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From: Tim Panton <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Use Case draft
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To be a little clearer.=20
If what the use case describes is a 1-for-1 replacement of an (0)800 =
number with a
rtcWeb call, (no context share, no leveraging of web identity) into a=20
standard current call center. I don't think that this case is worthwhile =
to add.

If the usecase describes how context sharing, web identity might be used =
in
a call center scenario - I'm all for it.=20

All I am saying is that if rtcWeb doesn't add any user benefits over =
(0)800 then,
based on my experience, almost no one will use it. So we should not =
allow
the like-for-like replacement case to unduly set requirements

T.
On 30 Apr 2012, at 23:27, Stephan Wenger wrote:

> Hi Randell,
> I don't buy this argument.
> Speakers are a standard piece of equipment now; almost everyone with a =
PC
> has them in some form.  Without a webcam, webrtc would be pretty =
useless,
> and every webcam nowadays has a microphone.
> I have no reason to distrust Tim's assessment, but the reason for the =
lack
> of success of PC facilitated sales doers you cited, IMO, cannot be the
> main reason, and certainly not among the expected users of webrtc (who
> WILL have speakers, camera and mic set up, ready to go, and used to =
make
> calls from their machine).  And if your argument refers to the call =
center
> site--well, those guys will get their equipment right and train their
> employees, or go out of business.
> Lets not dismiss a use case based on short term observations.
> Stephan
>=20
> On 4.30.2012 23:09 , "Randell Jesup" <randell-ietf@jesup.org> wrote:
>=20
>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>=20
>>> On 30 Apr 2012, at 09:42, Ravindran, Parthasarathi wrote:
>>=20
>>>> My experience is different. Click-to-call is attractive in case of
>>>> toll-free number in the site. WebRTC provides complete free call
>>>> without any toll.
>>>=20
>>> I can't tell you the actual numbers, but when presented with the =
choice
>>> of calling a toll free number
>>> or clicking a button marked "free internet call" - almost no-one on =
a
>>> real, busy site clicked the button.
>>> ( for every button click there were several orders of magnitude more
>>> 0800 calls from that page).
>>>=20
>>>=20
>>> So from my perspective this is a legacy interop use case with almost
>>> zero user acceptance.
>>=20
>> How many people browsing have mic, speakers, or headphones, set up, =
and
>> are comfortable with it?
>>=20
>> Things are changing, but a friend who consults for non-tech small
>> businesspeople and individual home-based businesses avoids using =
audio
>> when doing screen-share type stuff and instead starts a parallel =
audio
>> POTS call, because it's too frequently a waste of time for them to =
find
>> and set up headphones, get their mic to work.
>>=20
>> If it's people who are already using Skype/etc, there's a better =
chance
>> - but they still may only be comfortable with Skype for "calls".
>>=20
>>=20
>> --=20
>> Randell Jesup
>> randell-ietf@jesup.org
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From Jim.Barnett@genesyslab.com  Tue May  1 05:05:18 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Marshall Eubanks" <marshall.eubanks@gmail.com>, "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
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Subject: Re: [rtcweb] Use Case draft
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One way to describe the use case is to let the contact center's media =
server/gateway serve as the webRTC endpoint.  Then all the issues of =
call delivery, call monitoring, etc. disappear.  They are handled by =
application software that sits behind the webRTC endpoint.  The company =
I work for makes a good living selling software that deals with all =
these issues - including bathroom breaks - and that's how we would tend =
to think of this case.  To us, it's a new kind of call/connection coming =
into the contact center, which we translate into SIP at the border and =
then handle normally.=20

It's not clear to me if this use case adds any extra requirements.  We =
would just have to be careful not to assume that a webRTC endpoint is =
always a person/browser-based user agent.  It may seem a bit unsettling =
that the webRTC endpoint can distribute the call somewhere  else and let =
others listen in, but as far as I can tell that is already the case.  If =
Bob calls Alice with full authentication and security, he can be sure =
that he is connected to Alice's user agent and that no one in between =
can listen in, but there's nothing stopping Alice from recording the =
audio, or forwarding it to a third party.  So Bob could in fact be =
talking to Mary if that's how Alice wants to arrange things (_behind_ =
her user agent).  In general, Bob is assured only that he is talking to =
someone Alice wants him to talk to, and that no one can snoop without =
Alice's permission.  That's very much the way things work with the call =
center - you are sure that you are 1) connected securely to your bank 2) =
talking to someone that the bank wants you to talk to 3) being recorded =
or snooped on only when the bank explicitly chooses to do so. =20

- Jim=20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Marshall Eubanks
Sent: Monday, April 30, 2012 11:42 PM
To: Hutton, Andrew
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

On Mon, Apr 30, 2012 at 2:31 PM, Hutton, Andrew =
<andrew.hutton@siemens-enterprise.com> wrote:
> Whether anybody has been successful in the past with this type of use=20
> case is I think irrelevant.
>
>
>
> The enterprise call centre use case is I think a vital use case=20
> because it is a scenario in which one user is only concerned that they =

> can securely reach an organization/domain and is not concerned about=20
> the individual within that domain=A0 that they communicate with.=A0 A=20
> suspect quite a large percentage of RTCWEB applications will be like=20
> this and it is not covered in the current use case draft.

I agree that this is a very useful use case and one I think is going to =
get a lot of traction. There is a very solid business case for this.  =
However, I have a fair amount of experience with a video call center for =
a client, and it is not as simple as it might seem.

The essence of course is that you get the next available person, i.e., =
it is anycast. Determining who the next available person is is not =
trivial, nor is error recovery. (If I call you, and you don't answer or =
the call drops or whatever,  I can leave a message or try later. If I =
call a help desk, and this happens, I want a new agent, ideally
automatically.) Call forwarding (e.g., first tier to second tier =
technical support) is essential, and it may be anycast or directed.
There are also some security oddities  - if I am connecting from home, I =
may need to authenticate, use a credit card, etc. If I am connecting =
from inside a store, and providing in store video technical support is =
big part of the market, then the store authenticates me off line and the =
call really should just be a button push, which implies that the store =
has previously authenticated some sort of master session. In addition, =
unlike most video calls, in the enterprise call center a supervisor may =
need to be able to monitor (i.e., watch) a call, and in some =
circumstances (financial or medical calls, for example) there will need =
to be third party recording. I believe that  these details would be =
different from the typical RTCWEB scenario.

Also, there will be a temptation to do the anycasting by the techniques =
used to load balance servers in a data center, but I think that may not =
be sufficient. The call "center" may in fact be spread completely across =
the planet (daytime support in the US, nighttime support in India, for =
example) and be on multiple autonomous systems (and even from people's =
homes), which gives rise to some of the transport issues NVO3 may face, =
but without any opportunity for packet tagging. Plus, there will =
complicated rules about who can be selected next. RTCWEB shouldn't worry =
about the intricacies of bathroom break policies; these complexities =
should be dealt with by an enterprise-side database, which to me =
(together with some of the other issues above) suggests that this would =
probably benefit from  API support.

Regards
Marshall


>
>
>
> So I think we need it.
>
>
>
> Regards
>
> Andy
>
>
>
>
>
>
>
>
>
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Igor Faynberg
> Sent: 30 April 2012 17:41
> To: rtcweb@ietf.org
>
>
> Subject: Re: [rtcweb] Use Case draft
>
>
>
> Without numbers it is impossible to argue, but, if we talk about the=20
> perceived need, I disagree.=A0 Think of the people who travel abroad =
and=20
> cannot call the 800 number. (I routinely use Web interface for calls=20
> when
> traveling.)
>
>
>
> I am all for=A0 the use case, as described by Jim.
>
> Igor
>
> On 4/30/2012 9:54 AM, Tim Panton wrote:
>
> ...
>
> I can't tell you the actual numbers, but when presented with the=20
> choice of calling a toll free number
>
> or clicking a button marked "free internet call" - almost no-one on a=20
> real, busy site clicked the button.
>
> ( for every button click there were several orders of magnitude more=20
> 0800 calls from that page).
>
>
>
>
>
> So from my perspective this is a legacy interop use case with almost=20
> zero user acceptance.
>
>
>
> (as far as I can see no-one has made this use-case desirable in=20
> practice
> yet.)
>
> Tim.
>
>
>
>
>
>
>
> _______________________________________________
>
> 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 Jim.Barnett@genesyslab.com  Tue May  1 05:10:20 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Tim Panton" <tim@phonefromhere.com>, "Stephan Wenger" <stewe@stewe.org>
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I work for a contact center call center company, and we are interested
in the 1-for-1 replacement case.  However, we and our customers, are
_much more_ interested in the case including context sharing and web
identity, so I think that it should be part of the use case.  But an
optional part - we would like the scenario to work in the simple
PSTN-replacement case as well. Futhermore, even without web context, the
ability to include video already gives us something valuable that you
can't get with  the PSTN.

- Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Tim Panton
Sent: Tuesday, May 01, 2012 5:04 AM
To: Stephan Wenger
Cc: Randell Jesup; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

To be a little clearer.=20
If what the use case describes is a 1-for-1 replacement of an (0)800
number with a rtcWeb call, (no context share, no leveraging of web
identity) into a standard current call center. I don't think that this
case is worthwhile to add.

If the usecase describes how context sharing, web identity might be used
in a call center scenario - I'm all for it.=20

All I am saying is that if rtcWeb doesn't add any user benefits over
(0)800 then, based on my experience, almost no one will use it. So we
should not allow the like-for-like replacement case to unduly set
requirements

T.
On 30 Apr 2012, at 23:27, Stephan Wenger wrote:

> Hi Randell,
> I don't buy this argument.
> Speakers are a standard piece of equipment now; almost everyone with a

> PC has them in some form.  Without a webcam, webrtc would be pretty=20
> useless, and every webcam nowadays has a microphone.
> I have no reason to distrust Tim's assessment, but the reason for the=20
> lack of success of PC facilitated sales doers you cited, IMO, cannot=20
> be the main reason, and certainly not among the expected users of=20
> webrtc (who WILL have speakers, camera and mic set up, ready to go,=20
> and used to make calls from their machine).  And if your argument=20
> refers to the call center site--well, those guys will get their=20
> equipment right and train their employees, or go out of business.
> Lets not dismiss a use case based on short term observations.
> Stephan
>=20
> On 4.30.2012 23:09 , "Randell Jesup" <randell-ietf@jesup.org> wrote:
>=20
>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>=20
>>> On 30 Apr 2012, at 09:42, Ravindran, Parthasarathi wrote:
>>=20
>>>> My experience is different. Click-to-call is attractive in case of=20
>>>> toll-free number in the site. WebRTC provides complete free call=20
>>>> without any toll.
>>>=20
>>> I can't tell you the actual numbers, but when presented with the=20
>>> choice of calling a toll free number or clicking a button marked=20
>>> "free internet call" - almost no-one on a real, busy site clicked=20
>>> the button.
>>> ( for every button click there were several orders of magnitude more
>>> 0800 calls from that page).
>>>=20
>>>=20
>>> So from my perspective this is a legacy interop use case with almost

>>> zero user acceptance.
>>=20
>> How many people browsing have mic, speakers, or headphones, set up,=20
>> and are comfortable with it?
>>=20
>> Things are changing, but a friend who consults for non-tech small=20
>> businesspeople and individual home-based businesses avoids using=20
>> audio when doing screen-share type stuff and instead starts a=20
>> parallel audio POTS call, because it's too frequently a waste of time

>> for them to find and set up headphones, get their mic to work.
>>=20
>> If it's people who are already using Skype/etc, there's a better=20
>> chance
>> - but they still may only be comfortable with Skype for "calls".
>>=20
>>=20
>> --
>> Randell Jesup
>> randell-ietf@jesup.org
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
>=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

From marshall.eubanks@gmail.com  Tue May  1 05:33:04 2012
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Date: Tue, 1 May 2012 08:33:00 -0400
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From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: Jim Barnett <Jim.Barnett@genesyslab.com>
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Subject: Re: [rtcweb] Use Case draft
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Dear Jim;

I think that we are mostly in agreement here. One point.

On Tue, May 1, 2012 at 8:05 AM, Jim Barnett <Jim.Barnett@genesyslab.com> wr=
ote:
> One way to describe the use case is to let the contact center's media ser=
ver/gateway serve as the webRTC endpoint. =A0Then all the issues of call de=
livery, call monitoring, etc. disappear. =A0They are handled by application=
 software that sits behind the webRTC endpoint. =A0The company I work for m=
akes a good living selling software that deals with all these issues - incl=
uding bathroom breaks - and that's how we would tend to think of this case.=
 =A0To us, it's a new kind of call/connection coming into the contact cente=
r, which we translate into SIP at the border and then handle normally.
>
> It's not clear to me if this use case adds any extra requirements. =A0We =
would just have to be careful not to assume that a webRTC endpoint is alway=
s a person/browser-based user agent. =A0It may seem a bit unsettling that t=
he webRTC endpoint can distribute the call somewhere =A0else and let others=
 listen in, but as far as I can tell that is already the case. =A0If Bob ca=
lls Alice with full authentication and security, he can be sure that he is =
connected to Alice's user agent and that no one in between can listen in, b=
ut there's nothing stopping Alice from recording the audio, or forwarding i=
t to a third party. =A0So Bob could in fact be talking to Mary if that's ho=
w Alice wants to arrange things (_behind_ her user agent). =A0In general, B=
ob is assured only that he is talking to someone Alice wants him to talk to=
, and that no one can snoop without Alice's permission. =A0That's very much=
 the way things work with the call center - you are sure that you are 1) co=
nnected securely to your bank 2) talking to someone that the bank wants you=
 to talk to 3) being recorded or snooped on only when the bank explicitly c=
hooses to do so.

