the case where I call you from an anonymous phone at a domest=
ic violence shelter. I record the CNAME
> and then call all the DV shelters until I determine which one has a CNAME=
 from the same device.
> 	>
> 	>> RFC 6222 does define per-session RTCP CNAME values, if you're concern=
ed about the called party being able to
> correlate sessions based on the RTCP CNAME. We could mandate those instea=
d of short-term persistent RTCP CNAME values.
> If that's not sufficient, then someone will need to write a draft that de=
fines a new RTCP CNAME generation algorithm, which
> this draft can refer to.
> 	>
> 	> The problem is that those per-session CNAME values do not appear to be
> 	> unlinkable. I.e., they are distinct but they are generated from the sa=
me underlying data with insufficient entropy to
> prevent someone who knows two CNAMEs from determining if they are from th=
e same device.
> 	>
> 	> As far as new algorithm goes, is there some reason "Random" isn't good=
 enoguh?
>=20
>=20
>=20
> 	There's some discussion in RFC 3550 Section 6.5.1 and RFC 6222. It might=
 be possible to use a random choice,
> provided the same random value is used across all sessions that need to b=
e correlated, and the value is chosen to be unique
> with high probability. I haven't thought about it in detail. In any case,=
 it's an update to RFC 3550 and RFC 6222, so someone
> will have to write up the draft defining this.
>=20
>=20
> 	--
> 	Colin Perkins
> 	http://csperkins.org/
>=20
>=20
>=20
>=20
> 	_______________________________________________
> 	Audio/Video Transport Core Maintenance
> 	avt@ietf.org
> 	https://www.ietf.org/mailman/listinfo/avt
>=20
>=20


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--f46d04388e638372a004c306eff3
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RFC 4086 - Randomness Requirements for Security

On Thu, Jun 21, 2012 at 7:48 PM, Justin Uberti <juberti@google.com> wrote:

>
>
> On Thu, Jun 21, 2012 at 9:02 PM, Eric Rescorla <ekr@rtfm.com> wrote:
>
>> On Thu, Jun 21, 2012 at 5:46 PM, Ali C. Begen (abegen) <abegen@cisco.com>
>> wrote:
>> > Many implementors just assume "0" for any random number they need to
>> come up with. So, saying to choose a random number may not cut it.
>>
>> I would require it to be cryptographically random. That's not an issue
>> since this system needs such numbers all over the place.
>>
>>
> Right. In our implementation, we use random data for CNAMEs; it's simpler
> than performing the RFC 6222 steps and indistinguishable on the wire.
>

--f46d04388e638372a004c306eff3
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

RFC 4086 -=A0<span style=3D"white-space:pre-wrap">Randomness Requirements f=
or Security</span><br><br><div><div class=3D"gmail_quote">On Thu, Jun 21, 2=
012 at 7:48 PM, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juber=
ti@google.com" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<b=
r>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div style=3D"font-family:arial,helvetica,sa=
ns-serif"><font><br><br><div class=3D"gmail_quote"><div><div class=3D"h5">O=
n Thu, Jun 21, 2012 at 9:02 PM, Eric Rescorla <span dir=3D"ltr">&lt;<a href=
=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span> wrot=
e:<br>


<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">On Thu, Jun 21, 2012 at 5:46 PM, Ali C. Bege=
n (abegen) &lt;<a href=3D"mailto:abegen@cisco.com" target=3D"_blank">abegen=
@cisco.com</a>&gt; wrote:<br>



&gt; Many implementors just assume &quot;0&quot; for any random number they=
 need to come up with. So, saying to choose a random number may not cut it.=
<br>
<br>
I would require it to be cryptographically random. That&#39;s not an issue<=
br>
since this system needs such numbers all over the place.<br>
<div><div><br></div></div></blockquote><div><br></div></div></div><div>Righ=
t. In our implementation, we use random data for CNAMEs; it&#39;s simpler t=
han performing the RFC 6222 steps and indistinguishable on the wire.</div>


</div></font></div>
</blockquote></div><br></div>

--f46d04388e638372a004c306eff3--

From christer.holmberg@ericsson.com  Mon Jun 25 16:53:21 2012
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To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 26 Jun 2012 01:53:18 +0200
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Subject: Re: [rtcweb] PRANSWER and 3264
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Hi,=20

>>> In my opinion, a bigger problem is if the browser, after it has=20
>>> received a JSEP OFFER, starts sending a *PRANSWER/ANSWER sequence. In=20
>>> the case of SIP interworking, we can't simply map those to sequential=20
>>> different SDP answers - at least not with using the same To tag value.
>>
>> Yes, how to handle things in that direction must also be specified.
>
> Isn't that up to the application, which presumably knows it's talking to =
a SIP gateway, and can just decide to "not do that", just as=20
> it decides to take no action on ICE candidate callbacks, since SIP doesn'=
t support trickle candidates?

I am not sure I understand what you mean by "not do that".

If the browser sends a PRANSWER, which the app mapps to a SIP answer, and t=
hen sends another PRANSWER, the app cannot simply ignore it (it of course d=
epends on what information has changed from the previous PRANSWER).

Regards,

Christer


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Subject: Re: [rtcweb] PRANSWER and 3264
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On 25 June 2012 16:53, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
> If the browser sends a PRANSWER, which the app mapps to a SIP answer, and then sends another PRANSWER, the app cannot simply ignore it (it of course depends on what information has changed from the previous PRANSWER).

Does not compute.  The browser doesn't signal anything.  The app
decides what is and is sent.

From christer.holmberg@ericsson.com  Tue Jun 26 01:51:11 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Martin Thomson' <martin.thomson@gmail.com>
Date: Tue, 26 Jun 2012 10:51:08 +0200
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Hi,=20

>> If the browser sends a PRANSWER, which the app mapps to a SIP answer, an=
d then sends another=20
>> PRANSWER, the app cannot simply ignore it (it of course depends on what =
information has changed from the previous PRANSWER).
>
> Does not compute.  The browser doesn't signal anything.  The app decides =
what is and is sent.

True. There is no here-is-a-new-answer callback method - the app has to req=
uest for an answer.

Regards,

Christer

From magnus.westerlund@ericsson.com  Tue Jun 26 02:43:47 2012
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Subject: [rtcweb] Minutes Tuesday 12th of June 2012 Interim meeting
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Hi,

At this link you will find the first version of the minutes for the
interim meeting on the 12th of June.

http://www.ietf.org/proceedings/interim/2012/06/12/rtcweb/minutes/minutes-interim-2012-rtcweb-2

Please provide feedback

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Subject: Re: [rtcweb] Minutes Tuesday 12th of June 2012 Interim meeting
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WG,

Minor update based on Mary's notes.

I have identified one potential disagreement in the notes. It concerns
the following part:

Semantic Loss Tolerance (1) (Slide 23)
Keith Drage suggested that PLI could be a MAY and that we simply
describe the implications. Stephan Wenger thinks this is important and
should be REQUIRED to minimize interop failures. Colin commented that
support of this is rather trivial if you support FIR. Stephan Wenger and
Randal Jesup made it clear that how you react is up to the
implementation, it doesnt have the same semantics as FIR.

Consensus to Require Support.

My uncertainty is if this is Require or Recommend as in the slide.

Cheers

Magnus


On 2012-06-26 11:43, Magnus Westerlund wrote:
> Hi,
> 
> At this link you will find the first version of the minutes for the
> interim meeting on the 12th of June.
> 
> http://www.ietf.org/proceedings/interim/2012/06/12/rtcweb/minutes/minutes-interim-2012-rtcweb-2
> 
> Please provide feedback
> 
> Cheers
> 
> Magnus Westerlund
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

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




From randell-ietf@jesup.org  Tue Jun 26 07:00:16 2012
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On 6/26/2012 9:17 AM, Magnus Westerlund wrote:
> WG,
>
> Minor update based on Mary's notes.
>
> I have identified one potential disagreement in the notes. It concerns
> the following part:
>
> Semantic Loss Tolerance (1) (Slide 23)
> Keith Drage suggested that PLI could be a MAY and that we simply
> describe the implications. Stephan Wenger thinks this is important and
> should be REQUIRED to minimize interop failures. Colin commented that
> support of this is rather trivial if you support FIR. Stephan Wenger and
> Randal Jesup made it clear that how you react is up to the
> implementation, it doesnt have the same semantics as FIR.
>
> Consensus to Require Support.
>
> My uncertainty is if this is Require or Recommend as in the slide.

I took it (IIRC) as consensus to Require support, but I can't say that 
is 100% certain.


-- 
Randell Jesup
randell-ietf@jesup.org


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On 2012-06-26 15:59, Randell Jesup wrote:
> On 6/26/2012 9:17 AM, Magnus Westerlund wrote:
>> WG,
>>
>> Minor update based on Mary's notes.
>>
>> I have identified one potential disagreement in the notes. It concerns
>> the following part:
>>
>> Semantic Loss Tolerance (1) (Slide 23)
>> Keith Drage suggested that PLI could be a MAY and that we simply
>> describe the implications. Stephan Wenger thinks this is important and
>> should be REQUIRED to minimize interop failures. Colin commented that
>> support of this is rather trivial if you support FIR. Stephan Wenger and
>> Randal Jesup made it clear that how you react is up to the
>> implementation, it doesnt have the same semantics as FIR.
>>
>> Consensus to Require Support.
>>
>> My uncertainty is if this is Require or Recommend as in the slide.
>
> I took it (IIRC) as consensus to Require support, but I can't say that
> is 100% certain.
I have the same recollection (but am not certain).
>
>



From eckelcu@cisco.com  Tue Jun 26 10:51:56 2012
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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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In my notes I have PLI as REQUIRED.
I also have RFC 5761 RTP/RTCP multiplexing as REQUIRED, whereas the notes c=
urrently reads that support for separate ports is REQUIRED. I would appreci=
ate clarification on this point as well.
Was the consensus that RTCWEB implementations are REQUIRED to support RFC 5=
761, meaning support for multiplexing is negotiated and if it fails, using =
separate ports must be supported?

Cheers,
Charles

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stefan Hakansson LK
> Sent: Tuesday, June 26, 2012 7:43 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Minutes Tuesday 12th of June 2012 Interim meeting
>=20
> On 2012-06-26 15:59, Randell Jesup wrote:
> > On 6/26/2012 9:17 AM, Magnus Westerlund wrote:
> >> WG,
> >>
> >> Minor update based on Mary's notes.
> >>
> >> I have identified one potential disagreement in the notes. It concerns
> >> the following part:
> >>
> >> Semantic Loss Tolerance (1) (Slide 23)
> >> Keith Drage suggested that PLI could be a MAY and that we simply
> >> describe the implications. Stephan Wenger thinks this is important and
> >> should be REQUIRED to minimize interop failures. Colin commented that
> >> support of this is rather trivial if you support FIR. Stephan Wenger a=
nd
> >> Randal Jesup made it clear that how you react is up to the
> >> implementation, it doesn't have the same semantics as FIR.
> >>
> >> Consensus to Require Support.
> >>
> >> My uncertainty is if this is Require or Recommend as in the slide.
> >
> > I took it (IIRC) as consensus to Require support, but I can't say that
> > is 100% certain.
> I have the same recollection (but am not certain).
> >
> >
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Minutes Tuesday 12th of June 2012 Interim meeting
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On 26 June 2012 10:50, Charles Eckel (eckelcu) <eckelcu@cisco.com> wrote:
> In my notes I have PLI as REQUIRED.
That was my recollection.

> I also have RFC 5761 RTP/RTCP multiplexing as REQUIRED, whereas the notes currently reads that support for separate ports is REQUIRED. I would appreciate clarification on this point as well.
Both were required.

From ekr@rtfm.com  Tue Jun 26 16:00:20 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 26 Jun 2012 15:59:34 -0700
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Subject: [rtcweb] Why persistent consent for HTTP is a problem
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In Martin's review of draft-ietf-rtcweb-security-arch, he (re)raised
the question of persistent consent for camera/microphone access from
pages served over HTTP.  I had thought this issue closed, but in the
discussion at the interim there seemed to be a fair amount of
sentiment that people hadn't understood the security issues and now
believed that this was a bad idea. I agreed to attempt to re-start the
discussion with a clearer description of what the problem was, hence
this message.

Here's the basic attack:
Say that I have given persistent permission to access my camera and
microphone to any page from http://www.example.com/. Now, I go to an
Internet cafe and surf to *any* HTTP page, e.g.,
http://www.google.com/. At this point, any attacker who controls the
wireless network can redirect that access to point to
http://www.example.com/ and inject JS that reads from my camera and
microphone.

That's pretty bad, but made worse by the following facts:
1. It's not that hard for the attacker to open up
popups/popunders/iframes, etc. that are in the domain of example.com
but actually execute code from the attacker.  So, once you've opened
the browser in a hostile environment, the attacker can bug you at
times of his choosing until you close your browser.

2. If your browser does any sort of auto-refresh (e.g., because
the page refreshes itself) then you can get infected even if you
don't do anything deliberate.

3. This also applies if you are using any kind of open wireless
network, such as if your home network doesn't use a strong
enough WPA key.


So, the executive summary is: if you have a persistent permission
for an HTTP site, and you ever use your browser on any insecure
network, an attacker can bug your computer until you close your
browser. This seems bad.

Does this change people's opinions about what the rules should be
about whether browsers permit persistent HTTP permissions?

-Ekr

From igor.faynberg@alcatel-lucent.com  Tue Jun 26 16:52:16 2012
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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Not only that, but I find it hard to understand why anyone would need to 
give persistent consent for accessing media devices to any device.

I also thought the issue was closed for good.

(Sorry, I could not be present at the intereem meeting, even remotely...)

Igor

On 6/26/2012 6:59 PM, Eric Rescorla wrote:
> In Martin's review of draft-ietf-rtcweb-security-arch, he (re)raised
> the question of persistent consent for camera/microphone access from
> pages served over HTTP.  I had thought this issue closed, but in the
> discussion at the interim there seemed to be a fair amount of
> sentiment that people hadn't understood the security issues and now
> believed that this was a bad idea. I agreed to attempt to re-start the
> discussion with a clearer description of what the problem was, hence
> this message.
>
> Here's the basic attack:
> Say that I have given persistent permission to access my camera and
> microphone to any page from http://www.example.com/. Now, I go to an
> Internet cafe and surf to *any* HTTP page, e.g.,
> http://www.google.com/. At this point, any attacker who controls the
> wireless network can redirect that access to point to
> http://www.example.com/ and inject JS that reads from my camera and
> microphone.
>
> That's pretty bad, but made worse by the following facts:
> 1. It's not that hard for the attacker to open up
> popups/popunders/iframes, etc. that are in the domain of example.com
> but actually execute code from the attacker.  So, once you've opened
> the browser in a hostile environment, the attacker can bug you at
> times of his choosing until you close your browser.
>
> 2. If your browser does any sort of auto-refresh (e.g., because
> the page refreshes itself) then you can get infected even if you
> don't do anything deliberate.
>
> 3. This also applies if you are using any kind of open wireless
> network, such as if your home network doesn't use a strong
> enough WPA key.
>
>
> So, the executive summary is: if you have a persistent permission
> for an HTTP site, and you ever use your browser on any insecure
> network, an attacker can bug your computer until you close your
> browser. This seems bad.
>
> Does this change people's opinions about what the rules should be
> about whether browsers permit persistent HTTP permissions?
>
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Tue Jun 26 23:35:06 2012
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Subject: Re: [rtcweb] Minutes Tuesday 12th of June 2012 Interim meeting
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Thanks everyone,

Based on that so many has the same recollection then I am certain that
the minutes says the right thing, i.e. PLI Required. Just want to be
sure I wasn't misrepresenting the meetings consensus.

On 2012-06-26 21:43, Martin Thomson wrote:
> On 26 June 2012 10:50, Charles Eckel (eckelcu) <eckelcu@cisco.com>
> wrote:
>> In my notes I have PLI as REQUIRED.
> That was my recollection.
> 
>> I also have RFC 5761 RTP/RTCP multiplexing as REQUIRED, whereas the
>> notes currently reads that support for separate ports is REQUIRED.
>> I would appreciate clarification on this point as well.

Yes, requiring support of RTP and RTCP multiplexing was never a
questioned. The only question discussed was if one also was required
have support for non-multiplexing of RTP and RTCP.

Is this sufficiently clear:

A Consensus call was held: There was no clear consensus that an
end-point only needs to support RTCP multiplexing, thus the consensus is
that an end-point MUST support RTP and RTCP separately in addition to be
REQUIRED to support RTCP Multiplexing when signaled.

Cheers

Magnus Westerlund

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Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Subject: [rtcweb] Propose Vancouver Agenda Topics
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WG,

The WG chairs solicit topics that you think needs to be discussed in
Vancouver. Both authors/editors of documents that have topics they think
requires face to face discussion to reach consensus and WG participants
that think a particular topic should be discussed are of interest.

Two topics that is clear they need further discussion and was not taken
at the interim meeting are
- Additional Key-management for SRTP
- Selection of Mandatory to implement Media Codecs

Please add to the list or comment on the suitability to have these topics.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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Ericsson AB                | Phone  +46 10 7148287
Färögatan 6                | Mobile +46 73 0949079
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From magnus.westerlund@ericsson.com  Wed Jun 27 00:36:47 2012
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Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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WG,

We had a discussion at the interim if RTP Retransmission is to be
considered REQUIRED or RECOMMENDED to implement. I would like to see if
we can first have some discussion on this topic before moving on to see
if we can get a consensus here on the mailing list.

Please provide your views on this topic.

Cheers

Magnus Westerlund
(As Chair and document editor)


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



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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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If you use RTP over a Mobile Networks subject to strong packet loss
(especially 3G or area plenty of many mobile devices), the RTP
retransmission became a requirement.

If there is RTCP support within RTCWeb, we may define some "quality
threshold" after that the RTP Retransmition became REQUIRED.

Fabio

On 6/27/12 9:36 AM, Magnus Westerlund wrote:
> WG,
> 
> We had a discussion at the interim if RTP Retransmission is to be
> considered REQUIRED or RECOMMENDED to implement. I would like to see if
> we can first have some discussion on this topic before moving on to see
> if we can get a consensus here on the mailing list.
> 
> Please provide your views on this topic.
> 
> Cheers
> 
> Magnus Westerlund
> (As Chair and document editor)
> 
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From stefan.lk.hakansson@ericsson.com  Wed Jun 27 01:21:58 2012
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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Would it be feasible to allow only pages delivered over https to use 
persistent consent?