The slight difference here is that if I call Alice at home, I have end
to end security (encryption) to Alice and the call only gets recorded
or snooped from her machine. That may not be sufficient in the
distributed call center case - the "supervisor" needs to be able to
monitor _without the cooperation of either end user_.  The threat
model is Alice (on the inside) and Eve (on the outside) are in cahoots
on some bad deed and want to cover their tracks. The call monitoring
should be done outside of Alice's sphere of influence. Say Alice is
working in a SOHO and is alone some of the time. Monitoring /
recording shouldn't rely on any of the equipment under her potential
control, which means it should occur in route, which means you
probably won't really have end to end encryption in this case.

Regards
Marshall

>
> - Jim
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Marshall Eubanks
> Sent: Monday, April 30, 2012 11:42 PM
> To: Hutton, Andrew
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>
> On Mon, Apr 30, 2012 at 2:31 PM, Hutton, Andrew <andrew.hutton@siemens-en=
terprise.com> wrote:
>> Whether anybody has been successful in the past with this type of use
>> case is I think irrelevant.
>>
>>
>>
>> The enterprise call centre use case is I think a vital use case
>> because it is a scenario in which one user is only concerned that they
>> can securely reach an organization/domain and is not concerned about
>> the individual within that domain=A0 that they communicate with.=A0 A
>> suspect quite a large percentage of RTCWEB applications will be like
>> this and it is not covered in the current use case draft.
>
> I agree that this is a very useful use case and one I think is going to g=
et a lot of traction. There is a very solid business case for this. =A0Howe=
ver, I have a fair amount of experience with a video call center for a clie=
nt, and it is not as simple as it might seem.
>
> The essence of course is that you get the next available person, i.e., it=
 is anycast. Determining who the next available person is is not trivial, n=
or is error recovery. (If I call you, and you don't answer or the call drop=
s or whatever, =A0I can leave a message or try later. If I call a help desk=
, and this happens, I want a new agent, ideally
> automatically.) Call forwarding (e.g., first tier to second tier technica=
l support) is essential, and it may be anycast or directed.
> There are also some security oddities =A0- if I am connecting from home, =
I may need to authenticate, use a credit card, etc. If I am connecting from=
 inside a store, and providing in store video technical support is big part=
 of the market, then the store authenticates me off line and the call reall=
y should just be a button push, which implies that the store has previously=
 authenticated some sort of master session. In addition, unlike most video =
calls, in the enterprise call center a supervisor may need to be able to mo=
nitor (i.e., watch) a call, and in some circumstances (financial or medical=
 calls, for example) there will need to be third party recording. I believe=
 that =A0these details would be different from the typical RTCWEB scenario.
>
> Also, there will be a temptation to do the anycasting by the techniques u=
sed to load balance servers in a data center, but I think that may not be s=
ufficient. The call "center" may in fact be spread completely across the pl=
anet (daytime support in the US, nighttime support in India, for example) a=
nd be on multiple autonomous systems (and even from people's homes), which =
gives rise to some of the transport issues NVO3 may face, but without any o=
pportunity for packet tagging. Plus, there will complicated rules about who=
 can be selected next. RTCWEB shouldn't worry about the intricacies of bath=
room break policies; these complexities should be dealt with by an enterpri=
se-side database, which to me (together with some of the other issues above=
) suggests that this would probably benefit from =A0API support.
>
> Regards
> Marshall
>
>
>>
>>
>>
>> So I think we need it.
>>
>>
>>
>> Regards
>>
>> Andy
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Igor Faynberg
>> Sent: 30 April 2012 17:41
>> To: rtcweb@ietf.org
>>
>>
>> Subject: Re: [rtcweb] Use Case draft
>>
>>
>>
>> Without numbers it is impossible to argue, but, if we talk about the
>> perceived need, I disagree.=A0 Think of the people who travel abroad and
>> cannot call the 800 number. (I routinely use Web interface for calls
>> when
>> traveling.)
>>
>>
>>
>> I am all for=A0 the use case, as described by Jim.
>>
>> Igor
>>
>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>
>> ...
>>
>> I can't tell you the actual numbers, but when presented with the
>> choice of calling a toll free number
>>
>> or clicking a button marked "free internet call" - almost no-one on a
>> real, busy site clicked the button.
>>
>> ( for every button click there were several orders of magnitude more
>> 0800 calls from that page).
>>
>>
>>
>>
>>
>> So from my perspective this is a legacy interop use case with almost
>> zero user acceptance.
>>
>>
>>
>> (as far as I can see no-one has made this use-case desirable in
>> practice
>> yet.)
>>
>> Tim.
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> 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 andrew.hutton@siemens-enterprise.com  Tue May  1 06:14:22 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 1 May 2012 15:14:18 +0200
Thread-Topic: [rtcweb] Use Case draft - Eavesdropping.
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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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Hi,

A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements-0=
7 contain the statement "It is essential that the communication cannot be e=
avesdropped" however there is no definition of what is actually meant by "e=
avesdropped" although I think we all have an idea of what it means.

Maybe it would be better to replace these statements with something that re=
fers to wiretapping and RFC 2804 (RAVEN) which actually has a definition of=
 wiretapping.

Regards
Andy




> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ted Hardie
> Sent: 27 April 2012 17:15
> To: rtcweb@ietf.org
> Subject: [rtcweb] Use Case draft
>=20
> The chairs would like to ask the working group to focus on the use
> case draft.  If you have use cases that need to be added to the
> document or text changes you'd like to suggest, please send them in
> for discussion before May 15th.  After this round, we will look toward
> having a working group last call on the document (hopefully before the
> interim meeting).
>=20
> regards,
>=20
> Ted, Magnus, Cullen
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On 5/1/12 3:14 PM, Hutton, Andrew wrote:
> Hi,
> 
> A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements-07 contain the statement "It is essential that the communication cannot be eavesdropped" however there is no definition of what is actually meant by "eavesdropped" although I think we all have an idea of what it means.
> 
> Maybe it would be better to replace these statements with something that refers to wiretapping and RFC 2804 (RAVEN) which actually has a definition of wiretapping.

That's a core topic imho, especially because WebRTC would like to make
it appear like carrying on end-to-end security.

If WebRTC provide full end-to-end security no wiretapping should be
possible.

Because WebRTC, from the discussion done in this mailing lists, cannot
satisfy end-to-end security requirements (no wiretapping) in many cases,
then it's highly relevant that the user is aware if the connection is
secured peer-to-peer (so between the end-user-device of the parties that
are talking together) or if there's a gateway in the middle
decrypting/deciphering the call, and so potentially eavesdropping it.

I think that to avoid a "shame" on the reputation of WebRTC, it's very
and highly important to clearly state and explain how/when/who can do
eavesdropping.

At least to know "when" it's end-to-end security and when it's not, in a
clear, unique, identifiable way.

Fabio

From richard.ejzak@alcatel-lucent.com  Tue May  1 06:57:48 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: Justin Uberti <juberti@google.com>, "Nataraju A.B" <nataraju.sip@gmail.com>
Date: Tue, 1 May 2012 08:57:40 -0500
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi Justin, Partha,
The call flow from Partha is a very reasonable one in the presence of SIP f=
orking.  Let's assume that a first target responded with an SDP answer in a=
 183 that must be treated as a PRANSWER since other targets have not yet ha=
d a chance to respond.  Then the first target sends an UPDATE with offer2.

As far as SIP is concerned, there is no such thing as a PRANSWER.  The firs=
t target already responded with an SDP ANSWER and wants to UPDATE with a ne=
w offer.  This is perfectly legitimate and common SIP usage.  PRANSWER is a=
 local construct that should be usable by the application to support more c=
omplex offer/answer cases like forking.

You must respond to this offer by cloning the peer connection, closing the =
clone with an ANSWER identical to the previous PRANSWER and then applying o=
ffer2 to the clone (or just applying the offer2 directly with the understan=
ding that the previous transaction is closed).  The original peer connectio=
n must revert to the original OFFER state to allow for other targets to res=
pond.

Alternately, the original peer connection can support the first target and =
a new clone created if there is the potential for other forked responses to=
 the original offer.  Either way can be made to work.

This is a perfectly reasonable SIP use case that explains why we need to be=
 able to clone the peer connection.

Richard

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Justin Uberti
Sent: Monday, April 30, 2012 9:51 PM
To: Nataraju A.B
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

Yes, I was under the impression that SIP enforced this requirement, althoug=
h I am probably not aware of all the corner cases. Is there a real-world sc=
enario where this flow is required?
On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B <nataraju.sip@gmail.com<mail=
to:nataraju.sip@gmail.com>> wrote:
If the scenario considered is without reliable provisional responses. Then =
the first ANSWER must be in 200-INV and no more O/A allowed during INVITE(i=
nitial) transaction.

Basic requirement for reliable and unambiguous O/A is - At any point in tim=
e there could be only one O/A open.

Also only one O/A suggested during INVITE(initial) transaction.

For reference, rfc6337, list outs different O/A models...

<rfc6337>





            Offer                Answer             RFC    Ini Est Early

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

     1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N

     2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N

     3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N

     4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N

     5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y

     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y



          Table 1: Summary of SIP Usage of the Offer/Answer Model
</rfc6337>

Thanks,
Nataraju A B

On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi <pravindran@sonu=
snet.com<mailto:pravindran@sonusnet.com>> wrote:
Justin/Cullen,

Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:


Browser1-------------------------Browser2(SIP-JSEP gateway)
   |        SDP_OFFER(1)            |  INV with offer1
   |------------------------------->|--->
   |       SDP_PRANSWER(1)          |  183 with answer1
   |<-------------------------------|<---
   |       SDP_OFFER(2)             |   UPDATE with offer2
   |<-------------------------------|<---
   |       SDP_ANSWER(2?)           |
   |--------------------->?

The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).

Thanks
Partha

PS: For simplicity, PRACK message exchange is not chosen in SIP side.




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



--
Thanks,
Nataraju A.B.



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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi Justin, Partha,<o:p></o:p></span></=
font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The call flow from Partha is a very
reasonable one in the presence of SIP forking. &nbsp;Let&#8217;s assume tha=
t a
first target responded with an SDP answer in a 183 that must be treated as =
a
PRANSWER since other targets have not yet had a chance to respond.&nbsp; Th=
en
the first target sends an UPDATE with offer2.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>As far as SIP is concerned, there is n=
o
such thing as a PRANSWER. &nbsp;The first target already responded with an =
SDP
ANSWER and wants to UPDATE with a new offer. &nbsp;This is perfectly legiti=
mate
and common SIP usage.&nbsp; PRANSWER is a local construct that should be us=
able
by the application to support more complex offer/answer cases like forking.=
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>You must respond to this offer by clon=
ing
the peer connection, closing the clone with an ANSWER identical to the prev=
ious
PRANSWER and then applying offer2 to the clone (or just applying the offer2
directly with the understanding that the previous transaction is closed). &=
nbsp;The
original peer connection must revert to the original OFFER state to allow f=
or
other targets to respond.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Alternately, the original peer connect=
ion
can support the first target and a new clone created if there is the potent=
ial for
other forked responses to the original offer. &nbsp;Either way can be made =
to
work.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>This is a perfectly reasonable SIP use
case that explains why we need to be able to clone the peer connection.<o:p=
></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'>
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Justin Uberti<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, April 30, 2012=
 9:51
PM<br>
<b><span style=3D'font-weight:bold'>To:</span></b> Nataraju A.B<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] SDP_PR=
ANSWER
followed by SDP_OFFER scenario in JSEP</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Yes, I was under =
the
impression that SIP enforced this requirement, although I am probably not a=
ware
of all the corner cases. Is there a real-world scenario where this flow is
required?<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B &lt;<a
href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gmail=
.com</a>&gt;
wrote:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>If the scenario considered is without reliable provisional response=
s.
Then the first ANSWER must be in 200-INV and no more O/A allowed during INV=
ITE(initial)
transaction.<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Basic requirement for reliable and unambiguous O/A is -<b><span
style=3D'font-weight:bold'> At any point in time there could be only one O/=
A
open.&nbsp;</span></b><o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Also only one O/A suggested during INVITE(initial)&nbsp;transaction=
.<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>For reference, rfc6337, list outs different O/A models...<o:p></o:p=
></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>&lt;rfc6337&gt;<o:p></o:p></span></font></p>

</div>

<div><pre style=3D'word-wrap:break-word;white-space:pre-wrap'><font size=3D=
2
face=3D"Courier New"><span style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></sp=
an></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><o:p>&nbsp;<=
/o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Offer&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 Answer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; RFC&nbsp;&nbsp;&nbsp; Ini Est Early<o:p></o:p></span></font></pre><pre=
><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; --------------------------------------------------------------=
-----<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 1. INVITE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; 2xx INVITE Resp.&nbsp;&nbsp;&nbsp;&nbsp; RFC 3261&nbsp; Y&nbsp;&nbsp=
; Y&nbsp;&nbsp;&nbsp; N<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 2. 2xx INVITE Resp.&nbsp;&nbsp;&nbsp;&nbsp; ACK Req.&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3261&nbs=
p; Y&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; N<o:p></o:p></span></font></pre><pre><=
font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 3. INVITE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; 1xx-rel INVITE Resp. RFC 3262&nbsp; Y&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp=
; N<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 4. 1xx-rel INVITE Resp. PRACK Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3262&nbsp; Y&nbsp;&nbsp; Y&nbsp;&nbsp;=
&nbsp; N<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 5. PRACK Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; 200 PRACK Resp.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3262&nbsp; N&=
nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; Y<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp; 6. UPDATE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; 2xx UPDATE Resp.&nbsp;&nbsp;&nbsp;&nbsp; RFC 3311&nbsp; N&nbsp;&nbsp=
; Y&nbsp;&nbsp;&nbsp; Y<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'><o:p>&nbsp;<=
/o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span style=3D'font-size:10.0pt'>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Table 1: Summary of SIP Usage of=
 the Offer/Answer Model&nbsp;<o:p></o:p></span></font></pre></div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>&lt;/rfc6337&gt;<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Thanks,<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>Nataraju A B<o:p></o:p></span></font></p>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi &lt;<a
href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pravindran@sonusn=
et.com</a>&gt;
wrote:<o:p></o:p></span></font></p>