(I think that in certain situations it could be annoying to have to give 
consent every time)

Stefan

On 06/27/2012 01:52 AM, Igor Faynberg wrote:
> Not only that, but I find it hard to understand why anyone would need to
> give persistent consent for accessing media devices to any device.
>
> I also thought the issue was closed for good.
>
> (Sorry, I could not be present at the intereem meeting, even remotely...)
>
> Igor
>
> On 6/26/2012 6:59 PM, Eric Rescorla wrote:
>> In Martin's review of draft-ietf-rtcweb-security-arch, he (re)raised
>> the question of persistent consent for camera/microphone access from
>> pages served over HTTP.  I had thought this issue closed, but in the
>> discussion at the interim there seemed to be a fair amount of
>> sentiment that people hadn't understood the security issues and now
>> believed that this was a bad idea. I agreed to attempt to re-start the
>> discussion with a clearer description of what the problem was, hence
>> this message.
>>
>> Here's the basic attack:
>> Say that I have given persistent permission to access my camera and
>> microphone to any page from http://www.example.com/. Now, I go to an
>> Internet cafe and surf to *any* HTTP page, e.g.,
>> http://www.google.com/. At this point, any attacker who controls the
>> wireless network can redirect that access to point to
>> http://www.example.com/ and inject JS that reads from my camera and
>> microphone.
>>
>> That's pretty bad, but made worse by the following facts:
>> 1. It's not that hard for the attacker to open up
>> popups/popunders/iframes, etc. that are in the domain of example.com
>> but actually execute code from the attacker.  So, once you've opened
>> the browser in a hostile environment, the attacker can bug you at
>> times of his choosing until you close your browser.
>>
>> 2. If your browser does any sort of auto-refresh (e.g., because
>> the page refreshes itself) then you can get infected even if you
>> don't do anything deliberate.
>>
>> 3. This also applies if you are using any kind of open wireless
>> network, such as if your home network doesn't use a strong
>> enough WPA key.
>>
>>
>> So, the executive summary is: if you have a persistent permission
>> for an HTTP site, and you ever use your browser on any insecure
>> network, an attacker can bug your computer until you close your
>> browser. This seems bad.
>>
>> Does this change people's opinions about what the rules should be
>> about whether browsers permit persistent HTTP permissions?
>>
>> -Ekr
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



From internet-drafts@ietf.org  Wed Jun 27 05:33:55 2012
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-09.txt
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Real-Time Communication in WEB-browsers W=
orking Group of the IETF.

	Title           : Web Real-Time Communication Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-09.txt
	Pages           : 28
	Date            : 2012-06-27

Abstract:
   This document describes web based real-time communication use-cases.
   Based on the use-cases, the document also derives requirements
   related to the browser, and the API used by web applications to
   request and control media stream and data exchange services provided
   by the browser.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requiremen=
ts

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-use-cases-and-requirements-09

A diff from previous version is available at:
http://tools.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-use-cases-and-requir=
ements-09


Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/


From stefan.lk.hakansson@ericsson.com  Wed Jun 27 05:42:44 2012
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Date: Wed, 27 Jun 2012 14:42:34 +0200
From: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>
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Subject: [rtcweb] Use-case draft updated
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Hi,

as already automatically announced, the use-case draft has been updated. 
Extract from the change log:

* Changed "eavesdropping" to "wiretapping" and referenced RFC2804.

* Removed informal ref webrtc_req; that document has been abandoned by 
the W3C webrtc WG.

* Added use-case where one user is behind a FW that only allows http; 
derived req. F37.
					
* Changed F24 slightly; MUST-> SHOULD, inserted "available".
					
* Added a clause to "Simple video communication service" saying that the 
service provider monitors the quality of service, and derived reqs F38 
and A26.

Most of the above are things that was documented in the minutes of the 
Interim June 12th; I took the liberty to add some text (and reqs 
A26/F38) on the service provider monitoring the QoE.

Feedback and comments are most welcome.

Also, Ekr has the task to deliver an Identity related use-case (as 
discussed at the interim).

Stefan

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From: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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On Wed, Jun 27, 2012 at 1:21 AM, Stefan Hakansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> Would it be feasible to allow only pages delivered over https to use
> persistent consent?

Yes, that is what Iw ould propose.

-Ekr

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This makes sense to me.  Persistent consent over http is pretty dangerous.=
=20

> From: ekr@rtfm.com
> Date: Wed=2C 27 Jun 2012 05:47:04 -0700
> To: stefan.lk.hakansson@ericsson.com
> CC: rtcweb@ietf.org
> Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
>=20
> On Wed=2C Jun 27=2C 2012 at 1:21 AM=2C Stefan Hakansson LK
> <stefan.lk.hakansson@ericsson.com> wrote:
> > Would it be feasible to allow only pages delivered over https to use
> > persistent consent?
>=20
> Yes=2C that is what Iw ould propose.
>=20
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
This makes sense to me.&nbsp=3B Persistent consent over http is pretty dang=
erous. <br><br><div><div id=3D"SkyDrivePlaceholder"></div>&gt=3B From: ekr@=
rtfm.com<br>&gt=3B Date: Wed=2C 27 Jun 2012 05:47:04 -0700<br>&gt=3B To: st=
efan.lk.hakansson@ericsson.com<br>&gt=3B CC: rtcweb@ietf.org<br>&gt=3B Subj=
ect: Re: [rtcweb] Why persistent consent for HTTP is a problem<br>&gt=3B <b=
r>&gt=3B On Wed=2C Jun 27=2C 2012 at 1:21 AM=2C Stefan Hakansson LK<br>&gt=
=3B &lt=3Bstefan.lk.hakansson@ericsson.com&gt=3B wrote:<br>&gt=3B &gt=3B Wo=
uld it be feasible to allow only pages delivered over https to use<br>&gt=
=3B &gt=3B persistent consent?<br>&gt=3B <br>&gt=3B Yes=2C that is what Iw =
ould propose.<br>&gt=3B <br>&gt=3B -Ekr<br>&gt=3B _________________________=
______________________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.=
org<br>&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	  =
 		  </div></body>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Fabio said:=20

> If you use RTP over a Mobile Networks subject to strong packet loss
> (especially 3G or area plenty of many mobile devices)=2C the RTP
> retransmission became a requirement.

[BA] The usefulness of retransmission will depend on the RTT.  There are qu=
ite a few
mobile networks that suffer from buffer bloat to the point where retransmis=
sion is quite useless.=20
Also=2C retransmission (and FEC) both increase bandwidth usage which is a p=
roblem for congestion=20
control.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
Fabio said: <br><br><div>&gt=3B If you use RTP over a Mobile Networks subje=
ct to strong packet loss<br>&gt=3B (especially 3G or area plenty of many mo=
bile devices)=2C the RTP<br>&gt=3B retransmission became a requirement.<br>=
<br>[BA] The usefulness of retransmission will depend on the RTT.&nbsp=3B T=
here are quite a few<br>mobile networks that suffer from buffer bloat to th=
e point where retransmission is quite useless. <br>Also=2C retransmission (=
and FEC) both increase bandwidth usage which is a problem for congestion <b=
r>control. <br></div> 		 	   		  </div></body>
</html>=

--_052844c0-dec2-47fb-b0aa-42bad20a5a79_--

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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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On 06/27/2012 09:23 AM, Bernard Aboba wrote:
> This makes sense to me.  Persistent consent over http is pretty dangerous.

Yes, this makes sense to me as well. Even persistent consent to pages 
served over HTTPS would need some sort of certificate fingerprint 
caching to notify the user if the certificate of the site has changed 
since they provided consent, in my opinion.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org



From martin.thomson@gmail.com  Wed Jun 27 08:57:41 2012
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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This discussion is reminiscent of the long-standing joke about the
"ravish me" bit.  We should suggest that browsers hide the "remember
my choice" checkbox for unsecured sites.

Note that I also believe that this needs to include sites with mixed
content.  The idea that a site might degrade into mixed content after
an input device is acquired is probably a corner case that we can
ignore unless someone has a brilliant suggestion.

--Martin

From cb.list6@gmail.com  Wed Jun 27 09:19:01 2012
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I think Bernard noted this as well, but it is not so much buffer
bloat... but the issue is indeed latency, not loss.  Consequently,
re-transmission not productive.

On Wed, Jun 27, 2012 at 12:42 AM, Fabio Pietrosanti (naif)
<lists@infosecurity.ch> wrote:
> If you use RTP over a Mobile Networks subject to strong packet loss
> (especially 3G or area plenty of many mobile devices), the RTP
> retransmission became a requirement.
>

Please cite the packet loss you are referring to.

Here is one imperfect data source,
http://pam2012.ftw.at/papers/PAM2012paper6.pdf.

I say it is imperfect because it relies on SACK, which can infer
packet loss, but frequently for mobile it just infers delay.  So, the
mobile packet loss is overstated.

Mobile (UMTS, ...) networks have extremely robust forward error
correct and re-transmission capabilities at layer 1 and layer 2, this
induces delay but ensure packet delivery.

See: http://en.wikipedia.org/wiki/Hybrid_automatic_repeat_request

Attempting to optimize for packet delivery at higher levels (RTP
re-transmission...)  leads to various counter productive race
conditions, since the RAN (radio access network) is designed for 100%
packet delivery.

CB

> If there is RTCP support within RTCWeb, we may define some "quality
> threshold" after that the RTP Retransmition became REQUIRED.
>
> Fabio
>
> On 6/27/12 9:36 AM, Magnus Westerlund wrote:
>> WG,
>>
>> We had a discussion at the interim if RTP Retransmission is to be
>> considered REQUIRED or RECOMMENDED to implement. I would like to see if
>> we can first have some discussion on this topic before moving on to see
>> if we can get a consensus here on the mailing list.
>>
>> Please provide your views on this topic.
>>
>> Cheers
>>
>> Magnus Westerlund
>> (As Chair and document editor)
>>
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287
>> F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From igor.faynberg@alcatel-lucent.com  Wed Jun 27 09:48:38 2012
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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Still, my question is: Why would anyone want to give "persistent 
consent" for device access to any site in the first place?..

Sorry for badgering.

Igor

On 6/27/2012 11:12 AM, Kevin P. Fleming wrote:
> On 06/27/2012 09:23 AM, Bernard Aboba wrote:
>> This makes sense to me.  Persistent consent over http is pretty 
>> dangerous.
>
> Yes, this makes sense to me as well. Even persistent consent to pages 
> served over HTTPS would need some sort of certificate fingerprint 
> caching to notify the user if the certificate of the site has changed 
> since they provided consent, in my opinion.
>

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On 27 June 2012 09:48, Igor Faynberg <igor.faynberg@alcatel-lucent.com> wrote:
> Still, my question is: Why would anyone want to give "persistent consent"
> for device access to any site in the first place?..

That's a matter of a user's trust vs. annoyance tradeoff.  For a site
that a user has relatively high trust in, they might find the
annoyance of having to click through "Do you want to allow
_example.com_ to access your microphone? y/n" every time that they use
that site to make audio calls to be sufficiently high to have a
persistent permission for that site.

There is also a security benefit from persistence.  Requiring
click-through trains users to click through.  A user that is trained
in this fashion becomes easier to exploit.

Note that the manifestation of persistence is still entirely left to
browser implementers.  If and how they implement the feature is up to
them.  No doubt it will be subject to careful design and extensive
testing.

--Martin

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On 6/27/12 6:19 PM, Cameron Byrne wrote:
> I think Bernard noted this as well, but it is not so much buffer
> bloat... but the issue is indeed latency, not loss.  Consequently,
> re-transmission not productive.
> 
> On Wed, Jun 27, 2012 at 12:42 AM, Fabio Pietrosanti (naif)
> <lists@infosecurity.ch> wrote:
>> If you use RTP over a Mobile Networks subject to strong packet loss
>> (especially 3G or area plenty of many mobile devices), the RTP
>> retransmission became a requirement.
>>
> 
> Please cite the packet loss you are referring to.

My company does Mobile VoIP Clients for Blackberry, Nokia S60, iPhone,
Android and i know that Packet Loss is the *major cause of bad quality*
reported by my customer.

Given empirical practical experience i have from my customer:
 - In crowded areas in 3G
 - Within certain building (both 2G and 3G)

I don't know which kind of retransmission schemas you are referring to,
but i am sure that packet loss of UDP packets in a mobile network exists
and it's a concrete issue.

Fabio

From radhika.r.roy.civ@mail.mil  Wed Jun 27 11:14:45 2012
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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Cameron Byrne <cb.list6@gmail.com>, "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Hi, all:

Real-time protocols (Audio/Video payloads for two-way conversations) over U=
DP/IP were being developed based on the single thing: Retransmission delays=
 are unacceptable (more over congestion problems especially for the high-ba=
ndwidth videos).

If this fundamental performance characteristic for the Real-time protocols =
(Audio/Video payloads for two-way conversations) need to be changed, the ex=
perimental results with lots of actual measurements must be produced to the=
 technical communities.

Otherwise, someone may make some subjective statements here and there witho=
ut any actual measurements will only provide evidences what those folks do =
not know what they are actually talking about creating technical confusions=
.

Best regards,
Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cameron Byrne
Sent: Wednesday, June 27, 2012 12:19 PM
To: Fabio Pietrosanti (naif)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

I think Bernard noted this as well, but it is not so much buffer bloat... b=
ut the issue is indeed latency, not loss.  Consequently, re-transmission no=
t productive.

On Wed, Jun 27, 2012 at 12:42 AM, Fabio Pietrosanti (naif) <lists@infosecur=
ity.ch> wrote:
> If you use RTP over a Mobile Networks subject to strong packet loss=20
> (especially 3G or area plenty of many mobile devices), the RTP=20
> retransmission became a requirement.
>

Please cite the packet loss you are referring to.

Here is one imperfect data source,
http://pam2012.ftw.at/papers/PAM2012paper6.pdf.

I say it is imperfect because it relies on SACK, which can infer packet los=
s, but frequently for mobile it just infers delay.  So, the mobile packet l=
oss is overstated.

Mobile (UMTS, ...) networks have extremely robust forward error correct and=
 re-transmission capabilities at layer 1 and layer 2, this induces delay bu=
t ensure packet delivery.

See: http://en.wikipedia.org/wiki/Hybrid_automatic_repeat_request

Attempting to optimize for packet delivery at higher levels (RTP
re-transmission...)  leads to various counter productive race conditions, s=
ince the RAN (radio access network) is designed for 100% packet delivery.

CB

> If there is RTCP support within RTCWeb, we may define some "quality=20
> threshold" after that the RTP Retransmition became REQUIRED.
>
> Fabio
>
> On 6/27/12 9:36 AM, Magnus Westerlund wrote:
>> WG,
>>
>> We had a discussion at the interim if RTP Retransmission is to be=20
>> considered REQUIRED or RECOMMENDED to implement. I would like to see=20
>> if we can first have some discussion on this topic before moving on=20
>> to see if we can get a consensus here on the mailing list.
>>
>> Please provide your views on this topic.
>>
>> Cheers
>>
>> Magnus Westerlund
>> (As Chair and document editor)
>>
>>
>> ---------------------------------------------------------------------
>> - Multimedia Technologies, Ericsson Research EAB/TVM
>> ---------------------------------------------------------------------
>> - Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287 F=
=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0
>> | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ---------------------------------------------------------------------
>> -
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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On 6/27/12 8:14 PM, Roy, Radhika R CIV (US) wrote:
> Hi, all:
> 
> Real-time protocols (Audio/Video payloads for two-way conversations) over UDP/IP were being developed based on the single thing: Retransmission delays are unacceptable (more over congestion problems especially for the high-bandwidth videos).
> 
> If this fundamental performance characteristic for the Real-time protocols (Audio/Video payloads for two-way conversations) need to be changed, the experimental results with lots of actual measurements must be produced to the technical communities.
> 
> Otherwise, someone may make some subjective statements here and there without any actual measurements will only provide evidences what those folks do not know what they are actually talking about creating technical confusions.

It's not a subjective topic that over a mobile network there is packet
loss and that there are known condition where this lead to unacceptable
voip quality.

The Internet Measurement of VoIP on different Transport Layer Protocols
http://netarchlab.tsinghua.edu.cn/PersonalMainPage/wzl/public_html/pubs/2009_ICOIN_transport_measure.pdf

SipDroid: An opensource Mobile client using RTP Retransmission to
improve quality over 3G network with packet loss:
http://code.google.com/p/sipdroid/wiki/NewImprovedAudio

RTP Retransmission RFC (explaining how to handle dynamically
retransmission from RTCP feedback):
http://tools.ietf.org/html/rfc4588

Fabio

From radhika.r.roy.civ@mail.mil  Wed Jun 27 12:03:09 2012
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Hi, Fabio:

I could not retrieve the first one, but other two links are OK.

I understand these things, but still they are NOT sufficient enough to gene=
ralize these results because of the following:

1. Experiments seem to be local without any precise measurements under all =
possible circumstances noting the network traffic, audio quality, video qua=
lity, and precise applications and number of subjects under test using diff=
erent applications (e.g. point-to-point audio & video conf, multipoint audi=
o & video conf, streaming audio/video applications). That is, it is not glo=
bal, that means, there is NOT a wide area network (e.g. public Internet) co=
nnected with two different access networks where end-to-end one-way propaga=
tion delay is around 120-150 milliseconds, types of applications, traffic l=
oads, subjects who were under test, and others.

2. If we do experiments over the high-speed LANs connecting users in the sa=
me LANs, we may find many things are possible including retransmissions of =
audios/videos. We can then extend these networking considerations unless an=
d until we get a situation where retransmissions are no longer applicable. =
RFCs with retransmissions of audios/videos may be acceptable considering th=
e trade-offs of end-to-end-delay and jitter-buffer-sizes under those limite=
d circumstances. Each of those precise conditions can be noted generated. A=
gain, these are very specific and case-by-case basis and cannot be generali=
zed.=20

3. Video quality measurements need to be taken in terms of real-time point-=
to-point and multipoint conferencing along with audio and video. Then these=
 experimental results need to be measured separately for both audio and vid=
eo separately. For example, at what point lip-synchronization is no longer =
possible.

Unless one can note all of those special cases for retransmissions of audio=
/video, it will always be technically confusing to use the retransmissions =
of audio and video as a generalized one.

I also know one case such as wiretapping applications. For silent recording=
 in a distance location, audio & video retransmissions are needed because t=
he exact things that are produced legally in the court need to be near 100%=
 correct.

Hope this helps.