</div>

</div>

<blockquote style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;
margin-left:4.8pt;margin-right:0in'>

<div>

<div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>Justin/Cullen,<br=
>
<br>
Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies
with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected behavi=
or.
The exact callflow is as follows:<br>
<br>
<br>
Browser1-------------------------Browser2(SIP-JSEP gateway)<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;SDP_OFFER(1) &nbsp; &nbsp; &nbsp;
&nbsp; &nbsp; &nbsp;| &nbsp;INV with offer1<br>
&nbsp; &nbsp;|-------------------------------&gt;|---&gt;<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_PRANSWER(1) &nbsp; &nbsp; &nbsp; &n=
bsp;
&nbsp;| &nbsp;183 with answer1<br>
&nbsp; &nbsp;|&lt;-------------------------------|&lt;---<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_OFFER(2) &nbsp; &nbsp; &nbsp; &nbsp=
;
&nbsp; &nbsp; | &nbsp; UPDATE with offer2<br>
&nbsp; &nbsp;|&lt;-------------------------------|&lt;---<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_ANSWER(2?) &nbsp; &nbsp; &nbsp; &nb=
sp;
&nbsp; |<br>
&nbsp; &nbsp;|---------------------&gt;?<br>
<br>
The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in
Browser 2 (SIP-JSEP gateway).<br>
<br>
Thanks<br>
Partha<br>
<br>
PS: For simplicity, PRACK message exchange is not chosen in SIP side.<br>
<br>
<br>
<br>
<br>
<o:p></o:p></span></font></p>

</div>

</div>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>_______________________________________________<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><o:p></o:p></span></font></=
p>

</div>

</blockquote>

</div>

<p class=3DMsoNormal><font size=3D3 color=3D"#888888" face=3D"Times New Rom=
an"><span
style=3D'font-size:12.0pt;color:#888888'><br>
<br clear=3Dall>
<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 color=3D"#888888" face=3D"Times New Rom=
an"><span
style=3D'font-size:12.0pt;color:#888888'><o:p>&nbsp;</o:p></span></font></p=
>

</div>

<p class=3DMsoNormal><font size=3D3 color=3D"#888888" face=3D"Times New Rom=
an"><span
style=3D'font-size:12.0pt;color:#888888'>-- <br>
</span></font><font size=3D1 color=3D"#000099" face=3Dmonospace><span
style=3D'font-size:7.5pt;font-family:monospace;color:#000099'>Thanks,</span=
></font><font
color=3D"#888888"><span style=3D'color:#888888'><o:p></o:p></span></font></=
p>

<div>

<p class=3DMsoNormal><font size=3D1 color=3D"#000099" face=3Dmonospace><spa=
n
style=3D'font-size:7.5pt;font-family:monospace;color:#000099'>Nataraju A.B.=
</span></font><font
color=3D"#888888"><span style=3D'color:#888888'><o:p></o:p></span></font></=
p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

--_000_6F428EFD2B8C2F49A2FB1317291A76C1136029362AUSNAVSXCHMBSA_--

From ekr@rtfm.com  Tue May  1 07:05:41 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 1 May 2012 07:05:00 -0700
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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On Tue, May 1, 2012 at 6:14 AM, Hutton, Andrew
<andrew.hutton@siemens-enterprise.com> wrote:
> Hi,
>
> A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements=
-07 contain the statement "It is essential that the communication cannot be=
 eavesdropped" however there is no definition of what is actually meant by =
"eavesdropped" although I think we all have an idea of what it means.
>
> Maybe it would be better to replace these statements with something that =
refers to wiretapping and RFC 2804 (RAVEN) which actually has a definition =
of wiretapping.

This seems like it's creeping into the security requirements question.
Rather than try to make the use cases document more precise, I'd
prefer to have those statements be in draft-ietf-rtcweb-security,
which actually has (or at least is intended to) have fairly precise
descriptions of what the relevant security properties are.

-Ekr

From chris.cavigioli@intel.com  Tue May  1 07:14:44 2012
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From: "Cavigioli, Chris" <chris.cavigioli@intel.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft - Eavesdropping.
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Wording needs to be clarified.
- "Eavesdropping" has a negative connotation of someone inappropriately lis=
tening in to another communication.  This should not happen.
- "Lawful intercept" is something telecom service providers are required to=
 provide to law enforcement with all appropriate safeguards in place http:/=
/en.wikipedia.org/wiki/Lawful_interception=20
-chris

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Hutton, Andrew
Sent: Tuesday, May 01, 2012 6:14 AM
To: Ted Hardie; rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - Eavesdropping.

Hi,

A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements-0=
7 contain the statement "It is essential that the communication cannot be e=
avesdropped" however there is no definition of what is actually meant by "e=
avesdropped" although I think we all have an idea of what it means.

Maybe it would be better to replace these statements with something that re=
fers to wiretapping and RFC 2804 (RAVEN) which actually has a definition of=
 wiretapping.

Regards
Andy




> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ted Hardie
> Sent: 27 April 2012 17:15
> To: rtcweb@ietf.org
> Subject: [rtcweb] Use Case draft
>=20
> The chairs would like to ask the working group to focus on the use
> case draft.  If you have use cases that need to be added to the
> document or text changes you'd like to suggest, please send them in
> for discussion before May 15th.  After this round, we will look toward
> having a working group last call on the document (hopefully before the
> interim meeting).
>=20
> regards,
>=20
> Ted, Magnus, Cullen
> _______________________________________________
> 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 marshall.eubanks@gmail.com  Tue May  1 07:35:20 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On Tue, May 1, 2012 at 10:14 AM, Cavigioli, Chris
<chris.cavigioli@intel.com> wrote:
> Wording needs to be clarified.
> - "Eavesdropping" has a negative connotation of someone inappropriately l=
istening in to another communication. =A0This should not happen.
> - "Lawful intercept" is something telecom service providers are required =
to provide to law enforcement with all appropriate safeguards in place http=
://en.wikipedia.org/wiki/Lawful_interception

Lawful intercept is a legal term, which in my opinion the IETF should
stay away from.  You will go into the weeds amazing fast otherwise.

I prefer "monitoring," which can be done for many reasons (see the
parallel discussion on enterprise call centers).

Regards
Marshall

> -chris
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of Hutton, Andrew
> Sent: Tuesday, May 01, 2012 6:14 AM
> To: Ted Hardie; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
>
> Hi,
>
> A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements=
-07 contain the statement "It is essential that the communication cannot be=
 eavesdropped" however there is no definition of what is actually meant by =
"eavesdropped" although I think we all have an idea of what it means.
>
> Maybe it would be better to replace these statements with something that =
refers to wiretapping and RFC 2804 (RAVEN) which actually has a definition =
of wiretapping.
>
> Regards
> Andy
>
>
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Ted Hardie
>> Sent: 27 April 2012 17:15
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Use Case draft
>>
>> The chairs would like to ask the working group to focus on the use
>> case draft. =A0If you have use cases that need to be added to the
>> document or text changes you'd like to suggest, please send them in
>> for discussion before May 15th. =A0After this round, we will look toward
>> having a working group last call on the document (hopefully before the
>> interim meeting).
>>
>> regards,
>>
>> Ted, Magnus, Cullen
>> _______________________________________________
>> 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 randell-ietf@jesup.org  Tue May  1 08:57:10 2012
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Date: Tue, 01 May 2012 11:56:11 -0400
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Use Case draft
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On 5/1/2012 1:06 AM, Stefan Hakansson LK wrote:
> On 04/30/2012 11:00 PM, Randell Jesup wrote:
>> The game case can need both reliable and unreliable data at the same
>> time, which gets us reliable, unreliable and multiple streams.
>
> You're right; but currently only game state updates is mentioned.
> Probably that use case should be updated.

Game state updates for a real online game can involve both reliable and 
unreliable data.  Some state can be out-of-date (player positions, 
visual state issues (lighting), etc) and the game will extrapolate 
positions; others (global critical state (live, dead, etc)) must be 
reliable.

-- 
Randell Jesup
randell-ietf@jesup.org

From nataraju.sip@gmail.com  Tue May  1 09:12:31 2012
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--f46d0407152953fddc04befbd608
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The original scenario mentioned by Partha is quite logical and acceptable,
since there is only one O/A open all the times.

If not handled properly, this additional UPDATE (with OFFER2) during the
lifetime of INVITE transaction lead to ambiguities. Hence rfc6337, don't
support this as a valid O/A.

On Tue, May 1, 2012 at 7:27 PM, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

>  Hi Justin, Partha,****
>
> The call flow from Partha is a very reasonable one in the presence of SIP
> forking.  Let=92s assume that a first target responded with an SDP answer=
 in
> a 183 that must be treated as a PRANSWER since other targets have not yet
> had a chance to respond.  Then the first target sends an UPDATE with offe=
r2.
> ****
>
> ** **
>
> As far as SIP is concerned, there is no such thing as a PRANSWER.  The
> first target already responded with an SDP ANSWER and wants to UPDATE wit=
h
> a new offer.  This is perfectly legitimate and common SIP usage.  PRANSWE=
R
> is a local construct that should be usable by the application to support
> more complex offer/answer cases like forking.****
>
> ** **
>
> You must respond to this offer by cloning the peer connection, closing th=
e
> clone with an ANSWER identical to the previous PRANSWER and then applying
> offer2 to the clone (or just applying the offer2 directly with the
> understanding that the previous transaction is closed).  The original pee=
r
> connection must revert to the original OFFER state to allow for other
> targets to respond.****
>
> ** **
>
> Alternately, the original peer connection can support the first target an=
d
> a new clone created if there is the potential for other forked responses =
to
> the original offer.  Either way can be made to work.****
>
> ** **
>
> This is a perfectly reasonable SIP use case that explains why we need to
> be able to clone the peer connection.****
>
> ** **
>
> Richard****
>
> ** **
>  ------------------------------
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Justin Uberti
> *Sent:* Monday, April 30, 2012 9:51 PM
> *To:* Nataraju A.B
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in
> JSEP****
>
> ** **
>
> Yes, I was under the impression that SIP enforced this requirement,
> although I am probably not aware of all the corner cases. Is there a
> real-world scenario where this flow is required?****
>
> On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B <nataraju.sip@gmail.com>
> wrote:****
>
> If the scenario considered is without reliable provisional responses. The=
n
> the first ANSWER must be in 200-INV and no more O/A allowed during
> INVITE(initial) transaction.****
>
> ** **
>
> Basic requirement for reliable and unambiguous O/A is -* At any point in
> time there could be only one O/A open. *****
>
> ** **
>
> Also only one O/A suggested during INVITE(initial) transaction.****
>
> ** **
>
> For reference, rfc6337, list outs different O/A models...****
>
> ** **
>
> <rfc6337>****
>
> ** **
>
> ** **
>
>             Offer                Answer             RFC    Ini Est Early*=
***
>
>      -------------------------------------------------------------------*=
***
>
>      1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N***=
*
>
>      2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N***=
*
>
>      3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N***=
*
>
>      4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N***=
*
>
>      5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y***=
*
>
>      6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y***=
*
>
> ** **
>
>           Table 1: Summary of SIP Usage of the Offer/Answer Model ****
>
>  </rfc6337>****
>
> ** **
>
> Thanks,****
>
> Nataraju A B****
>
> ** **
>
> On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi <
> pravindran@sonusnet.com> wrote:****
>
>  Justin/Cullen,
>
> Could you please explain in case for an SDP_OFFER(1), the remote entity
> replies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the
> expected behavior. The exact callflow is as follows:
>
>
> Browser1-------------------------Browser2(SIP-JSEP gateway)
>    |        SDP_OFFER(1)            |  INV with offer1
>    |------------------------------->|--->
>    |       SDP_PRANSWER(1)          |  183 with answer1
>    |<-------------------------------|<---
>    |       SDP_OFFER(2)             |   UPDATE with offer2
>    |<-------------------------------|<---
>    |       SDP_ANSWER(2?)           |
>    |--------------------->?
>
> The reason for this O/A callflow is due to UPDATE method (RFC 3311)
> mapping in Browser 2 (SIP-JSEP gateway).
>
> Thanks
> Partha
>
> PS: For simplicity, PRACK message exchange is not chosen in SIP side.
>
>
>
>
> ****
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb****
>
>
>
> ****
>
> ** **
>
> --
> Thanks,****
>
> Nataraju A.B.****
>
> ** **
>
> ** **
>



--=20
Thanks,
Nataraju A.B.