Radhika=20


-----Original Message-----
From: Fabio Pietrosanti [mailto:naif@infosecurity.ch] On Behalf Of Fabio Pi=
etrosanti (naif)
Sent: Wednesday, June 27, 2012 2:23 PM
To: Roy, Radhika R CIV (US)
Cc: Cameron Byrne; rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

On 6/27/12 8:14 PM, Roy, Radhika R CIV (US) wrote:
> Hi, all:
>=20
> Real-time protocols (Audio/Video payloads for two-way conversations) over=
 UDP/IP were being developed based on the single thing: Retransmission dela=
ys are unacceptable (more over congestion problems especially for the high-=
bandwidth videos).
>=20
> If this fundamental performance characteristic for the Real-time protocol=
s (Audio/Video payloads for two-way conversations) need to be changed, the =
experimental results with lots of actual measurements must be produced to t=
he technical communities.
>=20
> Otherwise, someone may make some subjective statements here and there wit=
hout any actual measurements will only provide evidences what those folks d=
o not know what they are actually talking about creating technical confusio=
ns.

It's not a subjective topic that over a mobile network there is packet loss=
 and that there are known condition where this lead to unacceptable voip qu=
ality.

The Internet Measurement of VoIP on different Transport Layer Protocols htt=
p://netarchlab.tsinghua.edu.cn/PersonalMainPage/wzl/public_html/pubs/2009_I=
COIN_transport_measure.pdf

SipDroid: An opensource Mobile client using RTP Retransmission to improve q=
uality over 3G network with packet loss:
http://code.google.com/p/sipdroid/wiki/NewImprovedAudio

RTP Retransmission RFC (explaining how to handle dynamically retransmission=
 from RTCP feedback):
http://tools.ietf.org/html/rfc4588

Fabio

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On 27 Jun 2012, at 17:19, Cameron Byrne wrote:
> I think Bernard noted this as well, but it is not so much buffer
> bloat... but the issue is indeed latency, not loss.  Consequently,
> re-transmission not productive.

This surely depends on the network being used, and the acceptable =
latency for the application. Making blanket statements that =
retransmission is, or is not, productive are not helpful, since it's =
usefulness is very context dependent.

Colin




> On Wed, Jun 27, 2012 at 12:42 AM, Fabio Pietrosanti (naif)
> <lists@infosecurity.ch> wrote:
>> If you use RTP over a Mobile Networks subject to strong packet loss
>> (especially 3G or area plenty of many mobile devices), the RTP
>> retransmission became a requirement.
>>=20
>=20
> Please cite the packet loss you are referring to.
>=20
> Here is one imperfect data source,
> http://pam2012.ftw.at/papers/PAM2012paper6.pdf.
>=20
> I say it is imperfect because it relies on SACK, which can infer
> packet loss, but frequently for mobile it just infers delay.  So, the
> mobile packet loss is overstated.
>=20
> Mobile (UMTS, ...) networks have extremely robust forward error
> correct and re-transmission capabilities at layer 1 and layer 2, this
> induces delay but ensure packet delivery.
>=20
> See: http://en.wikipedia.org/wiki/Hybrid_automatic_repeat_request
>=20
> Attempting to optimize for packet delivery at higher levels (RTP
> re-transmission...)  leads to various counter productive race
> conditions, since the RAN (radio access network) is designed for 100%
> packet delivery.
>=20
> CB
>=20
>> If there is RTCP support within RTCWeb, we may define some "quality
>> threshold" after that the RTP Retransmition became REQUIRED.
>>=20
>> Fabio
>>=20
>> On 6/27/12 9:36 AM, Magnus Westerlund wrote:
>>> WG,
>>>=20
>>> We had a discussion at the interim if RTP Retransmission is to be
>>> considered REQUIRED or RECOMMENDED to implement. I would like to see =
if
>>> we can first have some discussion on this topic before moving on to =
see
>>> if we can get a consensus here on the mailing list.
>>>=20
>>> Please provide your views on this topic.
>>>=20
>>> Cheers
>>>=20
>>> Magnus Westerlund
>>> (As Chair and document editor)
>>>=20
>>>=20
>>> =
----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> =
----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




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On Wed, Jun 27, 2012 at 1:06 PM, Colin Perkins <csp@csperkins.org> wrote:
> On 27 Jun 2012, at 17:19, Cameron Byrne wrote:
>> I think Bernard noted this as well, but it is not so much buffer
>> bloat... but the issue is indeed latency, not loss. =A0Consequently,
>> re-transmission not productive.
>
> This surely depends on the network being used, and the acceptable latency=
 for the application. Making blanket statements that retransmission is, or =
is not, productive are not helpful, since it's usefulness is very context d=
ependent.
>
> Colin

Allow me to set context.

A comment was made that re-transmission of packets is a good idea
because mobile networks drop a lot of packets.

My comment is that re-transmission of RTP is not productive in 3GPP
wireless networks such as GSM, UMTS, or LTE.

This is because L2 mechanisms do forward error correction and PDU
acknowledgement and re-transmission.  The RAN system does not have
packet loss, generally speaking.  Packet loss may occur on the
internet prior to entering the RAN, this is a separate issue than the
specific case of mobile.

What is sometimes perceived as packet loss is more often than not,
just latency / jitter due to local L2 re-transmission.   RTP packets
that are too late are discarded today, and IMHO that is the right
approach (the late packet is no longer relevant)

RTP re-transmission from the source will not improve this situation.
In fact, if the issue is congestion (the channel is too narrow) then
re-transmission may make the problem worse (channel becomes more
narrow with higher load of re-transmitted packets)

I don't believe anyone has provided meaningful and relevant
quantitative data here that there is a problem, or a solution, for the
case of mobile and re-transmitted RTP.

Cameron

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Cameron,

I generally agree. Of course mobile is not the only scenario where =
WebRTC might be used, and the question being asked is not whether =
retransmission is to be used in any particular scenario, but whether it =
is RECOMMENDED or REQUIRED to implement.

Colin



On 27 Jun 2012, at 21:23, Cameron Byrne wrote:
> On Wed, Jun 27, 2012 at 1:06 PM, Colin Perkins <csp@csperkins.org> =
wrote:
>> On 27 Jun 2012, at 17:19, Cameron Byrne wrote:
>>> I think Bernard noted this as well, but it is not so much buffer
>>> bloat... but the issue is indeed latency, not loss.  Consequently,
>>> re-transmission not productive.
>>=20
>> This surely depends on the network being used, and the acceptable =
latency for the application. Making blanket statements that =
retransmission is, or is not, productive are not helpful, since it's =
usefulness is very context dependent.
>>=20
>> Colin
>=20
> Allow me to set context.
>=20
> A comment was made that re-transmission of packets is a good idea
> because mobile networks drop a lot of packets.
>=20
> My comment is that re-transmission of RTP is not productive in 3GPP
> wireless networks such as GSM, UMTS, or LTE.
>=20
> This is because L2 mechanisms do forward error correction and PDU
> acknowledgement and re-transmission.  The RAN system does not have
> packet loss, generally speaking.  Packet loss may occur on the
> internet prior to entering the RAN, this is a separate issue than the
> specific case of mobile.
>=20
> What is sometimes perceived as packet loss is more often than not,
> just latency / jitter due to local L2 re-transmission.   RTP packets
> that are too late are discarded today, and IMHO that is the right
> approach (the late packet is no longer relevant)
>=20
> RTP re-transmission from the source will not improve this situation.
> In fact, if the issue is congestion (the channel is too narrow) then
> re-transmission may make the problem worse (channel becomes more
> narrow with higher load of re-transmitted packets)
>=20
> I don't believe anyone has provided meaningful and relevant
> quantitative data here that there is a problem, or a solution, for the
> case of mobile and re-transmitted RTP.
>=20
> Cameron



--=20
Colin Perkins
http://csperkins.org/




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Cc: rtcweb@ietf.org, Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 6/27/12 10:23 PM, Cameron Byrne wrote:
> On Wed, Jun 27, 2012 at 1:06 PM, Colin Perkins <csp@csperkins.org> wrote:
>> On 27 Jun 2012, at 17:19, Cameron Byrne wrote:
>>> I think Bernard noted this as well, but it is not so much buffer
>>> bloat... but the issue is indeed latency, not loss.  Consequently,
>>> re-transmission not productive.
>>
>> This surely depends on the network being used, and the acceptable latency for the application. Making blanket statements that retransmission is, or is not, productive are not helpful, since it's usefulness is very context dependent.
>>
>> Colin
> 
> Allow me to set context.
> 
> A comment was made that re-transmission of packets is a good idea
> because mobile networks drop a lot of packets.
> 
> My comment is that re-transmission of RTP is not productive in 3GPP
> wireless networks such as GSM, UMTS, or LTE.
> 
> This is because L2 mechanisms do forward error correction and PDU
> acknowledgement and re-transmission.  The RAN system does not have
> packet loss, generally speaking.  Packet loss may occur on the
> internet prior to entering the RAN, this is a separate issue than the
> specific case of mobile.

You are saying that in EDGE, UMTS, HSDPA when an IP stack send an UDP
packet, it will handled from L2 like in the old age of GSM CSD and be
automatically retransmitted?

The L2 RLC can goes in:

* "acknolwedged mode" (handling retransmission)

* "unacknowledged mode" (no retransmission).

UDP streaming packets over Mobile Data Networks goes normally in
"unacknowledge mode".

As far as my professional experience VoIP over mobile can be normally be
subject to packet loss.

Fabio

p.s. however i agree that's a particular use cases of WebRTC

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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On Wed, Jun 27, 2012 at 1:50 PM, Fabio Pietrosanti (naif)
<lists@infosecurity.ch> wrote:
> On 6/27/12 10:23 PM, Cameron Byrne wrote:
>> On Wed, Jun 27, 2012 at 1:06 PM, Colin Perkins <csp@csperkins.org> wrote=
:
>>> On 27 Jun 2012, at 17:19, Cameron Byrne wrote:
>>>> I think Bernard noted this as well, but it is not so much buffer
>>>> bloat... but the issue is indeed latency, not loss. =A0Consequently,
>>>> re-transmission not productive.
>>>
>>> This surely depends on the network being used, and the acceptable laten=
cy for the application. Making blanket statements that retransmission is, o=
r is not, productive are not helpful, since it's usefulness is very context=
 dependent.
>>>
>>> Colin
>>
>> Allow me to set context.
>>
>> A comment was made that re-transmission of packets is a good idea
>> because mobile networks drop a lot of packets.
>>
>> My comment is that re-transmission of RTP is not productive in 3GPP
>> wireless networks such as GSM, UMTS, or LTE.
>>
>> This is because L2 mechanisms do forward error correction and PDU
>> acknowledgement and re-transmission. =A0The RAN system does not have
>> packet loss, generally speaking. =A0Packet loss may occur on the
>> internet prior to entering the RAN, this is a separate issue than the
>> specific case of mobile.
>
> You are saying that in EDGE, UMTS, HSDPA when an IP stack send an UDP
> packet, it will handled from L2 like in the old age of GSM CSD and be
> automatically retransmitted?
>

Yes. LTE too.

Cameron

> The L2 RLC can goes in:
>
> * "acknolwedged mode" (handling retransmission)
>
> * "unacknowledged mode" (no retransmission).
>
> UDP streaming packets over Mobile Data Networks goes normally in
> "unacknowledge mode".
>
> As far as my professional experience VoIP over mobile can be normally be
> subject to packet loss.
>
> Fabio
>
> p.s. however i agree that's a particular use cases of WebRTC

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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 6/27/12 10:54 PM, Cameron Byrne wrote:
>> You are saying that in EDGE, UMTS, HSDPA when an IP stack send an UDP
>> packet, it will handled from L2 like in the old age of GSM CSD and be
>> automatically retransmitted?
>>
> 
> Yes. LTE too.

I read (by googling cause i don't have LTE access) that also LTE uses
for IPTV and all other IP streaming applications the UM_RLC
(unacknowledged mode).

So also LTE should not do retransmission, unless being set in AM_RLC
(acknowledged mode).

Fabio

From ron.even.tlv@gmail.com  Wed Jun 27 23:25:21 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Hi,
My view is that retransmission have great value for content delivery or
streaming applications, when the communication is unidirectional.=20
For conversional applications like we are defining in RTCweb, =
retransmission
has the lower value comparing to other loss recovery options like FEC or
receiver based recovery. The latency involved with the extra roundtrips
makes it not effective comparing to typical jitter buffer sizes at the
receiver, note that receiver will take correction actions even for late
arrival and will not try to dynamically change the jitter buffer to a =
larger
size beyond the maximum threshold of end to end delay.=20
 If we look at the typical behavior of conversional applications, they =
will
add to audio on the sender side FEC or even send the media twice is
subsequent packets encoded in different encoders,( of course for the
conversation itself the listener can always ask the talker to repeat =
since
he did not hear it clearly). On the video FEC and receiver side recovery
like motion estimation is common.
I propose that retransmission can be recommended and in my view for
conversional usage even this is strong.

Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 27 June, 2012 9:37 AM
To: rtcweb@ietf.org
Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or =
RECOMMENDED

WG,

We had a discussion at the interim if RTP Retransmission is to be =
considered
REQUIRED or RECOMMENDED to implement. I would like to see if we can =
first
have some discussion on this topic before moving on to see if we can get =
a
consensus here on the mailing list.

Please provide your views on this topic.

Cheers

Magnus Westerlund
(As Chair and document editor)


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


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From g.kiranreddy4u@gmail.com  Thu Jun 28 03:09:45 2012
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From: kiran kumar <g.kiranreddy4u@gmail.com>
Date: Thu, 28 Jun 2012 11:55:57 +0530
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Subject: [rtcweb] Reg fig1 in draft-ietf-rtcweb-jsep-01
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--14dae9340e01c788b804c3826886
Content-Type: text/plain; charset=ISO-8859-1

Hi all,

IMO the figure1 in JSEP draft seems to be more appropriate if it has a
double headed arrow for SDP communication between Web App and Browser.

       +---------------+
+---------------+
       |  Web App  |<--- App-Specific Signaling --->|  Web App  |
       +----------------+
+---------------+

^
^

|
|
             |
SDP                                                                   |  SDP

V
 V
       +-----------+
 +--------------+
       |  Browser  |<----------- Media ------------------->|  Browser  |
       +-----------+
 +--------------+


Thanks,
Kiran.

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From magnus.westerlund@ericsson.com  Thu Jun 28 03:10:30 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Hi,

As Individual I like to state my position.

We have a video conference system developed by my colleagues used
internally at Ericsson that uses RTP Retransmission for video, not for
audio with great success. This is implemented such that we actually
allow the video to fall behind the audio when packet loss and
retransmission is not able to repair in a timely enough fashion. The
benefit is minimal overhead and still no loss induced degradations in
the video. Yes, we get degradation in form of frame display jittering
and short freezes. But those events that are truly visible are rare over
wired networks.

I am personally convinced that RTP Retransmission is great tool in the
toolbox when it comes to improve media quality in many use cases. Yes
there are scenarios where RTP retransmission is less efficient. Long
RTTs (over 200-400 ms) is the primary source of degradations. Compared
to FEC it so much more efficient from bandwidth consumption perspective.

I also think it is important that we have some mandatory to implement
tool for making the transport more robust now that we have a consensus
that we are not going for a FEC solution in the initial specification.

Thus my personal position is that RTP Retransmission should be REQUIRED
to implement.

Cheers

Magnus


On 2012-06-27 09:36, Magnus Westerlund wrote:
> WG,
> 
> We had a discussion at the interim if RTP Retransmission is to be
> considered REQUIRED or RECOMMENDED to implement. I would like to see if
> we can first have some discussion on this topic before moving on to see
> if we can get a consensus here on the mailing list.
> 
> Please provide your views on this topic.
> 
> Cheers
> 
> Magnus Westerlund
> (As Chair and document editor)
> 
> 
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> Färögatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

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




From michawe@ifi.uio.no  Thu Jun 28 03:30:13 2012
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Date: Thu, 28 Jun 2012 12:30:06 +0200
From: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Hi,

I second that.

Here's one more reason: I believe that it should also be possible for a 
receiver to make use of retransmitted I-macroblocks from old frames that 
have been shown long ago. Future MBs may still depend on them, and so 
their quality could be improved with that information.

This is a quite theoretical idea - I don't have any proof that this 
would work, and don't know of research that has that proof (but I 
suspect one can find some, if one looks). It's just one more reason why 
retransmissions could turn out to be valuable.

Cheers,
Michael


On 6/28/12 9:49 AM, Magnus Westerlund wrote:
> Hi,
>
> As Individual I like to state my position.
>
> We have a video conference system developed by my colleagues used
> internally at Ericsson that uses RTP Retransmission for video, not for
> audio with great success. This is implemented such that we actually
> allow the video to fall behind the audio when packet loss and
> retransmission is not able to repair in a timely enough fashion. The
> benefit is minimal overhead and still no loss induced degradations in
> the video. Yes, we get degradation in form of frame display jittering
> and short freezes. But those events that are truly visible are rare over
> wired networks.
>
> I am personally convinced that RTP Retransmission is great tool in the
> toolbox when it comes to improve media quality in many use cases. Yes
> there are scenarios where RTP retransmission is less efficient. Long
> RTTs (over 200-400 ms) is the primary source of degradations. Compared
> to FEC it so much more efficient from bandwidth consumption perspective.
>
> I also think it is important that we have some mandatory to implement
> tool for making the transport more robust now that we have a consensus
> that we are not going for a FEC solution in the initial specification.
>
> Thus my personal position is that RTP Retransmission should be REQUIRED
> to implement.
>
> Cheers
>
> Magnus
>
>
> On 2012-06-27 09:36, Magnus Westerlund wrote:
>> WG,
>>
>> We had a discussion at the interim if RTP Retransmission is to be
>> considered REQUIRED or RECOMMENDED to implement. I would like to see if
>> we can first have some discussion on this topic before moving on to see
>> if we can get a consensus here on the mailing list.
>>
>> Please provide your views on this topic.
>>
>> Cheers
>>
>> Magnus Westerlund
>> (As Chair and document editor)
>>
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>



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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Hi, Magnus:

The audio/video performances for both point-to-point and multipoint confere=
ncing must be very carefully articulated, not just what your colleagues did=
 or did not do. All networking parameters have been discussed in list email=
 including lip-synchronization.

So, let us see all the details of networking conditions in the published pa=
pers. We can then examine what is this all about.

The only thing that I am pointing out that these types of simplistic emails=
 are not good enough to convincing the technical communities unless all the=
 technical details are presented with precise measurements. Both subjective=
 and quantitative performance parameters are involved along the many subjec=
ts who remain in the test environments.=20

If anyone thinks that RTP retransmissions is a great tool, it is OK. Let th=
em keep is private as many usages of RTP retransmissions were also discusse=
d via emails.