--f46d0407152953fddc04befbd608
Content-Type: text/html; charset=windows-1252
Content-Transfer-Encoding: quoted-printable

The original scenario mentioned by Partha is quite logical and acceptable, =
since there is only one O/A open all the times.<div><br></div><div>If not h=
andled properly, this additional UPDATE (with OFFER2) during the lifetime o=
f INVITE transaction lead to ambiguities. Hence=A0rfc6337,=A0don&#39;t supp=
ort this as a valid O/A.</div>
<div><br><div class=3D"gmail_quote">On Tue, May 1, 2012 at 7:27 PM, Ejzak, =
Richard P (Richard) <span dir=3D"ltr">&lt;<a href=3D"mailto:richard.ejzak@a=
lcatel-lucent.com" target=3D"_blank">richard.ejzak@alcatel-lucent.com</a>&g=
t;</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 lang=3D"EN-US" link=3D"blue" vlink=3D"blue">

<div>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Hi Justin, Partha,<u></u><u><=
/u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">The call flow from Partha is =
a very
reasonable one in the presence of SIP forking. =A0Let=92s assume that a
first target responded with an SDP answer in a 183 that must be treated as =
a
PRANSWER since other targets have not yet had a chance to respond.=A0 Then
the first target sends an UPDATE with offer2.<u></u><u></u></span></font></=
p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">As far as SIP is concerned, t=
here is no
such thing as a PRANSWER. =A0The first target already responded with an SDP
ANSWER and wants to UPDATE with a new offer. =A0This is perfectly legitimat=
e
and common SIP usage.=A0 PRANSWER is a local construct that should be usabl=
e
by the application to support more complex offer/answer cases like forking.=
<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">You must respond to this offe=
r by cloning
the peer connection, closing the clone with an ANSWER identical to the prev=
ious
PRANSWER and then applying offer2 to the clone (or just applying the offer2
directly with the understanding that the previous transaction is closed). =
=A0The
original peer connection must revert to the original OFFER state to allow f=
or
other targets to respond.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Alternately, the original pee=
r connection
can support the first target and a new clone created if there is the potent=
ial for
other forked responses to the original offer. =A0Either way can be made to
work.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">This is a perfectly reasonabl=
e SIP use
case that explains why we need to be able to clone the peer connection.<u><=
/u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Richard<u></u><u></u></span><=
/font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></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">

<hr size=3D"2" width=3D"100%" align=3D"center">

</span></font></div>

<p class=3D"MsoNormal"><b><font face=3D"Tahoma"><span style=3D"font-size:10=
.0pt;font-family:Tahoma;font-weight:bold">From:</span></font></b><font face=
=3D"Tahoma"><span style=3D"font-size:10.0pt;font-family:Tahoma">
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"=
_blank">rtcweb-bounces@ietf.org</a>] <b><span style=3D"font-weight:bold">On=
 Behalf Of </span></b>Justin Uberti<br>

<b><span style=3D"font-weight:bold">Sent:</span></b> Monday, April 30, 2012=
 9:51
PM<br>
<b><span style=3D"font-weight:bold">To:</span></b> Nataraju A.B<br>
<b><span style=3D"font-weight:bold">Cc:</span></b> <a href=3D"mailto:rtcweb=
@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>
<b><span style=3D"font-weight:bold">Subject:</span></b> Re: [rtcweb] SDP_PR=
ANSWER
followed by SDP_OFFER scenario in JSEP</span></font><u></u><u></u></p>

</div><div><div class=3D"h5">

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><font size=3D"3" face=
=3D"Times New Roman"><span style=3D"font-size:12.0pt">Yes, I was under the
impression that SIP enforced this requirement, although I am probably not a=
ware
of all the corner cases. Is there a real-world scenario where this flow is
required?<u></u><u></u></span></font></p>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B &lt;<=
a href=3D"mailto:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gma=
il.com</a>&gt;
wrote:<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">If the scenario considered is without reliable provi=
sional responses.
Then the first ANSWER must be in 200-INV and no more O/A allowed during INV=
ITE(initial)
transaction.<u></u><u></u></span></font></p>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">Basic requirement for reliable and unambiguous O/A i=
s -<b><span style=3D"font-weight:bold"> At any point in time there could be=
 only one O/A
open.=A0</span></b><u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">Also only one O/A suggested during INVITE(initial)=
=A0transaction.<u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">For reference, rfc6337, list outs different O/A mode=
ls...<u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">&lt;rfc6337&gt;<u></u><u></u></span></font></p>

</div>

<div><pre style=3D"word-wrap:break-word;white-space:pre-wrap"><font face=3D=
"Courier New"><span style=3D"font-size:10.0pt"><u></u>=A0<u></u></span></fo=
nt></pre><pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt"><=
u></u>=A0<u></u></span></font></pre>
<pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=
=A0=A0=A0=A0=A0=A0=A0=A0 Offer=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
 Answer=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 RFC=A0=A0=A0 Ini Est Early<u></=
u><u></u></span></font></pre><pre><font face=3D"Courier New"><span style=3D=
"font-size:10.0pt">=A0=A0=A0=A0 -------------------------------------------=
------------------------<u></u><u></u></span></font></pre>
<pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=
=A0 1. INVITE Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0 2xx INVITE Resp.=A0=A0=A0=A0 =
RFC 3261=A0 Y=A0=A0 Y=A0=A0=A0 N<u></u><u></u></span></font></pre><pre><fon=
t face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=A0 2. 2xx=
 INVITE Resp.=A0=A0=A0=A0 ACK Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 RFC =
3261=A0 Y=A0=A0 Y=A0=A0=A0 N<u></u><u></u></span></font></pre>
<pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=
=A0 3. INVITE Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0 1xx-rel INVITE Resp. RFC 3262=
=A0 Y=A0=A0 Y=A0=A0=A0 N<u></u><u></u></span></font></pre><pre><font face=
=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=A0 4. 1xx-rel I=
NVITE Resp. PRACK Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 RFC 3262=A0 Y=A0=A0 Y=
=A0=A0=A0 N<u></u><u></u></span></font></pre>
<pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=
=A0 5. PRACK Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 200 PRACK Resp.=A0=A0=A0=A0=
=A0 RFC 3262=A0 N=A0=A0 Y=A0=A0=A0 Y<u></u><u></u></span></font></pre><pre>=
<font face=3D"Courier New"><span style=3D"font-size:10.0pt">=A0=A0=A0=A0 6.=
 UPDATE Req.=A0=A0=A0=A0=A0=A0=A0=A0=A0 2xx UPDATE Resp.=A0=A0=A0=A0 RFC 33=
11=A0 N=A0=A0 Y=A0=A0=A0 Y<u></u><u></u></span></font></pre>
<pre><font face=3D"Courier New"><span style=3D"font-size:10.0pt"><u></u>=A0=
<u></u></span></font></pre><pre><font face=3D"Courier New"><span style=3D"f=
ont-size:10.0pt">=A0=A0=A0=A0=A0=A0=A0=A0=A0 Table 1: Summary of SIP Usage =
of the Offer/Answer Model=A0<u></u><u></u></span></font></pre>
</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">&lt;/rfc6337&gt;<u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">Thanks,<u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">Nataraju A B<u></u><u></u></span></font></p>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

<div>

<div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthas=
arathi &lt;<a href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pra=
vindran@sonusnet.com</a>&gt;
wrote:<u></u><u></u></span></font></p>

</div>

</div>

<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">

<div>

<div>

<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><font size=3D"3" face=
=3D"Times New Roman"><span style=3D"font-size:12.0pt">Justin/Cullen,<br>
<br>
Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies
with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected behavi=
or.
The exact callflow is as follows:<br>
<br>
<br>
Browser1-------------------------Browser2(SIP-JSEP gateway)<br>
=A0 =A0| =A0 =A0 =A0 =A0SDP_OFFER(1) =A0 =A0 =A0
=A0 =A0 =A0| =A0INV with offer1<br>
=A0 =A0|-------------------------------&gt;|---&gt;<br>
=A0 =A0| =A0 =A0 =A0 SDP_PRANSWER(1) =A0 =A0 =A0 =A0
=A0| =A0183 with answer1<br>
=A0 =A0|&lt;-------------------------------|&lt;---<br>
=A0 =A0| =A0 =A0 =A0 SDP_OFFER(2) =A0 =A0 =A0 =A0
=A0 =A0 | =A0 UPDATE with offer2<br>
=A0 =A0|&lt;-------------------------------|&lt;---<br>
=A0 =A0| =A0 =A0 =A0 SDP_ANSWER(2?) =A0 =A0 =A0 =A0
=A0 |<br>
=A0 =A0|---------------------&gt;?<br>
<br>
The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in
Browser 2 (SIP-JSEP gateway).<br>
<br>
Thanks<br>
Partha<br>
<br>
PS: For simplicity, PRACK message exchange is not chosen in SIP side.<br>
<br>
<br>
<br>
<br>
<u></u><u></u></span></font></p>

</div>

</div>

<div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt">_______________________________________________<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><u></u><u></u></span></font=
></p>

</div>

</blockquote>

</div>

<p class=3D"MsoNormal"><font size=3D"3" color=3D"#888888" face=3D"Times New=
 Roman"><span style=3D"font-size:12.0pt;color:#888888"><br>
<br clear=3D"all">
<u></u><u></u></span></font></p>

<div>

<p class=3D"MsoNormal"><font size=3D"3" color=3D"#888888" face=3D"Times New=
 Roman"><span style=3D"font-size:12.0pt;color:#888888"><u></u>=A0<u></u></s=
pan></font></p>

</div>

<p class=3D"MsoNormal"><font size=3D"3" color=3D"#888888" face=3D"Times New=
 Roman"><span style=3D"font-size:12.0pt;color:#888888">-- <br>
</span></font><font size=3D"1" color=3D"#000099" face=3D"monospace"><span s=
tyle=3D"font-size:7.5pt;font-family:monospace;color:#000099">Thanks,</span>=
</font><font color=3D"#888888"><span style=3D"color:#888888"><u></u><u></u>=
</span></font></p>


<div>

<p class=3D"MsoNormal"><font size=3D"1" color=3D"#000099" face=3D"monospace=
"><span style=3D"font-size:7.5pt;font-family:monospace;color:#000099">Natar=
aju A.B.</span></font><font color=3D"#888888"><span style=3D"color:#888888"=
><u></u><u></u></span></font></p>


</div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div>

</div>

<p class=3D"MsoNormal"><font size=3D"3" face=3D"Times New Roman"><span styl=
e=3D"font-size:12.0pt"><u></u>=A0<u></u></span></font></p>

</div></div></div>

</div>


</blockquote></div><br><br clear=3D"all"><div><br></div>-- <br><font color=
=3D"#000099"><font face=3D"&#39;courier new&#39;, monospace" size=3D"1">Tha=
nks,</font></font><div><font color=3D"#000099"><font face=3D"&#39;courier n=
ew&#39;, monospace" size=3D"1">Nataraju A.B.</font></font></div>
<br>
</div>

--f46d0407152953fddc04befbd608--

From randell-ietf@jesup.org  Tue May  1 09:23:39 2012
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Date: Tue, 01 May 2012 12:22:40 -0400
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On 5/1/2012 10:35 AM, Marshall Eubanks wrote:
> On Tue, May 1, 2012 at 10:14 AM, Cavigioli, Chris
> <chris.cavigioli@intel.com>  wrote:
>> Wording needs to be clarified.
>> - "Eavesdropping" has a negative connotation of someone inappropriately listening in to another communication.  This should not happen.
>> - "Lawful intercept" is something telecom service providers are required to provide to law enforcement with all appropriate safeguards in place http://en.wikipedia.org/wiki/Lawful_interception
>
> Lawful intercept is a legal term, which in my opinion the IETF should
> stay away from.  You will go into the weeds amazing fast otherwise.
>
> I prefer "monitoring," which can be done for many reasons (see the
> parallel discussion on enterprise call centers).

I'm fine with discussing call center monitoring, and believe the 
authentication should be to "Key Bank" or "Key Bank Customer Service", 
not to "station 27 Boise Key Bank Call Center", which implies the WebRTC 
connection would terminate at the call center 'PBX'-equivalent and then 
have media (etc) routed (even via a second WebRTC connection) to the 
agent.  This means monitoring (and for financial firms, legally-required 
logging, etc) can occur via the PBX-equivalent (aka WebRTC Proxy).

We have explicitly avoided discussion of "Lawful intercept", which would 
involve complex (changing) legal requirements from every major 
jurisdiction, and we should continue to avoid wrestling with that 
python.  :-)

-- 
Randell Jesup
randell-ietf@jesup.org

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Nataraju A.B <nataraju.sip@gmail.com>, "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, Justin Uberti <juberti@google.com>
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Nataraju,

RFC 6337 supports early dialog UPDATE scenario which I mentioned in this ma=
il. The exact RFC
snippet with "Early column" explanation.


            Offer                Answer             RFC    Ini Est Early

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

     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y

   The 'Early' column indicates which patterns may be used to modify the
   established session in an early dialog.  There are two ways to
   exchange a subsequent offer/answer in an early dialog.

Here, UPDATE MUST not be used for initiating the session but it shall be us=
ed in the early dialog.
AFAIK, UPDATE With SDP is mostly useful in early-dialog as RE-INVITE with S=
DP is possible in the
mid-dialog ("Est")

Justin,

Richard provided some of the usecase using forked responses.  This usecase =
is not the corner
case and it is very normal scenario in SIP O/A. I'll add further usecase fo=
r this scenario:

One of the main usecase of UPDATE with SDP in the early dialog is remote ri=
ngback tone (RRBT)
from media server Usecase. Multiple scenario involved in this usecase:


1)      First usecase, the first offer/answer is completed between  Origina=
ting endpoint and
        media server, UPDATE with second offer is sent towards Originating =
endpoint to update
        SDP with terminating Endpoint.

2)      Second usecase, the first offer/answer is completed with direction =
attribute as

Sendonly in the answer by which early dialog is only for remote ringback to=
ne and  not

actual session establishment.  second offer in UPDATE is updated the direct=
ion attribute

as "sendrecv".

3)      Combination of First and second usecase wherein media server answer=
 with sendonly,

UPDATE with SDP for "sendrecv".


Common Topology:

Originating
Browser--------------Gateway---------SoftSwitch/PBX(B2BUA)-------SIP UA (Me=
dia Server/IVR)
                                                                           =
           |
                                                                           =
           -------------------SIP UA (IP phone)

RFC 6337 has lot of other corner scenario. Could you please let me know how=
 to map the second
OFFER and in PRANSWER state of the Originating endpoint (webbrowser in JSEP=
).

In case JSEP O/A is based on OFFER & ANSWER states only, this scenario will=
 not hit.

Thanks
Partha




From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Nataraju A.B
Sent: Tuesday, May 01, 2012 9:42 PM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

The original scenario mentioned by Partha is quite logical and acceptable, =
since there is only one O/A open all the times.

If not handled properly, this additional UPDATE (with OFFER2) during the li=
fetime of INVITE transaction lead to ambiguities. Hence rfc6337, don't supp=
ort this as a valid O/A.

On Tue, May 1, 2012 at 7:27 PM, Ejzak, Richard P (Richard) <richard.ejzak@a=
lcatel-lucent.com<mailto:richard.ejzak@alcatel-lucent.com>> wrote:
Hi Justin, Partha,
The call flow from Partha is a very reasonable one in the presence of SIP f=
orking.  Let's assume that a first target responded with an SDP answer in a=
 183 that must be treated as a PRANSWER since other targets have not yet ha=
d a chance to respond.  Then the first target sends an UPDATE with offer2.

As far as SIP is concerned, there is no such thing as a PRANSWER.  The firs=
t target already responded with an SDP ANSWER and wants to UPDATE with a ne=
w offer.  This is perfectly legitimate and common SIP usage.  PRANSWER is a=
 local construct that should be usable by the application to support more c=
omplex offer/answer cases like forking.

You must respond to this offer by cloning the peer connection, closing the =
clone with an ANSWER identical to the previous PRANSWER and then applying o=
ffer2 to the clone (or just applying the offer2 directly with the understan=
ding that the previous transaction is closed).  The original peer connectio=
n must revert to the original OFFER state to allow for other targets to res=
pond.

Alternately, the original peer connection can support the first target and =
a new clone created if there is the potential for other forked responses to=
 the original offer.  Either way can be made to work.

This is a perfectly reasonable SIP use case that explains why we need to be=
 able to clone the peer connection.

Richard

________________________________
From: rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org> [mailto:rtcwe=
b-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>] On Behalf Of Justin Ube=
rti
Sent: Monday, April 30, 2012 9:51 PM
To: Nataraju A.B
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

Yes, I was under the impression that SIP enforced this requirement, althoug=
h I am probably not aware of all the corner cases. Is there a real-world sc=
enario where this flow is required?
On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B <nataraju.sip@gmail.com<mail=
to:nataraju.sip@gmail.com>> wrote:
If the scenario considered is without reliable provisional responses. Then =
the first ANSWER must be in 200-INV and no more O/A allowed during INVITE(i=
nitial) transaction.

Basic requirement for reliable and unambiguous O/A is - At any point in tim=
e there could be only one O/A open.

Also only one O/A suggested during INVITE(initial) transaction.

For reference, rfc6337, list outs different O/A models...

<rfc6337>





            Offer                Answer             RFC    Ini Est Early

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

     1. INVITE Req.          2xx INVITE Resp.     RFC 3261  Y   Y    N

     2. 2xx INVITE Resp.     ACK Req.             RFC 3261  Y   Y    N

     3. INVITE Req.          1xx-rel INVITE Resp. RFC 3262  Y   Y    N

     4. 1xx-rel INVITE Resp. PRACK Req.           RFC 3262  Y   Y    N

     5. PRACK Req.           200 PRACK Resp.      RFC 3262  N   Y    Y

     6. UPDATE Req.          2xx UPDATE Resp.     RFC 3311  N   Y    Y



          Table 1: Summary of SIP Usage of the Offer/Answer Model
</rfc6337>

Thanks,
Nataraju A B

On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi <pravindran@sonu=
snet.com<mailto:pravindran@sonusnet.com>> wrote:
Justin/Cullen,

Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:


Browser1-------------------------Browser2(SIP-JSEP gateway)
   |        SDP_OFFER(1)            |  INV with offer1
   |------------------------------->|--->
   |       SDP_PRANSWER(1)          |  183 with answer1
   |<-------------------------------|<---
   |       SDP_OFFER(2)             |   UPDATE with offer2
   |<-------------------------------|<---
   |       SDP_ANSWER(2?)           |
   |--------------------->?

The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).

Thanks
Partha

PS: For simplicity, PRACK message exchange is not chosen in SIP side.



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



--
Thanks,
Nataraju A.B.





--
Thanks,
Nataraju A.B.


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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">Nataraju,<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">RFC 6337 supports early d=
ialog UPDATE scenario which I mentioned in this mail. The exact RFC
<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">snippet with &#8220;Early=
 column&#8221; explanation.<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>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Off=
er&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; Answer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; RFC&nbsp;&nbsp;&nbsp; Ini Est Early<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; ---------------------------------------------=
----------------------<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 6. UPDATE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 2xx UPDATE Resp.&nbsp;&nbsp;&nbsp;&nbsp; RFC 3311&n=
bsp; N&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; Y<o:p></o:p></pre>
<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:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; The 'Early' column indicates which patterns m=
ay be used to modify the<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; established session in an early dialog.&nbsp;=
 There are two ways to<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Co=
urier New&quot;">&nbsp;&nbsp; exchange a subsequent offer/answer in an earl=
y dialog.<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:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Here, UPDATE MUST not be =
used for initiating the session but it shall be used in the early dialog.<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">AFAIK, UPDATE With SDP is=
 mostly useful in early-dialog as RE-INVITE with SDP is possible in the
<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">mid-dialog (&#8220;Est&#8=
221;)<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:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Justin,<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">Richard provided some of =
the usecase using forked responses. &nbsp;This usecase is not the corner
<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">case and it is very norma=
l scenario in SIP O/A. I&#8217;ll add further usecase for this scenario:<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">One of the main usecase o=
f UPDATE with SDP in the early dialog is remote ringback tone (RRBT)<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">from media server Usecase=
. Multiple scenario involved in this usecase:&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;
<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"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">1)<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">First usecase, th=
e first offer/answer is completed between &nbsp;Originating endpoint and<o:=
p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:.25in"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &nbsp;media server, UPDATE with sec=
ond offer is sent towards Originating endpoint to update
<o:p></o:p></span></p>
<p class=3D"MsoNormal" style=3D"margin-left:.25in"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497=
D">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;SDP with terminating End=
point.
<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">2)<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Second usecase, t=
he first offer/answer is completed with direction attribute as
<o:p></o:p></span></p>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Sendonly in the an=
swer by which early dialog is only for remote ringback tone and&nbsp; not<o=
:p></o:p></span></p>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">actual session est=
ablishment. &nbsp;second offer in UPDATE is updated the direction attribute=
<o:p></o:p></span></p>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">as &#8220;sendrecv=
&#8221;.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-.25in;mso-list:l0 level=
1 lfo1"><![if !supportLists]><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><span style=3D"mso-=
list:Ignore">3)<span style=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">Combination of Fi=
rst and second usecase wherein media server answer with sendonly,<o:p></o:p=
></span></p>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">UPDATE with SDP fo=
r &#8220;sendrecv&#8221;.<o:p></o:p></span></p>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&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">Common Topology:<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">Originating<o:p></o:p></s=
pan></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">Browser--------------Gate=
way---------SoftSwitch/PBX(B2BUA)-------SIP UA (Media Server/IVR)<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">&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;&=
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;
<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">&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;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;-------------------SIP UA (IP phon=
e)<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">RFC 6337 has lot of other=
 corner scenario. Could you please let me know how to map the second
<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">OFFER and in PRANSWER sta=
te of the Originating endpoint (webbrowser in JSEP).
<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">In case JSEP O/A is based=
 on OFFER &amp; ANSWER states only, this scenario will not hit.<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>
<p class=3D"MsoListParagraph"><span style=3D"font-size:11.0pt;font-family:&=
quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"><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: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"><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>Nataraju A.B<br>
<b>Sent:</b> Tuesday, May 01, 2012 9:42 PM<br>
<b>To:</b> Ejzak, Richard P (Richard)<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in=
 JSEP<o:p></o:p></span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">The original scenario mentioned by Partha is quite l=
ogical and acceptable, since there is only one O/A open all the times.<o:p>=
</o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<div>
<p class=3D"MsoNormal">If not handled properly, this additional UPDATE (wit=
h OFFER2) during the lifetime of INVITE transaction lead to ambiguities. He=
nce&nbsp;rfc6337,&nbsp;don't support this as a valid O/A.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal">On Tue, May 1, 2012 at 7:27 PM, Ejzak, Richard P (Ri=
chard) &lt;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com" target=3D"_b=
lank">richard.ejzak@alcatel-lucent.com</a>&gt; wrote:<o:p></o:p></p>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Hi Justin, Partha,</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">The call flow from Partha is a very reasonab=
le one in the presence of SIP forking. &nbsp;Let&#8217;s assume that
 a first target responded with an SDP answer in a 183 that must be treated =
as a PRANSWER since other targets have not yet had a chance to respond.&nbs=
p; Then the first target sends an UPDATE with offer2.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">As far as SIP is concerned, there is no such=
 thing as a PRANSWER. &nbsp;The first target already responded
 with an SDP ANSWER and wants to UPDATE with a new offer. &nbsp;This is per=
fectly legitimate and common SIP usage.&nbsp; PRANSWER is a local construct=
 that should be usable by the application to support more complex offer/ans=
wer cases like forking.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">You must respond to this offer by cloning th=
e peer connection, closing the clone with an ANSWER identical
 to the previous PRANSWER and then applying offer2 to the clone (or just ap=
plying the offer2 directly with the understanding that the previous transac=
tion is closed). &nbsp;The original peer connection must revert to the orig=
inal OFFER state to allow for other targets
 to respond.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Alternately, the original peer connection ca=
n support the first target and a new clone created if there
 is the potential for other forked responses to the original offer. &nbsp;E=
ither way can be made to work.</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">This is a perfectly reasonable SIP use case =
that explains why we need to be able to clone the peer connection.</span><o=
:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">Richard</span><o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:10.0pt;font-family:&quot;Arial&quot;,&quo=
t;sans-serif&quot;;color:navy">&nbsp;</span><o:p></o:p></p>
<div>
<div class=3D"MsoNormal" align=3D"center" style=3D"text-align:center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,=
&quot;sans-serif&quot;">From:</span></b><span style=3D"font-size:10.0pt;fon=
t-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"=
_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Justin Uberti<br>
<b>Sent:</b> Monday, April 30, 2012 9:51 PM<br>
<b>To:</b> Nataraju A.B<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in=
 JSEP</span><o:p></o:p></p>
</div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">Yes, I was under the impression that SIP enforced this requirement, alth=
ough I am probably not aware of all the corner cases. Is there a real-world=
 scenario where this flow is required?<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Mon, Apr 30, 2012 at 10:55 AM, Nataraju A.B &lt;<a href=3D"mail=
to:nataraju.sip@gmail.com" target=3D"_blank">nataraju.sip@gmail.com</a>&gt;=
 wrote:<o:p></o:p></p>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">If the scenario considered is without reliable provisional respons=
es. Then the first ANSWER must be in 200-INV and no more O/A allowed during=
 INVITE(initial) transaction.<o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Basic requirement for reliable and unambiguous O/A is -<b> At any =
point in time there could be only one O/A open.&nbsp;</b><o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Also only one O/A suggested during INVITE(initial)&nbsp;transactio=
n.<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">For reference, rfc6337, list outs different O/A models...<o:p></o:=
p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&lt;rfc6337&gt;<o:p></o:p></p>
</div>
<div>
<pre style=3D"word-wrap:break-word;white-space:pre-wrap">&nbsp;<o:p></o:p><=
/pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Off=
er&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp; Answer&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp; RFC&nbsp;&nbsp;&nbsp; Ini Est Early<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; ---------------------------------------------=
----------------------<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 1. INVITE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 2xx INVITE Resp.&nbsp;&nbsp;&nbsp;&nbsp; RFC 3261&n=
bsp; Y&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; N<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 2. 2xx INVITE Resp.&nbsp;&nbsp;&nbsp;&nbsp; A=
CK Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; RFC 3261&nbsp; Y&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; N<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 3. INVITE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 1xx-rel INVITE Resp. RFC 3262&nbsp; Y&nbsp;&nbsp; Y=
&nbsp;&nbsp;&nbsp; N<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 4. 1xx-rel INVITE Resp. PRACK Req.&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; RFC 3262&nbsp; Y&nbsp;&nb=
sp; Y&nbsp;&nbsp;&nbsp; N<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 5. PRACK Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; 200 PRACK Resp.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
RFC 3262&nbsp; N&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; Y<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp; 6. UPDATE Req.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; 2xx UPDATE Resp.&nbsp;&nbsp;&nbsp;&nbsp; RFC 3311&n=
bsp; N&nbsp;&nbsp; Y&nbsp;&nbsp;&nbsp; Y<o:p></o:p></pre>
<pre>&nbsp;<o:p></o:p></pre>
<pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Table 1: Summar=
y of SIP Usage of the Offer/Answer Model&nbsp;<o:p></o:p></pre>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&lt;/rfc6337&gt;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Thanks,<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">Nataraju A B<o:p></o:p></p>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
<div>
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">On Mon, Apr 30, 2012 at 12:31 PM, Ravindran, Parthasarathi &lt;<a =
href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pravindran@sonusn=
et.com</a>&gt; wrote:<o:p></o:p></p>
</div>
</div>
<blockquote style=3D"border:none;border-left:solid #CCCCCC 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0in;margin-=
bottom:5.0pt">
<div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;margin-bottom:12.0p=
t">Justin/Cullen,<br>
<br>
Could you please explain in case for an SDP_OFFER(1), the remote entity rep=
lies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the expected b=
ehavior. The exact callflow is as follows:<br>
<br>
<br>
Browser1-------------------------Browser2(SIP-JSEP gateway)<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; &nbsp;SDP_OFFER(1) &nbsp; &nbsp; &nbsp;=
 &nbsp; &nbsp; &nbsp;| &nbsp;INV with offer1<br>
&nbsp; &nbsp;|-------------------------------&gt;|---&gt;<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_PRANSWER(1) &nbsp; &nbsp; &nbsp; &n=
bsp; &nbsp;| &nbsp;183 with answer1<br>
&nbsp; &nbsp;|&lt;-------------------------------|&lt;---<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_OFFER(2) &nbsp; &nbsp; &nbsp; &nbsp=
; &nbsp; &nbsp; | &nbsp; UPDATE with offer2<br>
&nbsp; &nbsp;|&lt;-------------------------------|&lt;---<br>
&nbsp; &nbsp;| &nbsp; &nbsp; &nbsp; SDP_ANSWER(2?) &nbsp; &nbsp; &nbsp; &nb=
sp; &nbsp; |<br>
&nbsp; &nbsp;|---------------------&gt;?<br>
<br>
The reason for this O/A callflow is due to UPDATE method (RFC 3311) mapping=
 in Browser 2 (SIP-JSEP gateway).<br>
<br>
Thanks<br>
Partha<br>
<br>
PS: For simplicity, PRACK message exchange is not chosen in SIP side.<br>
<br>
<br>
<br>
<o:p></o:p></p>
</div>
</div>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">_______________________________________________<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><o:p></o:p></p>
</div>
</blockquote>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888"><br>
<br clear=3D"all">
</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">&nbsp;</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"color:#888888">--
<br>
</span><span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;c=
olor:#000099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto"><span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;=
;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto">&nbsp;<o:p></o:p></p>
</div>
</div>
</div>
</div>
</div>
<p class=3D"MsoNormal"><br>
<br clear=3D"all">
<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
<p class=3D"MsoNormal">-- <br>
<span style=3D"font-size:7.5pt;font-family:&quot;Courier New&quot;;color:#0=
00099">Thanks,</span><o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:7.5pt;font-family:&quot;Cou=
rier New&quot;;color:#000099">Nataraju A.B.</span><o:p></o:p></p>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</div>
</body>
</html>