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: Thursday, June 28, 2012 3:49 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

Hi,

As Individual I like to state my position.

We have a video conference system developed by my colleagues used internall=
y at Ericsson that uses RTP Retransmission for video, not for audio with gr=
eat success. This is implemented such that we actually allow the video to f=
all behind the audio when packet loss and retransmission is not able to rep=
air in a timely enough fashion. The benefit is minimal overhead and still n=
o loss induced degradations in the video. Yes, we get degradation in form o=
f frame display jittering and short freezes. But those events that are trul=
y visible are rare over wired networks.

I am personally convinced that RTP Retransmission is great tool in the tool=
box when it comes to improve media quality in many use cases. Yes there are=
 scenarios where RTP retransmission is less efficient. Long RTTs (over 200-=
400 ms) is the primary source of degradations. Compared to FEC it so much m=
ore efficient from bandwidth consumption perspective.

I also think it is important that we have some mandatory to implement tool =
for making the transport more robust now that we have a consensus that we a=
re not going for a FEC solution in the initial specification.

Thus my personal position is that RTP Retransmission should be REQUIRED to =
implement.

Cheers

Magnus




From radhika.r.roy.civ@mail.mil  Thu Jun 28 03:39:20 2012
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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Roni Even <ron.even.tlv@gmail.com>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Hi, Roni:

Why do you want to recommend something that cannot be generalized conversio=
nal applications just because of more tools can be sold in the market?

No one is objecting to generalize the RTP retransmissions for the content d=
elivery or streaming applications.

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roni Even
Sent: Thursday, June 28, 2012 3:25 AM
To: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

Hi,
My view is that retransmission have great value for content delivery or str=
eaming applications, when the communication is unidirectional.=20
For conversional applications like we are defining in RTCweb, retransmissio=
n has the lower value comparing to other loss recovery options like FEC or =
receiver based recovery. The latency involved with the extra roundtrips mak=
es it not effective comparing to typical jitter buffer sizes at the receive=
r, note that receiver will take correction actions even for late arrival an=
d will not try to dynamically change the jitter buffer to a larger size bey=
ond the maximum threshold of end to end delay.=20
 If we look at the typical behavior of conversional applications, they will=
 add to audio on the sender side FEC or even send the media twice is subseq=
uent packets encoded in different encoders,( of course for the conversation=
 itself the listener can always ask the talker to repeat since he did not h=
ear it clearly). On the video FEC and receiver side recovery like motion es=
timation is common.
I propose that retransmission can be recommended and in my view for convers=
ional usage even this is strong.

Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: 27 June, 2012 9:37 AM
To: rtcweb@ietf.org
Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED

WG,

We had a discussion at the interim if RTP Retransmission is to be considere=
d REQUIRED or RECOMMENDED to implement. I would like to see if we can first=
 have some discussion on this topic before moving on to see if we can get a=
 consensus here on the mailing list.

Please provide your views on this topic.

Cheers

Magnus Westerlund
(As Chair and document editor)


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


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Thu Jun 28 04:34:40 2012
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 2012-06-28 12:30, Roy, Radhika R CIV (US) wrote:
> Hi, Magnus:
> 
> The audio/video performances for both point-to-point and multipoint
> conferencing must be very carefully articulated, not just what your
> colleagues did or did not do. All networking parameters have been
> discussed in list email including lip-synchronization.
> 
> So, let us see all the details of networking conditions in the
> published papers. We can then examine what is this all about.

Yes, this would be desirable but I see as unrealistic amounts of work to
go through for each technical functionality. Also it is something that
is not possible. As WebRTC will be used over the very heterogeneous
network of networks called Internet. We also don't have any firm
definition of the service that we tries to fulfill.

I have been part of standardization work for services, like 3GPP's
Packet-based Streaming Service (PSS) where we have a much clearer
service definition and a clear network model. Then you could argue much
more firmly for how well a given tool would work in this service.

For WebRTC it is about enable the application. So the question is on
what requirement level towards the implementation RTP Retransmission
shall be available on. Then it becomes a question for the application
implementors to use the available toolbox as best suiting their
application, media requirements and the current network characteristics.

When it comes to determine what functionality that is to be included in
WebRTC we are operating on a contributor driven model where people
provide proposals for what to include and the amount of motivation they
are willing to provide. Then people provide feedback on that based on
their knowledge. In some cases the proponents have to provide additional
information in other cases consensus is achieved with less.

In this particular case I requested that we continue the discussion from
the Interim meeting on the mailing list to see if more arguments are
raised that will sway people in either direction.


> 
> The only thing that I am pointing out that these types of simplistic
> emails are not good enough to convincing the technical communities
> unless all the technical details are presented with precise
> measurements. Both subjective and quantitative performance parameters
> are involved along the many subjects who remain in the test
> environments.

I was stating my position on this. I think I have a fairly good
understanding myself of when RTP retransmission works and when it is
less efficient. I see it as a very good tool. And the main obstacle is
when the RTT becomes to high. But also in cases with higher RTT > ~300
ms there are still tricks certain service will accept that gives them
much higher quality compared to not having retransmission.

Yes, I am bad at providing references to backing papers about this. I
draw my conviction from having seen or heard about working systems using
retransmission.

> 
> If anyone thinks that RTP retransmissions is a great tool, it is OK.
> Let them keep is private as many usages of RTP retransmissions were
> also discussed via emails.

Sorry, I don't understand what you want to say with the above paragraph.

Cheers

Magnus Westerlund

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




From ron.even.tlv@gmail.com  Thu Jun 28 06:53:22 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, <rtcweb@ietf.org>
References: <4FEAB80A.7040207@ericsson.com> <4FEC0C73.4030709@ericsson.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Hi Magnus,
So using retransmission is at the cost of losing lip synch which is very
noticeable. I am not saying that it does not work to do retransmission, =
my
claim that it is not practical and this is why I do not think it is
required.
Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 28 June, 2012 9:49 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
RECOMMENDED

Hi,

As Individual I like to state my position.

We have a video conference system developed by my colleagues used =
internally
at Ericsson that uses RTP Retransmission for video, not for audio with =
great
success. This is implemented such that we actually allow the video to =
fall
behind the audio when packet loss and retransmission is not able to =
repair
in a timely enough fashion. The benefit is minimal overhead and still no
loss induced degradations in the video. Yes, we get degradation in form =
of
frame display jittering and short freezes. But those events that are =
truly
visible are rare over wired networks.

I am personally convinced that RTP Retransmission is great tool in the
toolbox when it comes to improve media quality in many use cases. Yes =
there
are scenarios where RTP retransmission is less efficient. Long RTTs =
(over
200-400 ms) is the primary source of degradations. Compared to FEC it so
much more efficient from bandwidth consumption perspective.

I also think it is important that we have some mandatory to implement =
tool
for making the transport more robust now that we have a consensus that =
we
are not going for a FEC solution in the initial specification.

Thus my personal position is that RTP Retransmission should be REQUIRED =
to
implement.

Cheers

Magnus


On 2012-06-27 09:36, Magnus Westerlund wrote:
> WG,
>=20
> We had a discussion at the interim if RTP Retransmission is to be=20
> considered REQUIRED or RECOMMENDED to implement. I would like to see=20
> if we can first have some discussion on this topic before moving on to =

> see if we can get a consensus here on the mailing list.
>=20
> Please provide your views on this topic.
>=20
> Cheers
>=20
> Magnus Westerlund
> (As Chair and document editor)
>=20
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20


--=20

Magnus Westerlund

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



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From ron.even.tlv@gmail.com  Thu Jun 28 06:57:15 2012
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Roy, Radhika R CIV \(US\)'" <radhika.r.roy.civ@mail.mil>
References: <4FEAB80A.7040207@ericsson.com> <007c01cd54ff$207dee50$6179caf0$@gmail.com> <8486C8728176924BAF5BDB2F7D7EEDDF484E661A@ucolhp9d.easf.csd.disa.mil>
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Date: Thu, 28 Jun 2012 16:56:52 +0200
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Hi,
The question was asked if it should be required or recommended. From =
these
two I choose  recommended which is less strong even though I think that =
MAY
will be better for using retransmission in conversional applications.
Roni

-----Original Message-----
From: Roy, Radhika R CIV (US) [mailto:radhika.r.roy.civ@mail.mil]=20
Sent: 28 June, 2012 12:39 PM
To: Roni Even
Cc: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
RECOMMENDED

Hi, Roni:

Why do you want to recommend something that cannot be generalized
conversional applications just because of more tools can be sold in the
market?

No one is objecting to generalize the RTP retransmissions for the =
content
delivery or streaming applications.

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Roni Even
Sent: Thursday, June 28, 2012 3:25 AM
To: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
RECOMMENDED

Hi,
My view is that retransmission have great value for content delivery or
streaming applications, when the communication is unidirectional.=20
For conversional applications like we are defining in RTCweb, =
retransmission
has the lower value comparing to other loss recovery options like FEC or
receiver based recovery. The latency involved with the extra roundtrips
makes it not effective comparing to typical jitter buffer sizes at the
receiver, note that receiver will take correction actions even for late
arrival and will not try to dynamically change the jitter buffer to a =
larger
size beyond the maximum threshold of end to end delay.=20
 If we look at the typical behavior of conversional applications, they =
will
add to audio on the sender side FEC or even send the media twice is
subsequent packets encoded in different encoders,( of course for the
conversation itself the listener can always ask the talker to repeat =
since
he did not hear it clearly). On the video FEC and receiver side recovery
like motion estimation is common.
I propose that retransmission can be recommended and in my view for
conversional usage even this is strong.

Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
Magnus Westerlund
Sent: 27 June, 2012 9:37 AM
To: rtcweb@ietf.org
Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or =
RECOMMENDED

WG,

We had a discussion at the interim if RTP Retransmission is to be =
considered
REQUIRED or RECOMMENDED to implement. I would like to see if we can =
first
have some discussion on this topic before moving on to see if we can get =
a
consensus here on the mailing list.

Please provide your views on this topic.

Cheers

Magnus Westerlund
(As Chair and document editor)


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


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From radhika.r.roy.civ@mail.mil  Thu Jun 28 07:03:21 2012
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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Roni Even <ron.even.tlv@gmail.com>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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If this is your personal opinion, it is OK.

However, like many others, I would not recommend this with this kind of bla=
nket statement.

It is at all recommended, then the special circumstances under all those te=
chnical assumptions need to be noted. In this way, the RTP retransmission u=
sers will take their technically informed decisions and not to blame when p=
erils occur.

BR/Radhika

-----Original Message-----
From: Roni Even [mailto:ron.even.tlv@gmail.com]=20
Sent: Thursday, June 28, 2012 10:57 AM
To: Roy, Radhika R CIV (US)
Cc: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

Hi,
The question was asked if it should be required or recommended. From these =
two I choose  recommended which is less strong even though I think that MAY=
 will be better for using retransmission in conversional applications.
Roni

-----Original Message-----
From: Roy, Radhika R CIV (US) [mailto:radhika.r.roy.civ@mail.mil]
Sent: 28 June, 2012 12:39 PM
To: Roni Even
Cc: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

Hi, Roni:

Why do you want to recommend something that cannot be generalized conversio=
nal applications just because of more tools can be sold in the market?

No one is objecting to generalize the RTP retransmissions for the content d=
elivery or streaming applications.

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roni Even
Sent: Thursday, June 28, 2012 3:25 AM
To: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED

Hi,
My view is that retransmission have great value for content delivery or str=
eaming applications, when the communication is unidirectional.=20
For conversional applications like we are defining in RTCweb, retransmissio=
n has the lower value comparing to other loss recovery options like FEC or =
receiver based recovery. The latency involved with the extra roundtrips mak=
es it not effective comparing to typical jitter buffer sizes at the receive=
r, note that receiver will take correction actions even for late arrival an=
d will not try to dynamically change the jitter buffer to a larger size bey=
ond the maximum threshold of end to end delay.=20
 If we look at the typical behavior of conversional applications, they will=
 add to audio on the sender side FEC or even send the media twice is subseq=
uent packets encoded in different encoders,( of course for the conversation=
 itself the listener can always ask the talker to repeat since he did not h=
ear it clearly). On the video FEC and receiver side recovery like motion es=
timation is common.
I propose that retransmission can be recommended and in my view for convers=
ional usage even this is strong.

Roni

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Magnus Westerlund
Sent: 27 June, 2012 9:37 AM
To: rtcweb@ietf.org
Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED

WG,

We had a discussion at the interim if RTP Retransmission is to be considere=
d REQUIRED or RECOMMENDED to implement. I would like to see if we can first=
 have some discussion on this topic before moving on to see if we can get a=
 consensus here on the mailing list.

Please provide your views on this topic.

Cheers

Magnus Westerlund
(As Chair and document editor)


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


_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


From cb.list6@gmail.com  Thu Jun 28 07:04:41 2012
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--047d7b162db70dad4a04c388d04d
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On Jun 28, 2012 6:57 AM, "Roni Even" <ron.even.tlv@gmail.com> wrote:
>
> Hi,
> The question was asked if it should be required or recommended. From thes=
e
> two I choose  recommended which is less strong even though I think that
MAY
> will be better for using retransmission in conversional applications.
> Roni
>

+1 for MAY.

CB

> -----Original Message-----
> From: Roy, Radhika R CIV (US) [mailto:radhika.r.roy.civ@mail.mil]
> Sent: 28 June, 2012 12:39 PM
> To: Roni Even
> Cc: 'Magnus Westerlund'; rtcweb@ietf.org
> Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
> RECOMMENDED
>
> Hi, Roni:
>
> Why do you want to recommend something that cannot be generalized
> conversional applications just because of more tools can be sold in the
> market?
>
> No one is objecting to generalize the RTP retransmissions for the content
> delivery or streaming applications.
>
> BR/Radhika
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of
> Roni Even
> Sent: Thursday, June 28, 2012 3:25 AM
> To: 'Magnus Westerlund'; rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
> RECOMMENDED
>
> Hi,
> My view is that retransmission have great value for content delivery or
> streaming applications, when the communication is unidirectional.
> For conversional applications like we are defining in RTCweb,
retransmission
> has the lower value comparing to other loss recovery options like FEC or
> receiver based recovery. The latency involved with the extra roundtrips
> makes it not effective comparing to typical jitter buffer sizes at the
> receiver, note that receiver will take correction actions even for late
> arrival and will not try to dynamically change the jitter buffer to a
larger
> size beyond the maximum threshold of end to end delay.
>  If we look at the typical behavior of conversional applications, they
will
> add to audio on the sender side FEC or even send the media twice is
> subsequent packets encoded in different encoders,( of course for the
> conversation itself the listener can always ask the talker to repeat sinc=
e
> he did not hear it clearly). On the video FEC and receiver side recovery
> like motion estimation is common.
> I propose that retransmission can be recommended and in my view for
> conversional usage even this is strong.
>
> Roni
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of
> Magnus Westerlund
> Sent: 27 June, 2012 9:37 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDE=
D
>
> WG,
>
> We had a discussion at the interim if RTP Retransmission is to be
considered
> REQUIRED or RECOMMENDED to implement. I would like to see if we can first
> have some discussion on this topic before moving on to see if we can get =
a
> consensus here on the mailing list.
>
> Please provide your views on this topic.
>
> Cheers
>
> Magnus Westerlund
> (As Chair and document editor)
>
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--047d7b162db70dad4a04c388d04d
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<p><br>
On Jun 28, 2012 6:57 AM, &quot;Roni Even&quot; &lt;<a href=3D"mailto:ron.ev=
en.tlv@gmail.com">ron.even.tlv@gmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Hi,<br>
&gt; The question was asked if it should be required or recommended. From t=
hese<br>
&gt; two I choose =A0recommended which is less strong even though I think t=
hat MAY<br>
&gt; will be better for using retransmission in conversional applications.<=
br>
&gt; Roni<br>
&gt;</p>
<p>+1 for MAY.</p>
<p>CB</p>
<p>&gt; -----Original Message-----<br>
&gt; From: Roy, Radhika R CIV (US) [mailto:<a href=3D"mailto:radhika.r.roy.=
civ@mail.mil">radhika.r.roy.civ@mail.mil</a>]<br>
&gt; Sent: 28 June, 2012 12:39 PM<br>
&gt; To: Roni Even<br>
&gt; Cc: &#39;Magnus Westerlund&#39;; <a href=3D"mailto:rtcweb@ietf.org">rt=
cweb@ietf.org</a><br>
&gt; Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or<br>
&gt; RECOMMENDED<br>
&gt;<br>
&gt; Hi, Roni:<br>
&gt;<br>
&gt; Why do you want to recommend something that cannot be generalized<br>
&gt; conversional applications just because of more tools can be sold in th=
e<br>
&gt; market?<br>
&gt;<br>
&gt; No one is objecting to generalize the RTP retransmissions for the cont=
ent<br>
&gt; delivery or streaming applications.<br>
&gt;<br>
&gt; BR/Radhika<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ie=
tf.org</a>] On Behalf Of<br>
&gt; Roni Even<br>
&gt; Sent: Thursday, June 28, 2012 3:25 AM<br>
&gt; To: &#39;Magnus Westerlund&#39;; <a href=3D"mailto:rtcweb@ietf.org">rt=
cweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or<br>
&gt; RECOMMENDED<br>
&gt;<br>
&gt; Hi,<br>
&gt; My view is that retransmission have great value for content delivery o=
r<br>
&gt; streaming applications, when the communication is unidirectional.<br>
&gt; For conversional applications like we are defining in RTCweb, retransm=
ission<br>
&gt; has the lower value comparing to other loss recovery options like FEC =
or<br>
&gt; receiver based recovery. The latency involved with the extra roundtrip=
s<br>
&gt; makes it not effective comparing to typical jitter buffer sizes at the=
<br>
&gt; receiver, note that receiver will take correction actions even for lat=
e<br>
&gt; arrival and will not try to dynamically change the jitter buffer to a =
larger<br>
&gt; size beyond the maximum threshold of end to end delay.<br>
&gt; =A0If we look at the typical behavior of conversional applications, th=
ey will<br>
&gt; add to audio on the sender side FEC or even send the media twice is<br=
>
&gt; subsequent packets encoded in different encoders,( of course for the<b=
r>
&gt; conversation itself the listener can always ask the talker to repeat s=
ince<br>
&gt; he did not hear it clearly). On the video FEC and receiver side recove=
ry<br>
&gt; like motion estimation is common.<br>
&gt; I propose that retransmission can be recommended and in my view for<br=
>
&gt; conversional usage even this is strong.<br>
&gt;<br>
&gt; Roni<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ie=
tf.org</a>] On Behalf Of<br>
&gt; Magnus Westerlund<br>
&gt; Sent: 27 June, 2012 9:37 AM<br>
&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMME=
NDED<br>
&gt;<br>
&gt; WG,<br>
&gt;<br>
&gt; We had a discussion at the interim if RTP Retransmission is to be cons=
idered<br>
&gt; REQUIRED or RECOMMENDED to implement. I would like to see if we can fi=
rst<br>
&gt; have some discussion on this topic before moving on to see if we can g=
et a<br>
&gt; consensus here on the mailing list.<br>
&gt;<br>
&gt; Please provide your views on this topic.<br>
&gt;<br>
&gt; Cheers<br>
&gt;<br>
&gt; Magnus Westerlund<br>
&gt; (As Chair and document editor)<br>
&gt;<br>
&gt;<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt; ----------------------------------------------------------------------=
<br>
&gt; Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0+46 10 7148287<b=
r>
&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile +46 73 0949079=
<br>
&gt; SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlu=
nd@ericsson.com">magnus.westerlund@ericsson.com</a><br>
&gt; ----------------------------------------------------------------------=
<br>
&gt;<br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; rtcweb mailing list<br>
&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--047d7b162db70dad4a04c388d04d--

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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Cameron Byrne <cb.list6@gmail.com>, Roni Even <ron.even.tlv@gmail.com>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Yes, like CB, I will also support "MAY."