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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On 05/01/2012 04:05 PM, Eric Rescorla wrote:
> On Tue, May 1, 2012 at 6:14 AM, Hutton, Andrew
> <andrew.hutton@siemens-enterprise.com>  wrote:
>> Hi,
>>
>> A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements-07 contain the statement "It is essential that the communication cannot be eavesdropped" however there is no definition of what is actually meant by "eavesdropped" although I think we all have an idea of what it means.
>>
>> Maybe it would be better to replace these statements with something that refers to wiretapping and RFC 2804 (RAVEN) which actually has a definition of wiretapping.
>
> This seems like it's creeping into the security requirements question.
> Rather than try to make the use cases document more precise, I'd
> prefer to have those statements be in draft-ietf-rtcweb-security,
> which actually has (or at least is intended to) have fairly precise
> descriptions of what the relevant security properties are.

As one of the editors of the use case doc, I agree.

The document was written at a time when everything was much vaguer, and 
we as a group had much much less insight, and this is reflected in the 
document. 'Eavesdropping' is one example, 'stream' is another (we now 
have things like MediaStream, MediaStreamTrack), and so on.

I think we should keep it that way rather than doing a major update; I 
think there is little gain in doing that.

Stefan

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


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Subject: Re: [rtcweb] Use Case draft
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On 05/01/2012 02:05 PM, Jim Barnett wrote:
> One way to describe the use case is to let the contact center's media
> server/gateway serve as the webRTC endpoint.  Then all the issues of
> call delivery, call monitoring, etc. disappear.  They are handled by
> application software that sits behind the webRTC endpoint.  The
> company I work for makes a good living selling software that deals
> with all these issues - including bathroom breaks - and that's how we
> would tend to think of this case.  To us, it's a new kind of
> call/connection coming into the contact center, which we translate
> into SIP at the border and then handle normally.
>
> It's not clear to me if this use case adds any extra requirements.

I think this is important to sort out. If the use case does not add any 
extra requirements, what's the point of adding it?

> We would just have to be careful not to assume that a webRTC endpoint
> is always a person/browser-based user agent.  It may seem a bit
> unsettling that the webRTC endpoint can distribute the call somewhere
> else and let others listen in, but as far as I can tell that is
> already the case.  If Bob calls Alice with full authentication and
> security, he can be sure that he is connected to Alice's user agent
> and that no one in between can listen in, but there's nothing
> stopping Alice from recording the audio, or forwarding it to a third
> party.  So Bob could in fact be talking to Mary if that's how Alice
> wants to arrange things (_behind_ her user agent).  In general, Bob
> is assured only that he is talking to someone Alice wants him to talk
> to, and that no one can snoop without Alice's permission.  That's
> very much the way things work with the call center - you are sure
> that you are 1) connected securely to your bank 2) talking to someone
> that the bank wants you to talk to 3) being recorded or snooped on
> only when the bank explicitly chooses to do so.
>
> - Jim
>
> -----Original Message----- From: rtcweb-bounces@ietf.org
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>
> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
> Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
>> Whether anybody has been successful in the past with this type of
>> use case is I think irrelevant.
>>
>>
>>
>> The enterprise call centre use case is I think a vital use case
>> because it is a scenario in which one user is only concerned that
>> they can securely reach an organization/domain and is not concerned
>> about the individual within that domain  that they communicate
>> with.  A suspect quite a large percentage of RTCWEB applications
>> will be like this and it is not covered in the current use case
>> draft.
>
> I agree that this is a very useful use case and one I think is going
> to get a lot of traction. There is a very solid business case for
> this.  However, I have a fair amount of experience with a video call
> center for a client, and it is not as simple as it might seem.
>
> The essence of course is that you get the next available person,
> i.e., it is anycast. Determining who the next available person is is
> not trivial, nor is error recovery. (If I call you, and you don't
> answer or the call drops or whatever,  I can leave a message or try
> later. If I call a help desk, and this happens, I want a new agent,
> ideally automatically.) Call forwarding (e.g., first tier to second
> tier technical support) is essential, and it may be anycast or
> directed. There are also some security oddities  - if I am connecting
> from home, I may need to authenticate, use a credit card, etc. If I
> am connecting from inside a store, and providing in store video
> technical support is big part of the market, then the store
> authenticates me off line and the call really should just be a button
> push, which implies that the store has previously authenticated some
> sort of master session. In addition, unlike most video calls, in the
> enterprise call center a supervisor may need to be able to monitor
> (i.e., watch) a call, and in some circumstances (financial or medical
> calls, for example) there will need to be third party recording. I
> believe that  these details would be different from the typical
> RTCWEB scenario.
>
> Also, there will be a temptation to do the anycasting by the
> techniques used to load balance servers in a data center, but I think
> that may not be sufficient. The call "center" may in fact be spread
> completely across the planet (daytime support in the US, nighttime
> support in India, for example) and be on multiple autonomous systems
> (and even from people's homes), which gives rise to some of the
> transport issues NVO3 may face, but without any opportunity for
> packet tagging. Plus, there will complicated rules about who can be
> selected next. RTCWEB shouldn't worry about the intricacies of
> bathroom break policies; these complexities should be dealt with by
> an enterprise-side database, which to me (together with some of the
> other issues above) suggests that this would probably benefit from
> API support.
>
> Regards Marshall
>
>
>>
>>
>>
>> So I think we need it.
>>
>>
>>
>> Regards
>>
>> Andy
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>> rtcweb@ietf.org
>>
>>
>> Subject: Re: [rtcweb] Use Case draft
>>
>>
>>
>> Without numbers it is impossible to argue, but, if we talk about
>> the perceived need, I disagree.  Think of the people who travel
>> abroad and cannot call the 800 number. (I routinely use Web
>> interface for calls when traveling.)
>>
>>
>>
>> I am all for  the use case, as described by Jim.
>>
>> Igor
>>
>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>
>> ...
>>
>> I can't tell you the actual numbers, but when presented with the
>> choice of calling a toll free number
>>
>> or clicking a button marked "free internet call" - almost no-one on
>> a real, busy site clicked the button.
>>
>> ( for every button click there were several orders of magnitude
>> more 0800 calls from that page).
>>
>>
>>
>>
>>
>> So from my perspective this is a legacy interop use case with
>> almost zero user acceptance.
>>
>>
>>
>> (as far as I can see no-one has made this use-case desirable in
>> practice yet.)
>>
>> Tim.
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> 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
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] use cases, F20 and encryption, SCTP - comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 04/30/2012 09:16 AM, Magnus Westerlund wrote:
> Hi,
>
> My suggestion is that the use case document should be clarified. And I
> think the starting point is that the data channel must have the same
> security requirements as RTP media, both are media from the point of the
> application.

(speaking as use case doc editor): makes a lot of sense to me.

>
> Cheers
>
> Magnus
>
>
> On 2012-04-27 18:48, Dan Wing wrote:
>>> The chairs would like to ask the working group to focus on the use
>>> case draft.  If you have use cases that need to be added to the
>>> document or text changes you'd like to suggest, please send them in
>>> for discussion before May 15th.  After this round, we will look
>>> toward having a working group last call on the document (hopefully
>>> before the interim meeting).
>>
>> A few comments on draft-ietf-rtcweb-use-cases-and-requirements-07:
>>
>>
>> 1. Requirement F20 states:
>>
>>     F20  It MUST be possible to protect streams from eavesdropping.
>>
>> Consensus in the room during my presentation to RTCWEB at IETF83 was that we
>> don't need to support un-encrypted media (RTP) at all, and that all media
>> would be SRTP.  Can that be captured in F20 by re-wording, or perhaps in a
>> new requirement if we can't reword F20?  If there is a need or desire to
>> validate that consensus on list, let's please ask the chairs to do that.
>>
>>
>> 2. I noticed there is no requirement that we have a baseline for how SRTP
>> media is keyed (although there is a baseline requirement for codecs).  This
>> is a critical requirement.  I suggest adding "The browser MUST support a
>> baseline SRTP keying mechanism."  We have not reached consensus on that
>> keying mechanism, but the requirement is real.
>>
>>
>> 3. I see the document restricts its scope to media streams in the
>> Introduction with:
>>
>>    "The document focuses on requirements related to real-time media
>>     streams.  Requirements related to privacy, signalling between the
>>     browser and web server etc. are currently not considered."
>>
>> However, RTCWEB is also supports a data communication between browsers.  I
>> am worried if we do not specify requirements for the data communication we
>> will have problems.  I believe the expectation is that if the audio/video
>> stream works, that the data communication stream also work.  We need to
>> capture requirements for the data communication stream somewhere:
>>
>>    - a requirement to support data communication
>>    - that the chosen data communication protocol supports multiple streams
>> (which is why SCTP was chosen over TCP)
>>    - for NAT/firewall traversal of the data communication protocol (which is
>> why SCTP-over-UDP was chosen and another reason TCP was not chosen)
>>    - for encrypting that data communication session
>>    - a requirement for SCTP-over-UDP to work when UDP is blocked (aligning
>> with the existing F29 for audio/video)
>>    - a requirement to do ICE connectivity checks prior to bringing up DTLS (I
>> don't know if that is really a requirement, but I recall it mentioned at the
>> RTCWEB interim in Mountain View).
>>
>> Based on the scoping of the draft-ietf-rtcweb-use-cases-and-requirements,
>> the omission of the data communication stream is intentional.  If not in
>> draft-ietf-rtcweb-use-cases-and-requirements, where can we capture the
>> requirements for the data communication stream?
>>
>> -d
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft
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Date: Wed, 2 May 2012 09:05:38 +0000
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Subject: Re: [rtcweb] Use Case draft
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Stefan,

This usecase add its own requirements from identity perspective as=20
I mentioned earlier.=20

WebRTC MUST allow "Anonymous" users secure session for call center
usecase. "Anonymous" User may be agent in the call center side or=20
customer who does not require Identity to start the session.

Thanks
Partha=20

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Stefan Hakansson LK
>Sent: Wednesday, May 02, 2012 2:16 PM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Use Case draft
>
>On 05/01/2012 02:05 PM, Jim Barnett wrote:
>> One way to describe the use case is to let the contact center's media
>> server/gateway serve as the webRTC endpoint.  Then all the issues of
>> call delivery, call monitoring, etc. disappear.  They are handled by
>> application software that sits behind the webRTC endpoint.  The
>> company I work for makes a good living selling software that deals
>> with all these issues - including bathroom breaks - and that's how we
>> would tend to think of this case.  To us, it's a new kind of
>> call/connection coming into the contact center, which we translate
>> into SIP at the border and then handle normally.
>>
>> It's not clear to me if this use case adds any extra requirements.
>
>I think this is important to sort out. If the use case does not add any
>extra requirements, what's the point of adding it?
>
>> We would just have to be careful not to assume that a webRTC endpoint
>> is always a person/browser-based user agent.  It may seem a bit
>> unsettling that the webRTC endpoint can distribute the call somewhere
>> else and let others listen in, but as far as I can tell that is
>> already the case.  If Bob calls Alice with full authentication and
>> security, he can be sure that he is connected to Alice's user agent
>> and that no one in between can listen in, but there's nothing stopping
>> Alice from recording the audio, or forwarding it to a third party.  So
>> Bob could in fact be talking to Mary if that's how Alice wants to
>> arrange things (_behind_ her user agent).  In general, Bob is assured
>> only that he is talking to someone Alice wants him to talk to, and
>> that no one can snoop without Alice's permission.  That's very much
>> the way things work with the call center - you are sure that you are
>> 1) connected securely to your bank 2) talking to someone that the bank
>> wants you to talk to 3) being recorded or snooped on only when the
>> bank explicitly chooses to do so.
>>
>> - Jim
>>
>> -----Original Message----- From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>
>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>> Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
>>> Whether anybody has been successful in the past with this type of use
>>> case is I think irrelevant.
>>>
>>>
>>>
>>> The enterprise call centre use case is I think a vital use case
>>> because it is a scenario in which one user is only concerned that
>>> they can securely reach an organization/domain and is not concerned
>>> about the individual within that domain  that they communicate with.
>>> A suspect quite a large percentage of RTCWEB applications will be
>>> like this and it is not covered in the current use case draft.
>>
>> I agree that this is a very useful use case and one I think is going
>> to get a lot of traction. There is a very solid business case for
>> this.  However, I have a fair amount of experience with a video call
>> center for a client, and it is not as simple as it might seem.
>>
>> The essence of course is that you get the next available person, i.e.,
>> it is anycast. Determining who the next available person is is not
>> trivial, nor is error recovery. (If I call you, and you don't answer
>> or the call drops or whatever,  I can leave a message or try later. If
>> I call a help desk, and this happens, I want a new agent, ideally
>> automatically.) Call forwarding (e.g., first tier to second tier
>> technical support) is essential, and it may be anycast or directed.
>> There are also some security oddities  - if I am connecting from home,
>> I may need to authenticate, use a credit card, etc. If I am connecting
>> from inside a store, and providing in store video technical support is
>> big part of the market, then the store authenticates me off line and
>> the call really should just be a button push, which implies that the
>> store has previously authenticated some sort of master session. In
>> addition, unlike most video calls, in the enterprise call center a
>> supervisor may need to be able to monitor (i.e., watch) a call, and in
>> some circumstances (financial or medical calls, for example) there
>> will need to be third party recording. I believe that  these details
>> would be different from the typical RTCWEB scenario.
>>
>> Also, there will be a temptation to do the anycasting by the
>> techniques used to load balance servers in a data center, but I think
>> that may not be sufficient. The call "center" may in fact be spread
>> completely across the planet (daytime support in the US, nighttime
>> support in India, for example) and be on multiple autonomous systems
>> (and even from people's homes), which gives rise to some of the
>> transport issues NVO3 may face, but without any opportunity for packet
>> tagging. Plus, there will complicated rules about who can be selected
>> next. RTCWEB shouldn't worry about the intricacies of bathroom break
>> policies; these complexities should be dealt with by an
>> enterprise-side database, which to me (together with some of the other
>> issues above) suggests that this would probably benefit from API
>> support.
>>
>> Regards Marshall
>>
>>
>>>
>>>
>>>
>>> So I think we need it.
>>>
>>>
>>>
>>> Regards
>>>
>>> Andy
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>> rtcweb@ietf.org
>>>
>>>
>>> Subject: Re: [rtcweb] Use Case draft
>>>
>>>
>>>
>>> Without numbers it is impossible to argue, but, if we talk about the
>>> perceived need, I disagree.  Think of the people who travel abroad
>>> and cannot call the 800 number. (I routinely use Web interface for
>>> calls when traveling.)
>>>
>>>
>>>
>>> I am all for  the use case, as described by Jim.
>>>
>>> Igor
>>>
>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>
>>> ...
>>>
>>> I can't tell you the actual numbers, but when presented with the
>>> choice of calling a toll free number
>>>
>>> or clicking a button marked "free internet call" - almost no-one on a
>>> real, busy site clicked the button.
>>>
>>> ( for every button click there were several orders of magnitude more
>>> 0800 calls from that page).
>>>
>>>
>>>
>>>
>>>
>>> So from my perspective this is a legacy interop use case with almost
>>> zero user acceptance.
>>>
>>>
>>>
>>> (as far as I can see no-one has made this use-case desirable in
>>> practice yet.)
>>>
>>> Tim.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> 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
>> _______________________________________________ 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 pravindran@sonusnet.com  Wed May  2 06:01:48 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, Justin Uberti <juberti@google.com>, Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi Charles,

UPDATE is allowed in the early dialog and I haven't shown in the callflow f=
or simplicity as I mentioned in PS of below mail as " PS: For simplicity, P=
RACK message exchange is not chosen in SIP side."

Thanks
Partha

>-----Original Message-----
>From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
>Sent: Monday, April 30, 2012 9:53 PM
>To: Ravindran, Parthasarathi; Justin Uberti; Cullen Jennings
>Cc: rtcweb@ietf.org
>Subject: RE: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in
>JSEP
>
>Hi Partha,
>
>I believe such a flow illegal, per RFC 3311:
>
>      o  If the UPDATE is being sent before the completion of the INVITE
>         transaction, and the initial INVITE contained an offer, the
>         UPDATE cannot be sent with an offer unless the callee has
>         generated an answer in a reliable provisional response, has
>         received a PRACK for that reliable provisional response, has
>         not received any requests (PRACK or UPDATE) with offers that it
>         has not answered, and has not sent any UPDATE requests
>         containing offers that have not been answered.
>
>Cheers,
>Charles
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Ravindran, Parthasarathi
>> Sent: Monday, April 30, 2012 12:02 AM
>> To: Justin Uberti; Cullen Jennings
>> Cc: rtcweb@ietf.org
>> Subject: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
>>
>> Justin/Cullen,
>>
>> Could you please explain in case for an SDP_OFFER(1), the remote
>entity
>> replies with SDP_PRANSWER followed by new SDP_OFFER (2) what is the
>> expected behavior. The exact callflow is as follows:
>>
>>
>> Browser1-------------------------Browser2(SIP-JSEP gateway)
>>     |        SDP_OFFER(1)            |  INV with offer1
>>     |------------------------------->|--->
>>     |       SDP_PRANSWER(1)          |  183 with answer1
>>     |<-------------------------------|<---
>>     |       SDP_OFFER(2)             |   UPDATE with offer2
>>     |<-------------------------------|<---
>>     |       SDP_ANSWER(2?)           |
>>     |--------------------->?
>>
>> The reason for this O/A callflow is due to UPDATE method (RFC 3311)
>> mapping in Browser 2 (SIP-JSEP gateway).
>>
>> Thanks
>> Partha
>>
>> PS: For simplicity, PRACK message exchange is not chosen in SIP side.
>>
>>
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Wed May  2 06:13:10 2012
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On 2012-05-01 05:02, Justin Uberti wrote:

> 
>     According to my Colleague Per-Erik you can use video.width and
>     video.height to set the resolution the video element should have.
> 
>     see
>     http://dev.w3.org/html5/spec/the-video-element.html#the-video-element
> 
>     Thus the code you presented should be something like.
> 
>     video.src = webkitURL.createObjectURL(s);
>     video.width = 300;
>     video.height = 300;
>     canvas.getContext("2d").drawImage(video, 0, 0, 300, 300);
> 
>     That would set the video objects size to a 300, 300 video resolution.
>     The drawImage call I have removed the part which selects a cropping of
>     the source video to automatically show the full video in the specified
>     area.
> 
>     The adaptation of the video will of course not happen instantly so until
>     that happens the video will be scaled to the specified width and height.
>     And the actual delivered resolution will depend on the COP request and
>     what the sender support.
> 
> 
> I think this is a reasonable way to approach the problem as long as a
> <video/> element stays in the path. But as Harald mentions we have
> already considered several usages where a <video/> would not be in the
> path, i.e. connecting a MediaStream to a file, or perhaps as input to
> another PeerConnection.

I think in both these cases there is a need to take the information from
these sinks about what they request/desire in resolution and apply that
towards upstream. When it comes to relaying, this is no different what a
conference mixer will need to do in a multi-party session. Take in the
request from the multiple consumers of the media. Merge them into the
most suitable request and then send that to the media source.

For recording, which I don't think there exist a standardized interface
yet, it will need to indicate at what quality level the recording should
happen. I know that Per-Erik commented on this back in 2010, see:
http://lists.whatwg.org/pipermail/whatwg-whatwg.org/2010-November/029069.html

To my understanding this discussion was part of what resulted in the
removal of the proposed recorder interface.


> 
> Perhaps we need to provide hooks directly on the MediaStream to allow
> the resolution to be specified, and these hooks are invoked
> automatically when the MediaStream is connected to a <video/>.
> 

When it comes to video resolution, I still haven't seen an case where
the sink for the video still aren't able to specify a preferred resolution.

As I see it this isn't only about resolution for video. There is a bunch
of other parameters that also needs to be considered and they are
different depending on media type, audio, video, real-time text etc.

We are working on a Internet draft that goes into more details as a
response to Harald's draft-alvestrand-rtcweb-resolution-00. However, it
will take some more working days to complete and go through Ericsson's
internal process.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
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From harald@alvestrand.no  Wed May  2 06:35:24 2012
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 05/01/2012 03:57 PM, Ejzak, Richard P (Richard) wrote:
>
> Hi Justin, Partha,
>
> The call flow from Partha is a very reasonable one in the presence of 
> SIP forking.  Let's assume that a first target responded with an SDP 
> answer in a 183 that must be treated as a PRANSWER since other targets 
> have not yet had a chance to respond.  Then the first target sends an 
> UPDATE with offer2.
>
> As far as SIP is concerned, there is no such thing as a PRANSWER.  The 
> first target already responded with an SDP ANSWER and wants to UPDATE 
> with a new offer.
>
In that case, PRANSWER seems like the wrong semantic.... if you want to 
support multiple answers, shouldn't you do the cloning before you start 
accepting answers?
>
>  This is perfectly legitimate and common SIP usage.  PRANSWER is a 
> local construct that should be usable by the application to support 
> more complex offer/answer cases like forking.
>
> You must respond to this offer by cloning the peer connection, closing 
> the clone with an ANSWER identical to the previous PRANSWER and then 
> applying offer2 to the clone (or just applying the offer2 directly 
> with the understanding that the previous transaction is closed).  The 
> original peer connection must revert to the original OFFER state to 
> allow for other targets to respond.
>
> Alternately, the original peer connection can support the first target 
> and a new clone created if there is the potential for other forked 
> responses to the original offer.  Either way can be made to work.
>
> This is a perfectly reasonable SIP use case that explains why we need 
> to be able to clone the peer connection.
>

Hmm....
could you please propose a semantic for "clone" that allows us to 
evaluate whether/how it can be implemented?

Especially, I am worried about what the state of the peerconnection has 
to be before it is cloned, what the state of the peerconnection is after 
it is cloned, which peerconnection owns the various resources allocated 
(ports are the obvious part of it), and which peerconnection any streams 
(local or remote) attached to the peerconnection before the clone are 
attached to after the clone.

In an earlier thread, I suggested a peerconnection factory that could 
generate an offer, but required you to manufacture a real peerconnection 
before applying an answer (any answer).... since I'm not that interested 
in supporting forking, I haven't pursued that further. Others might want to.

I don't think any of this is impossible. I do think it requires rather 
precise definitions of what we want before we ask browser developers to 
do it.