BR/Radhika


-----Original Message-----
From: Cameron Byrne [mailto:cb.list6@gmail.com]=20
Sent: Thursday, June 28, 2012 10:05 AM
To: Roni Even
Cc: rtcweb@ietf.org; Roy, Radhika R CIV (US)
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED


On Jun 28, 2012 6:57 AM, "Roni Even" <ron.even.tlv@gmail.com> wrote:
>
> Hi,
> The question was asked if it should be required or recommended. From=20
> these two I choose  recommended which is less strong even though I=20
> think that MAY will be better for using retransmission in conversional ap=
plications.
> Roni
>

+1 for MAY.

CB

> -----Original Message-----
> From: Roy, Radhika R CIV (US) [mailto:radhika.r.roy.civ@mail.mil]
> Sent: 28 June, 2012 12:39 PM
> To: Roni Even
> Cc: 'Magnus Westerlund'; rtcweb@ietf.org
> Subject: RE: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or=20
> RECOMMENDED
>
> Hi, Roni:
>
> Why do you want to recommend something that cannot be generalized=20
> conversional applications just because of more tools can be sold in=20
> the market?
>
> No one is objecting to generalize the RTP retransmissions for the=20
> content delivery or streaming applications.
>
> BR/Radhika
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Roni Even
> Sent: Thursday, June 28, 2012 3:25 AM
> To: 'Magnus Westerlund'; rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or=20
> RECOMMENDED
>
> Hi,
> My view is that retransmission have great value for content delivery=20
> or streaming applications, when the communication is unidirectional.
> For conversional applications like we are defining in RTCweb,=20
> retransmission has the lower value comparing to other loss recovery=20
> options like FEC or receiver based recovery. The latency involved with=20
> the extra roundtrips makes it not effective comparing to typical=20
> jitter buffer sizes at the receiver, note that receiver will take=20
> correction actions even for late arrival and will not try to=20
> dynamically change the jitter buffer to a larger size beyond the maximum =
threshold of end to end delay.
>  If we look at the typical behavior of conversional applications, they=20
> will add to audio on the sender side FEC or even send the media twice=20
> is subsequent packets encoded in different encoders,( of course for=20
> the conversation itself the listener can always ask the talker to=20
> repeat since he did not hear it clearly). On the video FEC and=20
> receiver side recovery like motion estimation is common.
> I propose that retransmission can be recommended and in my view for=20
> conversional usage even this is strong.
>
> Roni
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Magnus Westerlund
> Sent: 27 June, 2012 9:37 AM
> To: rtcweb@ietf.org
> Subject: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or=20
> RECOMMENDED
>
> WG,
>
> We had a discussion at the interim if RTP Retransmission is to be=20
> considered REQUIRED or RECOMMENDED to implement. I would like to see=20
> if we can first have some discussion on this topic before moving on to=20
> see if we can get a consensus here on the mailing list.
>
> Please provide your views on this topic.
>
> Cheers
>
> Magnus Westerlund
> (As Chair and document editor)
>
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



From csp@csperkins.org  Thu Jun 28 07:14:57 2012
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Roni,

I don't understand how you can say this is not practical. Magnus has an =
implementation, and he states it works well. That seems practical to me.

There are also self-evidently network paths with low enough RTT for RTP =
retransmission to be used for interactive calls, if the playout buffer =
code knows the RTT and adjusts the buffer to allow this. Not all paths, =
I agree, but that doesn't mean it's not useful in some environments.=20

We're defining requirements for the RTP infrastructure in WebRTC clients =
here, not forcing all implementations to use every feature. My personal =
view is that retransmission should be REQUIRED. Given that the RTP/AVPF =
profile is REQUIRED, implementation of retransmission is =
straightforward. I see no benefit in leaving it out, and very little =
cost in requiring it.

Colin


On 28 Jun 2012, at 15:52, Roni Even wrote:
> Hi Magnus,
> So using retransmission is at the cost of losing lip synch which is =
very noticeable. I am not saying that it does not work to do =
retransmission, my claim that it is not practical and this is why I do =
not think it is required.
> Roni
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of
> Magnus Westerlund
> Sent: 28 June, 2012 9:49 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
> RECOMMENDED
>=20
> Hi,
>=20
> As Individual I like to state my position.
>=20
> We have a video conference system developed by my colleagues used =
internally at Ericsson that uses RTP Retransmission for video, not for =
audio with great success. This is implemented such that we actually =
allow the video to fall behind the audio when packet loss and =
retransmission is not able to repair in a timely enough fashion. The =
benefit is minimal overhead and still no loss induced degradations in =
the video. Yes, we get degradation in form of frame display jittering =
and short freezes. But those events that are truly visible are rare over =
wired networks.
>=20
> I am personally convinced that RTP Retransmission is great tool in the =
toolbox when it comes to improve media quality in many use cases. Yes =
there are scenarios where RTP retransmission is less efficient. Long =
RTTs (over 200-400 ms) is the primary source of degradations. Compared =
to FEC it so much more efficient from bandwidth consumption perspective.
>=20
> I also think it is important that we have some mandatory to implement =
tool for making the transport more robust now that we have a consensus =
that we are not going for a FEC solution in the initial specification.
>=20
> Thus my personal position is that RTP Retransmission should be =
REQUIRED to implement.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> On 2012-06-27 09:36, Magnus Westerlund wrote:
>> WG,
>>=20
>> We had a discussion at the interim if RTP Retransmission is to be=20
>> considered REQUIRED or RECOMMENDED to implement. I would like to see=20=

>> if we can first have some discussion on this topic before moving on =
to=20
>> see if we can get a consensus here on the mailing list.
>>=20
>> Please provide your views on this topic.
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>> (As Chair and document editor)
>>=20
>>=20
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




From magnus.westerlund@ericsson.com  Thu Jun 28 07:17:03 2012
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 2012-06-28 16:52, Roni Even wrote:
> Hi Magnus,
> So using retransmission is at the cost of losing lip synch which is very
> noticeable. I am not saying that it does not work to do retransmission, my
> claim that it is not practical and this is why I do not think it is
> required.

My experience with using this system several times a week for the last
two years says that it is not that noticable. The type of video image
degradations you get from unrepaired burst losses are so much worse.

I also think it is important that we consider the goals. If your goal is
a really high quality telepresence system then this approach is most
likely not usable. But, most WebRTC applications doesn't have that
quality requirements nor the user expectancies to be significantly
annoyed at a short loss of lip sync. The bit-rate savings that RTP
retransmission provides compared to FEC is much more important in my mind.

I think the Google hangout session during the interim showed the
importance of having some video being much higher than having no video
or requiring video with perfect sync with audio. And Justin said, Google
Hangout uses Retransmission.

Cheers

Magnus


> Roni
> 
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
> Magnus Westerlund
> Sent: 28 June, 2012 9:49 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
> RECOMMENDED
> 
> Hi,
> 
> As Individual I like to state my position.
> 
> We have a video conference system developed by my colleagues used internally
> at Ericsson that uses RTP Retransmission for video, not for audio with great
> success. This is implemented such that we actually allow the video to fall
> behind the audio when packet loss and retransmission is not able to repair
> in a timely enough fashion. The benefit is minimal overhead and still no
> loss induced degradations in the video. Yes, we get degradation in form of
> frame display jittering and short freezes. But those events that are truly
> visible are rare over wired networks.
> 
> I am personally convinced that RTP Retransmission is great tool in the
> toolbox when it comes to improve media quality in many use cases. Yes there
> are scenarios where RTP retransmission is less efficient. Long RTTs (over
> 200-400 ms) is the primary source of degradations. Compared to FEC it so
> much more efficient from bandwidth consumption perspective.
> 
> I also think it is important that we have some mandatory to implement tool
> for making the transport more robust now that we have a consensus that we
> are not going for a FEC solution in the initial specification.
> 
> Thus my personal position is that RTP Retransmission should be REQUIRED to
> implement.
> 
> Cheers
> 
> Magnus
> 
> 
> On 2012-06-27 09:36, Magnus Westerlund wrote:
>> WG,
>>
>> We had a discussion at the interim if RTP Retransmission is to be 
>> considered REQUIRED or RECOMMENDED to implement. I would like to see 
>> if we can first have some discussion on this topic before moving on to 
>> see if we can get a consensus here on the mailing list.
>>
>> Please provide your views on this topic.
>>
>> Cheers
>>
>> Magnus Westerlund
>> (As Chair and document editor)
>>
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
> 
> 


-- 

Magnus Westerlund

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




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I think you need to separate this for audio and video and be far more =
specific about what type of retransmit ion you are talking about. In =
many cases the retransmit ion schemes for audio make the end suer =
experience worse.=20

On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:

> WG,
>=20
> We had a discussion at the interim if RTP Retransmission is to be
> considered REQUIRED or RECOMMENDED to implement. I would like to see =
if
> we can first have some discussion on this topic before moving on to =
see
> if we can get a consensus here on the mailing list.
>=20
> Please provide your views on this topic.
>=20
> Cheers
>=20
> Magnus Westerlund
> (As Chair and document editor)
>=20
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From fluffy@iii.ca  Thu Jun 28 10:58:53 2012
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Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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Accepting persistent consent over a HTTP connection seems like a really =
really bad idea to me. Among other things, I would hope the IESG would =
just say no to this.=20

On Jun 26, 2012, at 15:59 , Eric Rescorla wrote:

> In Martin's review of draft-ietf-rtcweb-security-arch, he (re)raised
> the question of persistent consent for camera/microphone access from
> pages served over HTTP.  I had thought this issue closed, but in the
> discussion at the interim there seemed to be a fair amount of
> sentiment that people hadn't understood the security issues and now
> believed that this was a bad idea. I agreed to attempt to re-start the
> discussion with a clearer description of what the problem was, hence
> this message.
>=20
> Here's the basic attack:
> Say that I have given persistent permission to access my camera and
> microphone to any page from http://www.example.com/. Now, I go to an
> Internet cafe and surf to *any* HTTP page, e.g.,
> http://www.google.com/. At this point, any attacker who controls the
> wireless network can redirect that access to point to
> http://www.example.com/ and inject JS that reads from my camera and
> microphone.
>=20
> That's pretty bad, but made worse by the following facts:
> 1. It's not that hard for the attacker to open up
> popups/popunders/iframes, etc. that are in the domain of example.com
> but actually execute code from the attacker.  So, once you've opened
> the browser in a hostile environment, the attacker can bug you at
> times of his choosing until you close your browser.
>=20
> 2. If your browser does any sort of auto-refresh (e.g., because
> the page refreshes itself) then you can get infected even if you
> don't do anything deliberate.
>=20
> 3. This also applies if you are using any kind of open wireless
> network, such as if your home network doesn't use a strong
> enough WPA key.
>=20
>=20
> So, the executive summary is: if you have a persistent permission
> for an HTTP site, and you ever use your browser on any insecure
> network, an attacker can bug your computer until you close your
> browser. This seems bad.
>=20
> Does this change people's opinions about what the rules should be
> about whether browsers permit persistent HTTP permissions?
>=20
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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I'd like to discuss setting DSCP values and plan to have a draft out =
before the deadline. I know that it's hard to evaluate any interest in =
discussing this before the WG sees the draft but consider this a heads =
up of something that might want some time.=20

On Jun 26, 2012, at 23:41 , Magnus Westerlund wrote:

> WG,
>=20
> The WG chairs solicit topics that you think needs to be discussed in
> Vancouver. Both authors/editors of documents that have topics they =
think
> requires face to face discussion to reach consensus and WG =
participants
> that think a particular topic should be discussed are of interest.
>=20
> Two topics that is clear they need further discussion and was not =
taken
> at the interim meeting are
> - Additional Key-management for SRTP
> - Selection of Mandatory to implement Media Codecs
>=20
> Please add to the list or comment on the suitability to have these =
topics.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Colin & Magnus, given this is starting to look like it is likely to get =
consensus, could you update draft-ietf-rtcweb-rtp-usage to have text for =
random generation and resonating behind it?




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On 28 Jun 2012, at 15:53, Cullen Jennings wrote:
> Colin & Magnus, given this is starting to look like it is likely to =
get consensus, could you update draft-ietf-rtcweb-rtp-usage to have text =
for random generation and resonating behind it?


As I said, write a draft, and we'll reference it. I don't think this =
draft is the place to update the RTP specification.

--=20
Colin Perkins
http://csperkins.org/




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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: Target numbers for setup time (Re: Keeping up data channel)
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Subject: Re: [rtcweb] Target numbers for setup time (Re: Keeping up data channel)
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Agree - that's why I think this has to be done in zero RTT. I don't see any=
 problem with setting up a data channel and being ready to send media befor=
e having the UI tell the user they are "connected and can start talking".

Reducing from 7 RTT to 5 RTT does not help when you need to get to 0 RTT.

On Jun 22, 2012, at 10:09 , Harald Alvestrand wrote:

> On 06/22/2012 05:58 PM, Cullen Jennings (fluffy) wrote:
>> On Jun 13, 2012, at 5:29 , Randell Jesup wrote:
>>=20
>>>> How far down do you think we have to drive the setup time before you
>>>> would not call it "abysmal"?
>>=20
>> I'd probably consider above 250 ms abysmal but good news I don't see any=
 problem with getting it down around 100 ms in when both endpoints are in a=
 single country.
>>=20
> Coast-to-coast US is ~4800 km, so RTT (9600 km) is 32 ms (speed of light =
is 300 km/msec).
>=20
> So, without considering processing time, 3 RTT is 100 msec, 7 RTT is "aby=
smal".
>=20
> There are bigger countries than the US, but this will do for a back-of-th=
e-envelope.
>=20
>=20
>>=20
>=20
>=20


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Dan Burnett <dburnett@voxeo.com>
Thread-Topic: [rtcweb] New draft-burnett-rtcweb-constraints-registry-00
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I believe the answer is that informational would be fine for this. Lets go =
down the informational path unless someone has a really good reason not to.=
=20

Cullen <with my WG co-chair hat on>=20

On Jun 13, 2012, at 5:10 , Dan Burnett wrote:

> I defer to Cullen (as a chair of this group) on this issue.  I don't pers=
onally care one way or the other.
>=20
> -- dan
>=20
> On Mar 2, 2012, at 8:14 PM, DRAGE, Keith (Keith) wrote:
>=20
>> Pedantic process issue.
>>=20
>> Do we need a proposed standard to create an IANA registry which only nee=
ds expert review to add a value, or will informational suffice?
>>=20
>> Keith
>>=20
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behal=
f
>>> Of Dan Burnett
>>> Sent: 02 March 2012 21:18
>>> To: <rtcweb@ietf.org>
>>> Subject: [rtcweb] New draft-burnett-rtcweb-constraints-registry-00
>>>=20
>>> Group,
>>>=20
>>> I have just submitted a new draft [1] to create a registry for HTML Med=
ia
>>> constraints for use by the W3C getusermedia and WebRTC specs.  An early
>>> email proposal on this subject (not the registry, but the use of it) is
>>> available [2] for background.  I will be updating that with a fuller
>>> proposal that incorporates changes discussed on the Media Capture Task
>>> Force teleconference [3], so don't be surprised if you notice slight
>>> differences in the syntax in the registry from what was proposed in ema=
il
>>> [2].
>>>=20
>>> I don't know how much discussion will happen on this list for this draf=
t,
>>> but at the least this group should be aware of the W3C need for it.
>>>=20
>>> -- dan
>>>=20
>>> [1] https://datatracker.ietf.org/doc/draft-burnett-rtcweb-constraints-
>>> registry/
>>> [2] http://lists.w3.org/Archives/Public/public-webrtc/2012Feb/0198.html
>>> [3] http://lists.w3.org/Archives/Public/public-media-capture/2012Feb/at=
t-
>>> 0053/minutes-2012-02-28.html
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Colin Perkins'" <csp@csperkins.org>
References: <4FEAB80A.7040207@ericsson.com> <4FEC0C73.4030709@ericsson.com> <00af01cd553d$b72ce070$2586a150$@gmail.com> <451142FF-AB75-4C87-9936-30DDC212FBC8@csperkins.org>
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Date: Fri, 29 Jun 2012 00:17:50 +0200
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or	RECOMMENDED
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Colin,
Magnus also said that the penalty was losing lip synch, this is why it =
is
not practical. His solution is for proof of work which I agree it is =
working
but in my view not relevant for real usage. In order to use it you may =
need
a shorter jitter buffer and make a decision to ask for retransmit early, =
my
view is that larger jitter or playout buffer for late arrival will work
better.
So if there are some network path with short RTT (I assume not the =
Internet)
they can use retransmission, so recommended will be enough with text =
explain
when it make sense to support it.
Roni


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]=20
Sent: 28 June, 2012 4:15 PM
To: Roni Even
Cc: 'Magnus Westerlund'; rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
RECOMMENDED

Roni,

I don't understand how you can say this is not practical. Magnus has an
implementation, and he states it works well. That seems practical to me.

There are also self-evidently network paths with low enough RTT for RTP
retransmission to be used for interactive calls, if the playout buffer =
code
knows the RTT and adjusts the buffer to allow this. Not all paths, I =
agree,
but that doesn't mean it's not useful in some environments.=20

We're defining requirements for the RTP infrastructure in WebRTC clients
here, not forcing all implementations to use every feature. My personal =
view
is that retransmission should be REQUIRED. Given that the RTP/AVPF =
profile
is REQUIRED, implementation of retransmission is straightforward. I see =
no
benefit in leaving it out, and very little cost in requiring it.

Colin


On 28 Jun 2012, at 15:52, Roni Even wrote:
> Hi Magnus,
> So using retransmission is at the cost of losing lip synch which is =
very
noticeable. I am not saying that it does not work to do retransmission, =
my
claim that it is not practical and this is why I do not think it is
required.
> Roni
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Magnus Westerlund
> Sent: 28 June, 2012 9:49 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or=20
> RECOMMENDED
>=20
> Hi,
>=20
> As Individual I like to state my position.
>=20
> We have a video conference system developed by my colleagues used
internally at Ericsson that uses RTP Retransmission for video, not for =
audio
with great success. This is implemented such that we actually allow the
video to fall behind the audio when packet loss and retransmission is =
not
able to repair in a timely enough fashion. The benefit is minimal =
overhead
and still no loss induced degradations in the video. Yes, we get =
degradation
in form of frame display jittering and short freezes. But those events =
that
are truly visible are rare over wired networks.
>=20
> I am personally convinced that RTP Retransmission is great tool in the
toolbox when it comes to improve media quality in many use cases. Yes =
there
are scenarios where RTP retransmission is less efficient. Long RTTs =
(over
200-400 ms) is the primary source of degradations. Compared to FEC it so
much more efficient from bandwidth consumption perspective.
>=20
> I also think it is important that we have some mandatory to implement =
tool
for making the transport more robust now that we have a consensus that =
we
are not going for a FEC solution in the initial specification.
>=20
> Thus my personal position is that RTP Retransmission should be =
REQUIRED to
implement.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> On 2012-06-27 09:36, Magnus Westerlund wrote:
>> WG,
>>=20
>> We had a discussion at the interim if RTP Retransmission is to be=20
>> considered REQUIRED or RECOMMENDED to implement. I would like to see=20
>> if we can first have some discussion on this topic before moving on=20
>> to see if we can get a consensus here on the mailing list.
>>=20
>> Please provide your views on this topic.
>>=20
>> Cheers
>>=20
>> Magnus Westerlund
>> (As Chair and document editor)
>>=20
>>=20
>> ---------------------------------------------------------------------
>> - Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ---------------------------------------------------------------------
>> -
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--
Colin Perkins
http://csperkins.org/




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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
References: <4FEAB80A.7040207@ericsson.com> <4FEC0C73.4030709@ericsson.com> <00af01cd553d$b72ce070$2586a150$@gmail.com> <4FEC675B.8080002@ericsson.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Hi Magnus,
The demo in the interim of what happens in retransmission was not in par
with my experience with any other recovery  methods including receiver =
based
repair without FEC.
Roni

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: 28 June, 2012 4:17 PM
To: Roni Even
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
RECOMMENDED

On 2012-06-28 16:52, Roni Even wrote:
> Hi Magnus,
> So using retransmission is at the cost of losing lip synch which is=20
> very noticeable. I am not saying that it does not work to do=20
> retransmission, my claim that it is not practical and this is why I do =

> not think it is required.

My experience with using this system several times a week for the last =
two
years says that it is not that noticable. The type of video image
degradations you get from unrepaired burst losses are so much worse.

I also think it is important that we consider the goals. If your goal is =
a
really high quality telepresence system then this approach is most =
likely
not usable. But, most WebRTC applications doesn't have that quality
requirements nor the user expectancies to be significantly annoyed at a
short loss of lip sync. The bit-rate savings that RTP retransmission
provides compared to FEC is much more important in my mind.

I think the Google hangout session during the interim showed the =
importance
of having some video being much higher than having no video or requiring
video with perfect sync with audio. And Justin said, Google Hangout uses
Retransmission.

Cheers

Magnus


> Roni
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Magnus Westerlund
> Sent: 28 June, 2012 9:49 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or=20
> RECOMMENDED
>=20
> Hi,
>=20
> As Individual I like to state my position.
>=20
> We have a video conference system developed by my colleagues used=20
> internally at Ericsson that uses RTP Retransmission for video, not for =

> audio with great success. This is implemented such that we actually=20
> allow the video to fall behind the audio when packet loss and=20
> retransmission is not able to repair in a timely enough fashion. The=20
> benefit is minimal overhead and still no loss induced degradations in=20
> the video. Yes, we get degradation in form of frame display jittering=20
> and short freezes. But those events that are truly visible are rare =
over
wired networks.
>=20
> I am personally convinced that RTP Retransmission is great tool in the =

> toolbox when it comes to improve media quality in many use cases. Yes=20
> there are scenarios where RTP retransmission is less efficient. Long=20
> RTTs (over
> 200-400 ms) is the primary source of degradations. Compared to FEC it=20
> so much more efficient from bandwidth consumption perspective.
>=20
> I also think it is important that we have some mandatory to implement=20
> tool for making the transport more robust now that we have a consensus =

> that we are not going for a FEC solution in the initial specification.
>=20
> Thus my personal position is that RTP Retransmission should be=20
> REQUIRED to implement.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> On 2012-06-27 09:36, Magnus Westerlund wrote:
>> WG,
>>
>> We had a discussion at the interim if RTP Retransmission is to be=20
>> considered REQUIRED or RECOMMENDED to implement. I would like to see=20
>> if we can first have some discussion on this topic before moving on=20
>> to see if we can get a consensus here on the mailing list.
>>
>> Please provide your views on this topic.
>>
>> Cheers
>>
>> Magnus Westerlund
>> (As Chair and document editor)
>>
>>
>> ---------------------------------------------------------------------
>> - Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ---------------------------------------------------------------------
>> -
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>=20
>=20


--=20

Magnus Westerlund

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




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From: =?iso-8859-1?Q?G=F6ran_Eriksson_AP?= <goran.ap.eriksson@ericsson.com>
To: Cullen Jennings <fluffy@iii.ca>
Date: Thu, 28 Jun 2012 23:39:35 +0200
Thread-Topic: [rtcweb] Why persistent consent for HTTP is a problem
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Why persistent consent for HTTP is a problem
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+1

Sent from my iPad

On 28 jun 2012, at 19:58, "Cullen Jennings" <fluffy@iii.ca> wrote:

>=20
> Accepting persistent consent over a HTTP connection seems like a really r=
eally bad idea to me. Among other things, I would hope the IESG would just =
say no to this.=20
>=20
> On Jun 26, 2012, at 15:59 , Eric Rescorla wrote:
>=20
>> In Martin's review of draft-ietf-rtcweb-security-arch, he (re)raised
>> the question of persistent consent for camera/microphone access from
>> pages served over HTTP.  I had thought this issue closed, but in the
>> discussion at the interim there seemed to be a fair amount of
>> sentiment that people hadn't understood the security issues and now
>> believed that this was a bad idea. I agreed to attempt to re-start the
>> discussion with a clearer description of what the problem was, hence
>> this message.
>>=20
>> Here's the basic attack:
>> Say that I have given persistent permission to access my camera and
>> microphone to any page from http://www.example.com/. Now, I go to an
>> Internet cafe and surf to *any* HTTP page, e.g.,
>> http://www.google.com/. At this point, any attacker who controls the
>> wireless network can redirect that access to point to
>> http://www.example.com/ and inject JS that reads from my camera and
>> microphone.
>>=20
>> That's pretty bad, but made worse by the following facts:
>> 1. It's not that hard for the attacker to open up
>> popups/popunders/iframes, etc. that are in the domain of example.com
>> but actually execute code from the attacker.  So, once you've opened
>> the browser in a hostile environment, the attacker can bug you at
>> times of his choosing until you close your browser.
>>=20
>> 2. If your browser does any sort of auto-refresh (e.g., because
>> the page refreshes itself) then you can get infected even if you
>> don't do anything deliberate.
>>=20
>> 3. This also applies if you are using any kind of open wireless
>> network, such as if your home network doesn't use a strong
>> enough WPA key.
>>=20
>>=20
>> So, the executive summary is: if you have a persistent permission
>> for an HTTP site, and you ever use your browser on any insecure
>> network, an attacker can bug your computer until you close your
>> browser. This seems bad.
>>=20
>> Does this change people's opinions about what the rules should be
>> about whether browsers permit persistent HTTP permissions?
>>=20
>> -Ekr
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From anne.raby@ericsson.com  Thu Jun 28 17:45:35 2012
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From: Anne Raby <anne.raby@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 28 Jun 2012 20:45:29 -0400
Thread-Topic: JSEP-01 Handling Early Media, using SIP, clarifications
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Hi,

Concerning the SIP Early Media example in draft-ietf-rtcweb-jsep-01 I had s=
ome interrogations about the usage of the "pranswer" state when the Offerer=
 receives the answer in a 180 response from the Anserer[1]. What is the adv=
antage of using the "pranswer" state over the "answer"?

Copy of the example:

A.2.6. Handling early media (e.g. 1-800-FEDEX), using SIP

   This example demonstrates how early media could be handled; for
   simplicity, only the offerer side of the call is shown.

   // Call is initiated toward Answerer
   OffererJS->OffererUA:   pc =3D new PeerConnection();
   OffererJS->OffererUA:   pc.addStream(localStream, null);
   OffererUA->OffererJS:   onicecandidate(candidate);
   OffererJS->OffererUA:   offer =3D pc.createOffer(null);
   OffererJS->OffererUA:   pc.setLocalDescription("offer", offer);
   OffererJS:              sip =3D createInvite(offer);
   OffererJS->AnswererJS:  SIP INVITE w/ SDP

   // 180 Ringing is received by offerer, w/ SDP
   OffererJS:              answer =3D parseResponse(sip);
1->OffererJS->OffererUA:   pc.setRemoteDescription("pranswer", answer);
   OffererUA->OffererJS:   onaddstream(remoteStream);

   // ICE Completes (at Offerer)
   OffererUA->OffererJS:   onopen();
2->OffererUA->AnswererUA:  Media

   // 200 OK arrives at Offerer
3->OffererJS:              answer =3D parseResponse(sip);
   OffererJS->OffererUA:   pc.setRemoteDescription("answer", answer);
   OffererJS->AnswererJS:  ACK



I am assuming the example is using Reliable Provisional Response since the =
Media seems to be flowing from Offerer to Answerer before the 200 OK and AC=
K [2]. Below, I depict my understanding of the associated SIP:

   UAC(offerer)          UAS(answerer)
    | F1  INVITE (SDP)    |
    |-------------------->|
    | F2 180-rel (SDP)    |
    |<--------------------|
   =20
    | F3   PRACK (no SDP) |
    |-------------------->|
    | F4 200 PRA (no SDP) |
    |<--------------------|
    |                     |
    | F5  200 INV (SDP)   |
    |<--------------------|
    | F6     ACK          |
    |-------------------->|
 =20
I am trying to reconcile this example with RFC3261, RFC3262 and RFC3264. Wi=
th the help of RFC 6337, I understand that if a UAC receives a SDP answer i=
n a reliable response, it should ignore any SDP ANSWER in a subsequent resp=
onse in the same INVITE transaction. If the 2nd SDP is different then the U=
AC should only consider the 1st SDP answer.

If this is the case, then the SDP answer from the 200 OK should be the same=
 as in the 180. What is then the advantage of using the "pranswer" state? I=
 am trying to figure out in which case the "pranswer" could be used?

Thanks for helping me better understand this JSEP SIP early media example.


Best regards,

Anne

-----------------------------------------
ANNE RABY=20
-----------------------------------------
Ericsson Canada
Business Unit Networks, PDU Converged Core, H2S
8500 D=E9carie
Mont-Royal, Qu=E9bec, H4P 2N2, Canada
Phone +1 514 345-7900 x16685
Mobile +1 514 502-3073
anne.raby@ericsson.com
www.ericsson.com

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On 06/29/2012 03:31 AM, éçµè/denglingli wrote:
> Hi, Stefan
>
> Have the group ever considered adding use-cases for on-line presence
> notification for subscribed friends or user-defined groups?
> They are basic features of RTC services today I believe, and surely would
> promote users' on-line communication practice.

Hi Lingli,

what has been discussed is to add a use-case where _identity_ handling 
of peers communicating is important (e.g. allow the browser, without 
involvement of the application, to verify the identity of the peer using 
an Identity Provider).

I don't think adding on-line presence, notification, group handling has 
been discussed for addition to the document- I agree fully that these 
are important features of RTC services, but the reasoning (this is my 
personal understanding - others may disagree) has been that exactly 
these features are already supported by numerous services (I won't name 
them here, but you know what I'm thinking of) that use the browser as 
client environment (at least for desktop). So this seems like a problem 
solved, what is not yet solved (and what this group is addressing) is 
peer-to-peer audiovisual (and data) communication (without plug-ins).

Br,
Stefan

>
> BR
> Lingli
>
> -----é®ä»¶åä»¶-----
> åä»¶äºº: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] ä»£è¡š Stefan
> Hakansson LK
> åéæ¶éŽ: 2012å¹Ž6æ27æ¥ 20:43
> æ¶ä»¶äºº: rtcweb@ietf.org
> äž»é¢: [rtcweb] Use-case draft updated
>
> Hi,
>
> as already automatically announced, the use-case draft has been updated.
> Extract from the change log:
>
> * Changed "eavesdropping" to "wiretapping" and referenced RFC2804.
>
> * Removed informal ref webrtc_req; that document has been abandoned by the
> W3C webrtc WG.
>
> * Added use-case where one user is behind a FW that only allows http;
> derived req. F37.
> 					
> * Changed F24 slightly; MUST-> SHOULD, inserted "available".
> 					
> * Added a clause to "Simple video communication service" saying that the
> service provider monitors the quality of service, and derived reqs F38 and
> A26.
>
> Most of the above are things that was documented in the minutes of the
> Interim June 12th; I took the liberty to add some text (and reqs
> A26/F38) on the service provider monitoring the QoE.
>
> Feedback and comments are most welcome.
>
> Also, Ekr has the task to deliver an Identity related use-case (as discussed
> at the interim).
>
> Stefan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 2012-06-28 16:36, Cullen Jennings wrote:
> 
> I think you need to separate this for audio and video and be far more
> specific about what type of retransmit ion you are talking about. In
> many cases the retransmit ion schemes for audio make the end suer
> experience worse.

I am not disagreeing unless the RTT is really low.

What I am asking the WG is if RTP retransmission is a RECOMMENDED or
REQUIRED feature in the toolbox that an WebRTC end-point supports. This
says nothing on when you select to use it and on which media. If we want
to include such recommendations we can do it. In fact the RTP usage
draft has a bit on text discussing the issue with RTT.

Cheers

Magnus
(As WG chair)

> 
> On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:
> 
>> WG,
>> 
>> We had a discussion at the interim if RTP Retransmission is to be 
>> considered REQUIRED or RECOMMENDED to implement. I would like to
>> see if we can first have some discussion on this topic before
>> moving on to see if we can get a consensus here on the mailing
>> list.
>> 
>> Please provide your views on this topic.
>> 
>> Cheers
>> 
>> Magnus Westerlund (As Chair and document editor)
>> 
>> 
>> ----------------------------------------------------------------------
>>
>> 
Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>>
>> 
Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079 SE-164 80
>> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com 
>> ----------------------------------------------------------------------
>>
>>
>>
>> 
_______________________________________________
>> rtcweb mailing list rtcweb@ietf.org 
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

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




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To: Roni Even <ron.even.tlv@gmail.com>
References: <4FEAB80A.7040207@ericsson.com> <4FEC0C73.4030709@ericsson.com> <00af01cd553d$b72ce070$2586a150$@gmail.com> <451142FF-AB75-4C87-9936-30DDC212FBC8@csperkins.org> <00eb01cd557b$dc03d730$940b8590$@gmail.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 2012-06-29 00:17, Roni Even wrote:
> Colin,
> Magnus also said that the penalty was losing lip synch, this is why it is
> not practical. His solution is for proof of work which I agree it is working
> but in my view not relevant for real usage. In order to use it you may need
> a shorter jitter buffer and make a decision to ask for retransmit early, my
> view is that larger jitter or playout buffer for late arrival will work
> better.

The loss of lip sync is one possible way of implementing retransmission
to handle situations where the RTT is large enough.

My main point is that all applications are not created equal and what
quality expectations an application has depends on its needs.

> So if there are some network path with short RTT (I assume not the Internet)
> they can use retransmission, so recommended will be enough with text explain
> when it make sense to support it.

I have low enough RTT within most of Europe from Stockholm over wired
access to wired access to use retransmission with minimal impact on user
experience. If I am willing to accept some degradation in the quality in
form of increased delay or jitter I can use retransmission on
intercontinental paths.

Yes, we should have a reasonable description on what to consider when
using RTP retransmission. Would you please be kind to review the current
text in the RTP usage draft and propose text or at least what should be
changed.

Best Regards

Magnus


> Roni
> 
> 
> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org] 
> Sent: 28 June, 2012 4:15 PM
> To: Roni Even
> Cc: 'Magnus Westerlund'; rtcweb@ietf.org
> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or
> RECOMMENDED
> 
> Roni,
> 
> I don't understand how you can say this is not practical. Magnus has an
> implementation, and he states it works well. That seems practical to me.
> 
> There are also self-evidently network paths with low enough RTT for RTP
> retransmission to be used for interactive calls, if the playout buffer code
> knows the RTT and adjusts the buffer to allow this. Not all paths, I agree,
> but that doesn't mean it's not useful in some environments. 
> 
> We're defining requirements for the RTP infrastructure in WebRTC clients
> here, not forcing all implementations to use every feature. My personal view
> is that retransmission should be REQUIRED. Given that the RTP/AVPF profile
> is REQUIRED, implementation of retransmission is straightforward. I see no
> benefit in leaving it out, and very little cost in requiring it.
> 
> Colin
> 
> 
> On 28 Jun 2012, at 15:52, Roni Even wrote:
>> Hi Magnus,
>> So using retransmission is at the cost of losing lip synch which is very
> noticeable. I am not saying that it does not work to do retransmission, my
> claim that it is not practical and this is why I do not think it is
> required.
>> Roni
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On 
>> Behalf Of Magnus Westerlund
>> Sent: 28 June, 2012 9:49 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or 
>> RECOMMENDED
>>
>> Hi,
>>
>> As Individual I like to state my position.
>>
>> We have a video conference system developed by my colleagues used
> internally at Ericsson that uses RTP Retransmission for video, not for audio
> with great success. This is implemented such that we actually allow the
> video to fall behind the audio when packet loss and retransmission is not
> able to repair in a timely enough fashion. The benefit is minimal overhead
> and still no loss induced degradations in the video. Yes, we get degradation
> in form of frame display jittering and short freezes. But those events that
> are truly visible are rare over wired networks.
>>
>> I am personally convinced that RTP Retransmission is great tool in the
> toolbox when it comes to improve media quality in many use cases. Yes there
> are scenarios where RTP retransmission is less efficient. Long RTTs (over
> 200-400 ms) is the primary source of degradations. Compared to FEC it so
> much more efficient from bandwidth consumption perspective.
>>
>> I also think it is important that we have some mandatory to implement tool
> for making the transport more robust now that we have a consensus that we
> are not going for a FEC solution in the initial specification.
>>
>> Thus my personal position is that RTP Retransmission should be REQUIRED to
> implement.
>>
>> Cheers
>>
>> Magnus
>>
>>
>> On 2012-06-27 09:36, Magnus Westerlund wrote:
>>> WG,
>>>
>>> We had a discussion at the interim if RTP Retransmission is to be 
>>> considered REQUIRED or RECOMMENDED to implement. I would like to see 
>>> if we can first have some discussion on this topic before moving on 
>>> to see if we can get a consensus here on the mailing list.
>>>
>>> Please provide your views on this topic.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>> (As Chair and document editor)
>>>
>>>
>>> ---------------------------------------------------------------------
>>> - Multimedia Technologies, Ericsson Research EAB/TVM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> Färögatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> ---------------------------------------------------------------------
>>> -
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>
>>
>> --
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> Färögatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 
> 
> --
> Colin Perkins
> http://csperkins.org/
> 
> 
> 
> 
> 


-- 

Magnus Westerlund

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




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To: Roni Even <ron.even.tlv@gmail.com>
References: <4FEAB80A.7040207@ericsson.com> <4FEC0C73.4030709@ericsson.com> <00af01cd553d$b72ce070$2586a150$@gmail.com> <4FEC675B.8080002@ericsson.com> <00ec01cd557c$40a42fa0$c1ec8ee0$@gmail.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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On 2012-06-29 00:20, Roni Even wrote:
> Hi Magnus,
> The demo in the interim of what happens in retransmission was not in par
> with my experience with any other recovery  methods including receiver based
> repair without FEC.

Agreed, but it did work. It was a session with intercontinental paths,
multiple of them in fact and I have no knowledge of where the hangouts
central node we used where located. Although the retransmission is
actually improved as the legs to and from are separated from a repair
perspective, there is some impact from the first leg onto the second
when it comes to delay.

I am also used to better quality in our system. But that is also not
trying to do things in a browser plugin.

Cheers

Magnus Westerlund

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




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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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--e89a8f23579def730e04c3976c9d
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Doesn't this in the end boil down to what tools we have for baseline
resilience? Retransmissions are fairly easy to implement and will in most
cases give a much better experience than for example relying on periodic
key frames or doing key frame requests. Also it isn't necessary for the
decoder to support decoding streams with lost packets.

/Stefan

On Fri, Jun 29, 2012 at 8:43 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> On 2012-06-28 16:36, Cullen Jennings wrote:
> >
> > I think you need to separate this for audio and video and be far more
> > specific about what type of retransmit ion you are talking about. In
> > many cases the retransmit ion schemes for audio make the end suer
> > experience worse.
>
> I am not disagreeing unless the RTT is really low.
>
> What I am asking the WG is if RTP retransmission is a RECOMMENDED or
> REQUIRED feature in the toolbox that an WebRTC end-point supports. This
> says nothing on when you select to use it and on which media. If we want
> to include such recommendations we can do it. In fact the RTP usage
> draft has a bit on text discussing the issue with RTT.
>
> Cheers
>
> Magnus
> (As WG chair)
>
> >
> > On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:
> >
> >> WG,
> >>
> >> We had a discussion at the interim if RTP Retransmission is to be
> >> considered REQUIRED or RECOMMENDED to implement. I would like to
> >> see if we can first have some discussion on this topic before
> >> moving on to see if we can get a consensus here on the mailing
> >> list.
> >>
> >> Please provide your views on this topic.
> >>
> >> Cheers
> >>
> >> Magnus Westerlund (As Chair and document editor)
> >>
> >>
> >> ----------------------------------------------------------------------
> >>
> >>
> Multimedia Technologies, Ericsson Research EAB/TVM
> >> ----------------------------------------------------------------------
> >>
> >>
> Ericsson AB                | Phone  +46 10 7148287
> >> F=E4r=F6gatan 6                | Mobile +46 73 0949079 SE-164 80
> >> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> >> ----------------------------------------------------------------------
> >>
> >>
> >>
> >>
> _______________________________________________
> >> rtcweb mailing list rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >
> >
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--e89a8f23579def730e04c3976c9d
Content-Type: text/html; charset=ISO-8859-1
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Doesn&#39;t this in the end boil down to what tools we have for baseline re=
silience? Retransmissions are fairly easy to implement and will in most cas=
es give a much better experience than for example relying on periodic key f=
rames or doing key frame requests. Also it isn&#39;t necessary for the deco=
der to support decoding streams with lost packets.=A0<div>
<br></div><div>/Stefan</div><br><div class=3D"gmail_quote">On Fri, Jun 29, =
2012 at 8:43 AM, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:=
magnus.westerlund@ericsson.com" target=3D"_blank">magnus.westerlund@ericsso=
n.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">On 2012-06-28 16:36, Culle=
n Jennings wrote:<br>
&gt;<br>
&gt; I think you need to separate this for audio and video and be far more<=
br>
&gt; specific about what type of retransmit ion you are talking about. In<b=
r>
&gt; many cases the retransmit ion schemes for audio make the end suer<br>
&gt; experience worse.<br>
<br>
</div>I am not disagreeing unless the RTT is really low.<br>
<br>
What I am asking the WG is if RTP retransmission is a RECOMMENDED or<br>
REQUIRED feature in the toolbox that an WebRTC end-point supports. This<br>
says nothing on when you select to use it and on which media. If we want<br=
>
to include such recommendations we can do it. In fact the RTP usage<br>
draft has a bit on text discussing the issue with RTT.<br>
<br>
Cheers<br>
<br>
Magnus<br>
(As WG chair)<br>
<div class=3D"im HOEnZb"><br>
&gt;<br>
&gt; On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:<br>
&gt;<br>
&gt;&gt; WG,<br>
&gt;&gt;<br>
&gt;&gt; We had a discussion at the interim if RTP Retransmission is to be<=
br>
&gt;&gt; considered REQUIRED or RECOMMENDED to implement. I would like to<b=
r>
&gt;&gt; see if we can first have some discussion on this topic before<br>
&gt;&gt; moving on to see if we can get a consensus here on the mailing<br>
&gt;&gt; list.<br>
&gt;&gt;<br>
&gt;&gt; Please provide your views on this topic.<br>
&gt;&gt;<br>
&gt;&gt; Cheers<br>
&gt;&gt;<br>
&gt;&gt; Magnus Westerlund (As Chair and document editor)<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt;<br>
&gt;&gt;<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt;<br>
&gt;&gt;<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
&gt;&gt; F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D=
"tel:%2B46%2073%200949079" value=3D"+46730949079">+46 73 0949079</a> SE-164=
 80<br>
&gt;&gt; Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@eri=
csson.com">magnus.westerlund@ericsson.com</a><br>
&gt;&gt; ------------------------------------------------------------------=
----<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt;<br>
_______________________________________________<br>
&gt;&gt; rtcweb mailing list <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a><br>
&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D=
"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt;<br>
<br>
<br>
</div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
<br>
Magnus Westerlund<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
----------------------------------------------------------------------<br>
Multimedia Technologies, Ericsson Research EAB/TVM<br>
----------------------------------------------------------------------<br>
Ericsson AB =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Phone =A0<a href=3D"tel:%2B46%=
2010%207148287" value=3D"+46107148287">+46 10 7148287</a><br>
F=E4r=F6gatan 6 =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0| Mobile <a href=3D"tel:%2B4=
6%2073%200949079" value=3D"+46730949079">+46 73 0949079</a><br>
SE-164 80 Stockholm, Sweden| mailto: <a href=3D"mailto:magnus.westerlund@er=
icsson.com">magnus.westerlund@ericsson.com</a><br>
----------------------------------------------------------------------<br>
<br>
<br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

--e89a8f23579def730e04c3976c9d--

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To: Colin Perkins <csp@csperkins.org>, Cullen Jennings <fluffy@iii.ca>
Date: Fri, 29 Jun 2012 10:26:28 +0200
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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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Colin is correct.

This is not a profiling activity.

This is an update activity to an existing AVTCORE specification.

Regards

Keith

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Colin Perkins
Sent: 28 June 2012 19:10
To: Cullen Jennings
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs

On 28 Jun 2012, at 15:53, Cullen Jennings wrote:
> Colin & Magnus, given this is starting to look like it is likely to get c=
onsensus, could you update draft-ietf-rtcweb-rtp-usage to have text for ran=
dom generation and resonating behind it?


As I said, write a draft, and we'll reference it. I don't think this draft =
is the place to update the RTP specification.

--=20
Colin Perkins
http://csperkins.org/



_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Target numbers for setup time (Re: Keeping up data channel)
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as I said during the interim
we can reduce the 3 way handshake in the channel setup, but at cost of 
complicate "a little" the data framing


On 6/28/12 10:49 PM, Cullen Jennings (fluffy) wrote:
> Agree - that's why I think this has to be done in zero RTT. I don't see any problem with setting up a data channel and being ready to send media before having the UI tell the user they are "connected and can start talking".
>
> Reducing from 7 RTT to 5 RTT does not help when you need to get to 0 RTT.
>
> On Jun 22, 2012, at 10:09 , Harald Alvestrand wrote:
>
>> On 06/22/2012 05:58 PM, Cullen Jennings (fluffy) wrote:
>>> On Jun 13, 2012, at 5:29 , Randell Jesup wrote:
>>>
>>>>> How far down do you think we have to drive the setup time before you
>>>>> would not call it "abysmal"?
>>> I'd probably consider above 250 ms abysmal but good news I don't see any problem with getting it down around 100 ms in when both endpoints are in a single country.
>>>
>> Coast-to-coast US is ~4800 km, so RTT (9600 km) is 32 ms (speed of light is 300 km/msec).
>>
>> So, without considering processing time, 3 RTT is 100 msec, 7 RTT is "abysmal".
>>
>> There are bigger countries than the US, but this will do for a back-of-the-envelope.
>>
>>
>>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



From john@jlc.net  Fri Jun 29 07:25:30 2012
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From: John Leslie <john@jlc.net>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> 
> I am not disagreeing unless the RTT is really low.

   Actually, there is a balancing act between RTT and jitter...

> What I am asking the WG is if RTP retransmission is a RECOMMENDED or
> REQUIRED feature in the toolbox that an WebRTC end-point supports. This
> says nothing on when you select to use it and on which media. If we want
> to include such recommendations we can do it. In fact the RTP usage
> draft has a bit on text discussing the issue with RTT.

   I'd like to support Magnus here: I believe retransmission SHOULD be a
tool available to the recipient to request. There will be cases where
this is the most appropriate tool.

--
John Leslie <john@jlc.net>

From fluffy@iii.ca  Fri Jun 29 08:11:40 2012
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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Right - so on the question of it retransmission is mandatory to =
implement for audio codec, I am on a definitely "No". The bulk of =
systems today do not do it and work fine. Vendors can easily choose to =
do if they want in an interoperable way with out it being MTI. Why we =
should add a bunch of stuff in to version 1 of this that we can live =
without is beyond me. This is how IPv6 got big and hard, by everyone =
taking their favorite technology and attaching it to v6. I don't even =
want it as RECOMMENDED for audio - I see it as OPTIONAL.=20

I probably feel differently about video.=20


On Jun 28, 2012, at 23:43 , Magnus Westerlund wrote:

> On 2012-06-28 16:36, Cullen Jennings wrote:
>>=20
>> I think you need to separate this for audio and video and be far more
>> specific about what type of retransmit ion you are talking about. In
>> many cases the retransmit ion schemes for audio make the end suer
>> experience worse.
>=20
> I am not disagreeing unless the RTT is really low.
>=20
> What I am asking the WG is if RTP retransmission is a RECOMMENDED or
> REQUIRED feature in the toolbox that an WebRTC end-point supports. =
This
> says nothing on when you select to use it and on which media. If we =
want
> to include such recommendations we can do it. In fact the RTP usage
> draft has a bit on text discussing the issue with RTT.
>=20
> Cheers
>=20
> Magnus
> (As WG chair)
>=20
>>=20
>> On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:
>>=20
>>> WG,
>>>=20
>>> We had a discussion at the interim if RTP Retransmission is to be=20
>>> considered REQUIRED or RECOMMENDED to implement. I would like to
>>> see if we can first have some discussion on this topic before
>>> moving on to see if we can get a consensus here on the mailing
>>> list.
>>>=20
>>> Please provide your views on this topic.
>>>=20
>>> Cheers
>>>=20
>>> Magnus Westerlund (As Chair and document editor)
>>>=20
>>>=20
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
> Multimedia Technologies, Ericsson Research EAB/TVM
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
> Ericsson AB                | Phone  +46 10 7148287
>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079 SE-164 80
>>> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com=20
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
>>>=20
>>>=20
> _______________________________________________
>>> rtcweb mailing list rtcweb@ietf.org=20
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
>=20


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Cullen,

You're being asked what RTP tools the browser needs to implement, not =
what RTP tools you want use with audio or video codecs.=20

Colin


On 29 Jun 2012, at 16:11, Cullen Jennings wrote:
> Right - so on the question of it retransmission is mandatory to =
implement for audio codec, I am on a definitely "No". The bulk of =
systems today do not do it and work fine. Vendors can easily choose to =
do if they want in an interoperable way with out it being MTI. Why we =
should add a bunch of stuff in to version 1 of this that we can live =
without is beyond me. This is how IPv6 got big and hard, by everyone =
taking their favorite technology and attaching it to v6. I don't even =
want it as RECOMMENDED for audio - I see it as OPTIONAL.=20
>=20
> I probably feel differently about video.=20
>=20
>=20
> On Jun 28, 2012, at 23:43 , Magnus Westerlund wrote:
>=20
>> On 2012-06-28 16:36, Cullen Jennings wrote:
>>>=20
>>> I think you need to separate this for audio and video and be far =
more
>>> specific about what type of retransmit ion you are talking about. In
>>> many cases the retransmit ion schemes for audio make the end suer
>>> experience worse.
>>=20
>> I am not disagreeing unless the RTT is really low.
>>=20
>> What I am asking the WG is if RTP retransmission is a RECOMMENDED or
>> REQUIRED feature in the toolbox that an WebRTC end-point supports. =
This
>> says nothing on when you select to use it and on which media. If we =
want
>> to include such recommendations we can do it. In fact the RTP usage
>> draft has a bit on text discussing the issue with RTT.
>>=20
>> Cheers
>>=20
>> Magnus
>> (As WG chair)
>>=20
>>>=20
>>> On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:
>>>=20
>>>> WG,
>>>>=20
>>>> We had a discussion at the interim if RTP Retransmission is to be=20=

>>>> considered REQUIRED or RECOMMENDED to implement. I would like to
>>>> see if we can first have some discussion on this topic before
>>>> moving on to see if we can get a consensus here on the mailing
>>>> list.
>>>>=20
>>>> Please provide your views on this topic.
>>>>=20
>>>> Cheers
>>>>=20
>>>> Magnus Westerlund (As Chair and document editor)
>>>>=20
>>>>=20
>>>> =
----------------------------------------------------------------------
>>>>=20
>>>>=20
>> Multimedia Technologies, Ericsson Research EAB/TVM
>>>> =
----------------------------------------------------------------------
>>>>=20
>>>>=20
>> Ericsson AB                | Phone  +46 10 7148287
>>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079 SE-164 80
>>>> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com=20
>>>> =
----------------------------------------------------------------------
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>> _______________________________________________
>>>> rtcweb mailing list rtcweb@ietf.org=20
>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>>=20
>>=20
>>=20
>> --=20
>>=20
>> Magnus Westerlund
>>=20
>> =
----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM
>> =
----------------------------------------------------------------------
>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>>=20
>>=20
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Colin Perkins
http://csperkins.org/




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From: Cullen Jennings <fluffy@iii.ca>
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To: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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RFC 3550 has the words=20

   To meet these requirements, the
   following format SHOULD be used unless a profile specifies an
   alternate syntax or semantics. =20

what I am suggesting here is that we define a profile that specifies a =
alternative format but still meets all the other SHOULD and MUST in 3550 =
that are not part of the "following format" referred to in that =
sentence.=20

Now several people are making the argument we need to go and deprecate =
6222 and replace it with something else. But I'm not seeing any process =
reasons why this could not be done as profile if that was what we wanted =
to do.=20

Thoughts?



On Jun 29, 2012, at 1:26 , DRAGE, Keith (Keith) wrote:

> Colin is correct.
>=20
> This is not a profiling activity.
>=20
> This is an update activity to an existing AVTCORE specification.
>=20
> Regards
>=20
> Keith
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Colin Perkins
> Sent: 28 June 2012 19:10
> To: Cullen Jennings
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
>=20
> On 28 Jun 2012, at 15:53, Cullen Jennings wrote:
>> Colin & Magnus, given this is starting to look like it is likely to =
get consensus, could you update draft-ietf-rtcweb-rtp-usage to have text =
for random generation and resonating behind it?
>=20
>=20
> As I said, write a draft, and we'll reference it. I don't think this =
draft is the place to update the RTP specification.
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] [AVTCORE]  Randomly-generated CNAMEs
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A profile is something specific in RTP. You really don't want to define =
a new RTP profile for this - that's a far bigger job than updating =
RFC3550.

Colin



On 29 Jun 2012, at 16:35, Cullen Jennings wrote:
> RFC 3550 has the words=20
>=20
>   To meet these requirements, the
>   following format SHOULD be used unless a profile specifies an
>   alternate syntax or semantics. =20
>=20
> what I am suggesting here is that we define a profile that specifies a =
alternative format but still meets all the other SHOULD and MUST in 3550 =
that are not part of the "following format" referred to in that =
sentence.=20
>=20
> Now several people are making the argument we need to go and deprecate =
6222 and replace it with something else. But I'm not seeing any process =
reasons why this could not be done as profile if that was what we wanted =
to do.=20
>=20
> Thoughts?
>=20
>=20
>=20
> On Jun 29, 2012, at 1:26 , DRAGE, Keith (Keith) wrote:
>=20
>> Colin is correct.
>>=20
>> This is not a profiling activity.
>>=20
>> This is an update activity to an existing AVTCORE specification.
>>=20
>> Regards
>>=20
>> Keith
>>=20
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Colin Perkins
>> Sent: 28 June 2012 19:10
>> To: Cullen Jennings
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
>>=20
>> On 28 Jun 2012, at 15:53, Cullen Jennings wrote:
>>> Colin & Magnus, given this is starting to look like it is likely to =
get consensus, could you update draft-ietf-rtcweb-rtp-usage to have text =
for random generation and resonating behind it?
>>=20
>>=20
>> As I said, write a draft, and we'll reference it. I don't think this =
draft is the place to update the RTP specification.
>>=20
>> --=20
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20



--=20
Colin Perkins
http://csperkins.org/




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Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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as Ted regularly points out, this is used by stuff other than browsers. =
So some RTCWeb endpoints will not do video. All I am saying is that I =
don't think things that are audio only should have to implement =
retransmission.=20

In general my answer to what RTP tools are needed will be the minimal =
set that has proven to be needed to get things to work on the internet. =
At the Google IO conference this week, Google announced three major =
browser vendors plan to ship WebRTC before the end of 2012 and it would =
also work inside IE using chrome frame. I've also been looking at our =
milestones dates - we are going to need to have some long discussions =
about what we can safely leave till later if the IETF wants to produce =
standards that are done soon enough that they are not ignored.=20


On Jun 29, 2012, at 8:26 , Colin Perkins wrote:

> Cullen,
>=20
> You're being asked what RTP tools the browser needs to implement, not =
what RTP tools you want use with audio or video codecs.=20
>=20
> Colin
>=20
>=20
> On 29 Jun 2012, at 16:11, Cullen Jennings wrote:
>> Right - so on the question of it retransmission is mandatory to =
implement for audio codec, I am on a definitely "No". The bulk of =
systems today do not do it and work fine. Vendors can easily choose to =
do if they want in an interoperable way with out it being MTI. Why we =
should add a bunch of stuff in to version 1 of this that we can live =
without is beyond me. This is how IPv6 got big and hard, by everyone =
taking their favorite technology and attaching it to v6. I don't even =
want it as RECOMMENDED for audio - I see it as OPTIONAL.=20
>>=20
>> I probably feel differently about video.=20
>>=20
>>=20
>> On Jun 28, 2012, at 23:43 , Magnus Westerlund wrote:
>>=20
>>> On 2012-06-28 16:36, Cullen Jennings wrote:
>>>>=20
>>>> I think you need to separate this for audio and video and be far =
more
>>>> specific about what type of retransmit ion you are talking about. =
In
>>>> many cases the retransmit ion schemes for audio make the end suer
>>>> experience worse.
>>>=20
>>> I am not disagreeing unless the RTT is really low.
>>>=20
>>> What I am asking the WG is if RTP retransmission is a RECOMMENDED or
>>> REQUIRED feature in the toolbox that an WebRTC end-point supports. =
This
>>> says nothing on when you select to use it and on which media. If we =
want
>>> to include such recommendations we can do it. In fact the RTP usage
>>> draft has a bit on text discussing the issue with RTT.
>>>=20
>>> Cheers
>>>=20
>>> Magnus
>>> (As WG chair)
>>>=20
>>>>=20
>>>> On Jun 27, 2012, at 24:36 , Magnus Westerlund wrote:
>>>>=20
>>>>> WG,
>>>>>=20
>>>>> We had a discussion at the interim if RTP Retransmission is to be=20=

>>>>> considered REQUIRED or RECOMMENDED to implement. I would like to
>>>>> see if we can first have some discussion on this topic before
>>>>> moving on to see if we can get a consensus here on the mailing
>>>>> list.
>>>>>=20
>>>>> Please provide your views on this topic.
>>>>>=20
>>>>> Cheers
>>>>>=20
>>>>> Magnus Westerlund (As Chair and document editor)
>>>>>=20
>>>>>=20
>>>>> =
----------------------------------------------------------------------
>>>>>=20
>>>>>=20
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>>>> =
----------------------------------------------------------------------
>>>>>=20
>>>>>=20
>>> Ericsson AB                | Phone  +46 10 7148287
>>>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079 SE-164 80
>>>>> Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com=20
>>>>> =
----------------------------------------------------------------------
>>>>>=20
>>>>>=20
>>>>>=20
>>>>>=20
>>> _______________________________________________
>>>>> rtcweb mailing list rtcweb@ietf.org=20
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>=20
>>>>=20
>>>=20
>>>=20
>>> --=20
>>>=20
>>> Magnus Westerlund
>>>=20
>>> =
----------------------------------------------------------------------
>>> Multimedia Technologies, Ericsson Research EAB/TVM
>>> =
----------------------------------------------------------------------
>>> Ericsson AB                | Phone  +46 10 7148287
>>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>>> =
----------------------------------------------------------------------
>>>=20
>>>=20
>>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20


From fluffy@iii.ca  Fri Jun 29 08:49:20 2012
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From: Cullen Jennings <fluffy@iii.ca>
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To: Colin Perkins <csp@csperkins.org>
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ok - Fair enough.

On Jun 29, 2012, at 8:40 , Colin Perkins wrote:

> A profile is something specific in RTP. You really don't want to =
define a new RTP profile for this - that's a far bigger job than =
updating RFC3550.
>=20
> Colin
>=20
>=20
>=20
> On 29 Jun 2012, at 16:35, Cullen Jennings wrote:
>> RFC 3550 has the words=20
>>=20
>>  To meet these requirements, the
>>  following format SHOULD be used unless a profile specifies an
>>  alternate syntax or semantics. =20
>>=20
>> what I am suggesting here is that we define a profile that specifies =
a alternative format but still meets all the other SHOULD and MUST in =
3550 that are not part of the "following format" referred to in that =
sentence.=20
>>=20
>> Now several people are making the argument we need to go and =
deprecate 6222 and replace it with something else. But I'm not seeing =
any process reasons why this could not be done as profile if that was =
what we wanted to do.=20
>>=20
>> Thoughts?
>>=20
>>=20
>>=20
>> On Jun 29, 2012, at 1:26 , DRAGE, Keith (Keith) wrote:
>>=20
>>> Colin is correct.
>>>=20
>>> This is not a profiling activity.
>>>=20
>>> This is an update activity to an existing AVTCORE specification.
>>>=20
>>> Regards
>>>=20
>>> Keith
>>>=20
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Colin Perkins
>>> Sent: 28 June 2012 19:10
>>> To: Cullen Jennings
>>> Cc: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] [AVTCORE] Randomly-generated CNAMEs
>>>=20
>>> On 28 Jun 2012, at 15:53, Cullen Jennings wrote:
>>>> Colin & Magnus, given this is starting to look like it is likely to =
get consensus, could you update draft-ietf-rtcweb-rtp-usage to have text =
for random generation and resonating behind it?
>>>=20
>>>=20
>>> As I said, write a draft, and we'll reference it. I don't think this =
draft is the place to update the RTP specification.
>>>=20
>>> --=20
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20


From radhika.r.roy.civ@mail.mil  Fri Jun 29 09:18:00 2012
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From: "Roy, Radhika R CIV (US)" <radhika.r.roy.civ@mail.mil>
To: Cullen Jennings <fluffy@iii.ca>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMENDED
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Yes, I support Cullen because so many problems are there for RTP-retransmis=
sions since the inception real-time conversational applications including l=
ip-synch.

With respect video quality, it is a question mark what you are getting at t=
he cost of what, where, and how. Who is paying for the extra-bandwidth that=
 may even lead to severe congestions over the networks if it is not done ve=
ry carefully and selectively and if this is what all users try to do "fun" =
through retransmissions? Video-RTP-retransmissions are NOT well proven in t=
he public Internet for wider use.

I even NOT want for even for video retransmissions, if it's the audio and v=
ideo conversational application. If anyone likes to do this, let us keep th=
is a private (optional) at users' discretion without publicly supporting in=
 the IETF RFCs. At best, we can support the word "MAY."

BR/Radhika

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings
Sent: Friday, June 29, 2012 11:12 AM
To: Magnus Westerlund
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] RTP Usage: Is RTP Retransmission REQUIRED or RECOMMEN=
DED


Right - so on the question of it retransmission is mandatory to implement f=
or audio codec, I am on a definitely "No". The bulk of systems today do not=
 do it and work fine. Vendors can easily choose to do if they want in an in=
teroperable way with out it being MTI. Why we should add a bunch of stuff i=
n to version 1 of this that we can live without is beyond me. This is how I=
Pv6 got big and hard, by everyone taking their favorite technology and atta=
ching it to v6. I don't even want it as RECOMMENDED for audio - I see it as=
 OPTIONAL.=20

I probably feel differently about video.=20


On Jun 28, 2012, at 23:43 , Magnus Westerlund wrote:



From randell-ietf@jesup.org  Sat Jun 30 05:12:37 2012
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On 6/28/2012 3:49 AM, Magnus Westerlund wrote:
> Hi,
>
> As Individual I like to state my position.
>
> We have a video conference system developed by my colleagues used
> internally at Ericsson that uses RTP Retransmission for video, not for
> audio with great success. This is implemented such that we actually
> allow the video to fall behind the audio when packet loss and
> retransmission is not able to repair in a timely enough fashion.

With my videophone-engineer's hat on.... I'm really surprised that this 
produces a good User Experience.  Sync failure (especially 
audio-before-video) is jarring to users.  Some classes of users may 
learn to ignore it more, or with very large displays they may prefer 
that to quality degradation, or the alternatives they've been used to 
were even worse.

I believe however that there are better ways to deal with loss.  But 
this really does come down to "what sort of artifacts do you find less 
annoying/noticeable/jarring?"  There's also a tradeoff with jitter 
buffer depths (deeper buffers allow more time for recovery, at a cost).

> The
> benefit is minimal overhead and still no loss induced degradations in
> the video. Yes, we get degradation in form of frame display jittering
> and short freezes. But those events that are truly visible are rare over
> wired networks.

The operative issues are: how long after expected arrival time is the 
re-xmit requested; how long does it take to get there, how long to get 
the re-transmitted packet, how much induced delay in the video occurs, 
what's the interaction with jitter buffer depth, what's the typical BW 
increase, packet size impacts, etc.

If you run longish jitter buffers (compared to actual jitter), and are 
on a fast network with low loss, it might be an improvement -- the long 
jitter buffer and fast network lets re-xmits arrive "in-time" or close 
to, minimizing induced delay and visual glitches, and total delay can 
stay in bounds.  (and this may be an overly-generous assessment)  Most 
real-world, open internet cases it will not make sense; not even close IMHO.

> I am personally convinced that RTP Retransmission is great tool in the
> toolbox when it comes to improve media quality in many use cases. Yes
> there are scenarios where RTP retransmission is less efficient. Long
> RTTs (over 200-400 ms) is the primary source of degradations. Compared
> to FEC it so much more efficient from bandwidth consumption perspective.

I'd also almost never want to use FEC for video, but my preference is 
for accepting temporary visual glitches and recovering as fast as 
possible with a high, constant framerate.  Don't ever freeze someone for 
more than a frame if you can avoid it.

> I also think it is important that we have some mandatory to implement
> tool for making the transport more robust now that we have a consensus
> that we are not going for a FEC solution in the initial specification.
>
> Thus my personal position is that RTP Retransmission should be REQUIRED
> to implement.

The other trick is *knowing* when to use it.  Something has to know if a 
re-xmit is likely to help instead of just wasting more bandwidth.  Is 
this under app control?  Is this logic baked into the core webrtc code? 
  How long is data held for possible retransmission?  (Perhaps the RFC 
says, I didn't check.)  How aggressive is it at asking for retransmits 
of late packets?  To make it work, do we need to artificially 
lower-bound the jitter-buffer depth?  Does the application get to decide 
to block video and let sync fail?

I understand Magnus' request, but I'm tempted to RECOMMEND.  Also: lack 
of support will not make calls fail, they'll just use "normal" 
error-recovery techniques.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 6/28/2012 3:49 PM, Cullen Jennings (fluffy) wrote:
> Agree - that's why I think this has to be done in zero RTT. I don't see any problem with setting up a data channel and being ready to send media before having the UI tell the user they are "connected and can start talking".

I agree, and have suggested this as well, but it's really in the 
application domain (though we can give examples that use it and 
recommend it).  There are some mild security concerns with accepting the 
data-only connection before user decision, but manageable I believe.

Roughly: call start (user)
      caller-> Offer(SDP(data-only)) as "setup" -> server -> answerer
             (also indicate if it's audio-only or audio+video)
             (maybe offer data+audio (and video) to warm up ICE candidates)
      answerer -> Answer (SDP(data-only)) as "setup" -> server -> answerer
      answerer -> alert user
      caller-> install Answer
      <N RTT of data channel setup setting up "negotiation" channel>
      caller-> Offer(SDP(data+media)) as "call" -> answerer (over 
"negotiation" channel)
      ...
      answerer (user) accepts
      answerer-> Answer(SDP(data+media)) as "call" -> caller (over 
"negotiation" channel)
      answerer starts sending
      caller installs Answer
      caller media starts sending

 From user accepts -> media starts, it's 0 RTT for answer sending, 1 RTT 
until he sees media from the caller, which is pretty much close to 
theoretical minimum.  (You can do better only by starting media channels 
"in the background" from the caller before user accepts, perhaps with an 
Answer(SDP(data+receive-only-media)).)

Again, I believe this is all possible in the application domain today 
with the spec.

>
> Reducing from 7 RTT to 5 RTT does not help when you need to get to 0 RTT.
>
> On Jun 22, 2012, at 10:09 , Harald Alvestrand wrote:
>
>> On 06/22/2012 05:58 PM, Cullen Jennings (fluffy) wrote:
>>> On Jun 13, 2012, at 5:29 , Randell Jesup wrote:
>>>
>>>>> How far down do you think we have to drive the setup time before you
>>>>> would not call it "abysmal"?
>>> I'd probably consider above 250 ms abysmal but good news I don't see any problem with getting it down around 100 ms in when both endpoints are in a single country.
>>>
>> Coast-to-coast US is ~4800 km, so RTT (9600 km) is 32 ms (speed of light is 300 km/msec).
>>
>> So, without considering processing time, 3 RTT is 100 msec, 7 RTT is "abysmal".
>>
>> There are bigger countries than the US, but this will do for a back-of-the-envelope.
>>
>> -- 
>> Randell Jesup
>> randell-ietf@jesup.org

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On 6/29/2012 3:30 AM, Stefan Holmer wrote:
> Doesn't this in the end boil down to what tools we have for baseline
> resilience? Retransmissions are fairly easy to implement and will in
> most cases give a much better experience than for example relying on
> periodic key frames or doing key frame requests. Also it isn't necessary
> for the decoder to support decoding streams with lost packets.

"Better experience" incorporates a bunch of assumptions.

Assuming we're talking video only:

If you want it to be seamless, you have to be running a jitter buffer 
depth of at least 1 RTT plus time-to-decide-packet-may-be-lost plus 
random-constant.  The second part of that can be tricky and 
network-state dependent.  On corporate networks, I'd see sudden 100ms 
changes in base delay over 1-3 packets; a short time-to-decide would 
flag each of those as missing.  This is ok (the re-xmit will get ignored 
as a dup), but adds packets at an apparent bad point for the network.

That said, small RTTs certainly happen within LANs and corporations, and 
between neighbors in a single provider often.

Note that often/normally with small RTTs you also get reasonably small 
jitter, and thus the adaptive jitter buffer might run very small 
numbers, so one might need to put a lower limit on the jitter buffer in 
these cases.

The alternatives:

1) request re-xmit and freeze video queue when the jitter buffer runs 
dry.  If the packet comes in ASAP, freeze <= RTT+constant (modulo 
application scheduling delay at the sender side).  Then you can repair 
the broken frame, decode it, and then decode any following frames. 
Typically after a freeze you drop a frame or more; you might end up 
showing some of them depending on decode speed and how long the freeze 
lasted.  If the re-xmit (or the request) is lost, then delays can 
stretch much longer, since you need to wait RTT+bigger constant before 
requesting again.

In some conference apps even if the mixer just forwards video packets, 
the RTT is receiver<->mixer, not receiver<->mixer<->sender (this 
requires the mixer to buffer packets for possible retransmission).

2) request IDR (or refresh of corrupted slice).  Downside here is that 
you need to wait for the next frame encode at the sender side (so 0-33ms 
normally, or perhaps 0-100ms), and then the frame typically takes many 
packets to send (which also may be lost, and take time to receive 
especially on low-BW links), and IDR initial quality may be noticeably 
lower.  The receiver may freeze the video until the IDR is received or 
keep decoding with errors.  If there's a significant chance of the IDR 
losing a packet, a freeze could be substantial time.

3) request repair.  Note that #2 and #3 may be the same at the receiver 
side, with the sender deciding the best available repair.  In this case, 
the sender would instead of an IDR use a some other set of (smaller) 
packets to create an error-free up-to-date state at the receiver, such 
as encoding using a long-term-reference-frame, or using some other 
known-error-free frame in the receiver.  Note that repair is basically 
just a normal p-frame, albeit with somewhat lower quality and/or 
somewhat higher bandwidth used.  A big plus is that the end state is 
close to the same as re-transmit, but you don't have to speed through 
decoding any skipped frames.  Repair also allows the receiver to use 
alternate recovery mechanisms than simply freezing, including simply 
continuing to decode p-frames ignoring the lost frame.  This induces 
artifacts, but keeps motion loss to a minimum, especially in longer-RTT 
links.

Note that #3 in particular should result in similar freeze length as #1, 
perhaps 1 frame longer, if it freezes.  If it plays with errors instead, 
the freeze is typically 1 frame (the lost one).

(Others?)

My normal preference is #3. But, as discussed above, in low-RTT networks 
it may be a good option and might give 0-frames of freeze.  But, you 
still have all the complexities about how to decide when and how to use 
re-xmit; I think the 1 frame freeze is a reasonable option and a 
reasonable fallback if the source doesn't re-transmit.

Also, if retransmit is REQUIRED, and the media is gatewayed to other 
sources (non-webrtc), and they don't support retransmit (likely), then 
the gateway would need to buffer and act as a retransmit agent 
(complicating the gateway).  If it's RECOMMENDED, this is not a big deal 
and the gateway can stay simpler.
-- 
Randell Jesup
randell-ietf@jesup.org