            Harald


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    On 05/01/2012 03:57 PM, Ejzak, Richard P (Richard) wrote:
    <blockquote
cite="mid:6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com"
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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">Hi Justin, Partha,<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">The call flow from
              Partha is a very
              reasonable one in the presence of SIP forking. &nbsp;Let&#8217;s
              assume that a
              first target responded with an SDP answer in a 183 that
              must be treated as a
              PRANSWER since other targets have not yet had a chance to
              respond.&nbsp; Then
              the first target sends an UPDATE with offer2.<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;">As
              far as SIP is concerned, there is no
              such thing as a PRANSWER. &nbsp;The first target already
              responded with an SDP
              ANSWER and wants to UPDATE with a new offer.</span></font></p>
      </div>
    </blockquote>
    In that case, PRANSWER seems like the wrong semantic.... if you want
    to support multiple answers, shouldn't you do the cloning before you
    start accepting answers?<br>
    <blockquote
cite="mid:6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com"
      type="cite">
      <div class="Section1">
        <p class="MsoNormal"><font color="navy" face="Arial" size="2"><span
              style="font-size:
              10.0pt;font-family:Arial;color:navy"> &nbsp;This is perfectly
              legitimate
              and common SIP usage.&nbsp; PRANSWER is a local construct that
              should be usable
              by the application to support more complex offer/answer
              cases like forking.<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">You must respond to
              this offer by cloning
              the peer connection, closing the clone with an ANSWER
              identical to the previous
              PRANSWER and then applying offer2 to the clone (or just
              applying the offer2
              directly with the understanding that the previous
              transaction is closed). &nbsp;The
              original peer connection must revert to the original OFFER
              state to allow for
              other targets to respond.<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">Alternately, the
              original peer connection
              can support the first target and a new clone created if
              there is the potential for
              other forked responses to the original offer. &nbsp;Either way
              can be made to
              work.<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;">This
              is a perfectly reasonable SIP use
              case that explains why we need to be able to clone the
              peer connection.</span></font></p>
      </div>
    </blockquote>
    <br>
    Hmm....<br>
    could you please propose a semantic for "clone" that allows us to
    evaluate whether/how it can be implemented?<br>
    <br>
    Especially, I am worried about what the state of the peerconnection
    has to be before it is cloned, what the state of the peerconnection
    is after it is cloned, which peerconnection owns the various
    resources allocated (ports are the obvious part of it), and which
    peerconnection any streams (local or remote) attached to the
    peerconnection before the clone are attached to after the clone.<br>
    <br>
    In an earlier thread, I suggested a peerconnection factory that
    could generate an offer, but required you to manufacture a real
    peerconnection before applying an answer (any answer).... since I'm
    not that interested in supporting forking, I haven't pursued that
    further. Others might want to.<br>
    <br>
    I don't think any of this is impossible. I do think it requires
    rather precise definitions of what we want before we ask browser
    developers to do it.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
  </body>
</html>

--------------070408090808060708040105--

From harald@alvestrand.no  Wed May  2 06:40:48 2012
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Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
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On 05/01/2012 04:14 PM, Cavigioli, Chris wrote:
> Wording needs to be clarified.
> - "Eavesdropping" has a negative connotation of someone inappropriately listening in to another communication.  This should not happen.
> - "Lawful intercept" is something telecom service providers are required to provide to law enforcement with all appropriate safeguards in place http://en.wikipedia.org/wiki/Lawful_interception
> -chris
In fact the RAVEN RFC spent some time defining wiretapping:

    Wiretapping is what occurs when information passed across the
    Internet from one party to one or more other parties is delivered to
    a third party:

    1. Without the sending party knowing about the third party

    2. Without any of the recipient parties knowing about the delivery to
       the third party

    3. When the normal expectation of the sender is that the transmitted
       information will only be seen by the recipient parties or parties
       obliged to keep the information in confidence

    4. When the third party acts deliberately to target the transmission
       of the first party, either because he is of interest, or because
       the second party's reception is of interest.

    The term "party", as used here, can refer to one person, a group of
    persons, or equipment acting on behalf of persons; the term "party"
    is used for brevity.

    Of course, many wiretaps will be bidirectional, monitoring traffic
    sent by two or more parties to each other.

    Thus, for instance, monitoring public newsgroups is not wiretapping
    (condition 3 violated), random monitoring of a large population is
    not wiretapping (condition 4 violated), a recipient passing on
    private email is not wiretapping (condition 2 violated).

The call center case isn't wiretapping by RAVEN's definition, since 
conditions 1 and 2 don't hold.

Note also that chapter 4 of RAVEN is entitled "Why the IETF does not 
take a moral position".

>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of Hutton, Andrew
> Sent: Tuesday, May 01, 2012 6:14 AM
> To: Ted Hardie; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft - Eavesdropping.
>
> Hi,
>
> A number of use cases within Draft-ietf-rtcweb-use-cases-and-requirements-07 contain the statement "It is essential that the communication cannot be eavesdropped" however there is no definition of what is actually meant by "eavesdropped" although I think we all have an idea of what it means.
>
> Maybe it would be better to replace these statements with something that refers to wiretapping and RFC 2804 (RAVEN) which actually has a definition of wiretapping.
>
> Regards
> Andy
>
>
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Ted Hardie
>> Sent: 27 April 2012 17:15
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Use Case draft
>>
>> The chairs would like to ask the working group to focus on the use
>> case draft.  If you have use cases that need to be added to the
>> document or text changes you'd like to suggest, please send them in
>> for discussion before May 15th.  After this round, we will look toward
>> having a working group last call on the document (hopefully before the
>> interim meeting).
>>
>> regards,
>>
>> Ted, Magnus, Cullen
>> _______________________________________________
>> 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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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Use Case draft
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On 05/02/2012 11:05 AM, Ravindran, Parthasarathi wrote:
> Stefan,
>
> This usecase add its own requirements from identity perspective as
> I mentioned earlier.
>
> WebRTC MUST allow "Anonymous" users secure session for call center
> usecase. "Anonymous" User may be agent in the call center side or
> customer who does not require Identity to start the session.

There is currently no requirement on "identity" in the document at all. 
It was deemed beyond the scope of it when the first version was created 
(and I think this what we tried say with the initial section on that the 
"document focuses on requirements related to real-time media streams."). 
This was also inspired by that web applications usually deal with 
identity in the app itself.

Identity has since been discussed at length. I think it is up to the 
group and the chairs to decide if the document should be updated to 
cover identity, and how that (in that case) should be phrased.

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 02, 2012 2:16 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use Case draft
>>
>> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>>> One way to describe the use case is to let the contact center's media
>>> server/gateway serve as the webRTC endpoint.  Then all the issues of
>>> call delivery, call monitoring, etc. disappear.  They are handled by
>>> application software that sits behind the webRTC endpoint.  The
>>> company I work for makes a good living selling software that deals
>>> with all these issues - including bathroom breaks - and that's how we
>>> would tend to think of this case.  To us, it's a new kind of
>>> call/connection coming into the contact center, which we translate
>>> into SIP at the border and then handle normally.
>>>
>>> It's not clear to me if this use case adds any extra requirements.
>>
>> I think this is important to sort out. If the use case does not add any
>> extra requirements, what's the point of adding it?
>>
>>> We would just have to be careful not to assume that a webRTC endpoint
>>> is always a person/browser-based user agent.  It may seem a bit
>>> unsettling that the webRTC endpoint can distribute the call somewhere
>>> else and let others listen in, but as far as I can tell that is
>>> already the case.  If Bob calls Alice with full authentication and
>>> security, he can be sure that he is connected to Alice's user agent
>>> and that no one in between can listen in, but there's nothing stopping
>>> Alice from recording the audio, or forwarding it to a third party.  So
>>> Bob could in fact be talking to Mary if that's how Alice wants to
>>> arrange things (_behind_ her user agent).  In general, Bob is assured
>>> only that he is talking to someone Alice wants him to talk to, and
>>> that no one can snoop without Alice's permission.  That's very much
>>> the way things work with the call center - you are sure that you are
>>> 1) connected securely to your bank 2) talking to someone that the bank
>>> wants you to talk to 3) being recorded or snooped on only when the
>>> bank explicitly chooses to do so.
>>>
>>> - Jim
>>>
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>>
>>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>>> Andrew<andrew.hutton@siemens-enterprise.com>   wrote:
>>>> Whether anybody has been successful in the past with this type of use
>>>> case is I think irrelevant.
>>>>
>>>>
>>>>
>>>> The enterprise call centre use case is I think a vital use case
>>>> because it is a scenario in which one user is only concerned that
>>>> they can securely reach an organization/domain and is not concerned
>>>> about the individual within that domain  that they communicate with.
>>>> A suspect quite a large percentage of RTCWEB applications will be
>>>> like this and it is not covered in the current use case draft.
>>>
>>> I agree that this is a very useful use case and one I think is going
>>> to get a lot of traction. There is a very solid business case for
>>> this.  However, I have a fair amount of experience with a video call
>>> center for a client, and it is not as simple as it might seem.
>>>
>>> The essence of course is that you get the next available person, i.e.,
>>> it is anycast. Determining who the next available person is is not
>>> trivial, nor is error recovery. (If I call you, and you don't answer
>>> or the call drops or whatever,  I can leave a message or try later. If
>>> I call a help desk, and this happens, I want a new agent, ideally
>>> automatically.) Call forwarding (e.g., first tier to second tier
>>> technical support) is essential, and it may be anycast or directed.
>>> There are also some security oddities  - if I am connecting from home,
>>> I may need to authenticate, use a credit card, etc. If I am connecting
>>> from inside a store, and providing in store video technical support is
>>> big part of the market, then the store authenticates me off line and
>>> the call really should just be a button push, which implies that the
>>> store has previously authenticated some sort of master session. In
>>> addition, unlike most video calls, in the enterprise call center a
>>> supervisor may need to be able to monitor (i.e., watch) a call, and in
>>> some circumstances (financial or medical calls, for example) there
>>> will need to be third party recording. I believe that  these details
>>> would be different from the typical RTCWEB scenario.
>>>
>>> Also, there will be a temptation to do the anycasting by the
>>> techniques used to load balance servers in a data center, but I think
>>> that may not be sufficient. The call "center" may in fact be spread
>>> completely across the planet (daytime support in the US, nighttime
>>> support in India, for example) and be on multiple autonomous systems
>>> (and even from people's homes), which gives rise to some of the
>>> transport issues NVO3 may face, but without any opportunity for packet
>>> tagging. Plus, there will complicated rules about who can be selected
>>> next. RTCWEB shouldn't worry about the intricacies of bathroom break
>>> policies; these complexities should be dealt with by an
>>> enterprise-side database, which to me (together with some of the other
>>> issues above) suggests that this would probably benefit from API
>>> support.
>>>
>>> Regards Marshall
>>>
>>>
>>>>
>>>>
>>>>
>>>> So I think we need it.
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Andy
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>>> rtcweb@ietf.org
>>>>
>>>>
>>>> Subject: Re: [rtcweb] Use Case draft
>>>>
>>>>
>>>>
>>>> Without numbers it is impossible to argue, but, if we talk about the
>>>> perceived need, I disagree.  Think of the people who travel abroad
>>>> and cannot call the 800 number. (I routinely use Web interface for
>>>> calls when traveling.)
>>>>
>>>>
>>>>
>>>> I am all for  the use case, as described by Jim.
>>>>
>>>> Igor
>>>>
>>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>>
>>>> ...
>>>>
>>>> I can't tell you the actual numbers, but when presented with the
>>>> choice of calling a toll free number
>>>>
>>>> or clicking a button marked "free internet call" - almost no-one on a
>>>> real, busy site clicked the button.
>>>>
>>>> ( for every button click there were several orders of magnitude more
>>>> 0800 calls from that page).
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> So from my perspective this is a legacy interop use case with almost
>>>> zero user acceptance.
>>>>
>>>>
>>>>
>>>> (as far as I can see no-one has made this use-case desirable in
>>>> practice yet.)
>>>>
>>>> Tim.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> 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
>>> _______________________________________________ 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 Jim.Barnett@genesyslab.com  Wed May  2 07:39:39 2012
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Date: Wed, 2 May 2012 07:39:28 -0700
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com><BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl><387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com><2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com><387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com><E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com><4F9EC0B2.10903@alcatel-lucent.com><101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net><CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Stefan Hakansson LK" <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Use Case draft
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When I say that this use case may not add further requirements, I mean
that it looks like it would be possible to implement it given the
current definitions of the protocols.  However, the current use cases
are all written in terms of "the browser", which is not further defined.
But if "browser" means Mozilla, Chrome, etc., then I think it is
important to add a use case in which one of the end points is not a
browser, but an enterprise gateway (which will route the call to an
employee of its choice, and may record the call, etc.) It is important
to note that this is not a peer-to-peer use case; the gateway is not the
caller's peer.  The employee that the caller ends up talking to may be
considered a peer, but the webRTC call does not extend all the way to
that employee - it stops at the gateway. =20

This is a very different use case from any in the current document.
That's why it's important to add it, even though (as far as I can tell)
it doesn't require us to change any of the work we've done. =20

- Jim
-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Stefan Hakansson LK
Sent: Wednesday, May 02, 2012 4:46 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

On 05/01/2012 02:05 PM, Jim Barnett wrote:
> One way to describe the use case is to let the contact center's media=20
> server/gateway serve as the webRTC endpoint.  Then all the issues of=20
> call delivery, call monitoring, etc. disappear.  They are handled by=20
> application software that sits behind the webRTC endpoint.  The=20
> company I work for makes a good living selling software that deals=20
> with all these issues - including bathroom breaks - and that's how we=20
> would tend to think of this case.  To us, it's a new kind of=20
> call/connection coming into the contact center, which we translate=20
> into SIP at the border and then handle normally.
>
> It's not clear to me if this use case adds any extra requirements.

I think this is important to sort out. If the use case does not add any
extra requirements, what's the point of adding it?

> We would just have to be careful not to assume that a webRTC endpoint=20
> is always a person/browser-based user agent.  It may seem a bit=20
> unsettling that the webRTC endpoint can distribute the call somewhere=20
> else and let others listen in, but as far as I can tell that is=20
> already the case.  If Bob calls Alice with full authentication and=20
> security, he can be sure that he is connected to Alice's user agent=20
> and that no one in between can listen in, but there's nothing stopping

> Alice from recording the audio, or forwarding it to a third party.  So

> Bob could in fact be talking to Mary if that's how Alice wants to=20
> arrange things (_behind_ her user agent).  In general, Bob is assured=20
> only that he is talking to someone Alice wants him to talk to, and=20
> that no one can snoop without Alice's permission.  That's very much=20
> the way things work with the call center - you are sure that you are=20
> 1) connected securely to your bank 2) talking to someone that the bank

> wants you to talk to 3) being recorded or snooped on only when the=20
> bank explicitly chooses to do so.
>
> - Jim
>
> -----Original Message----- From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>
> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,=20
> Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
>> Whether anybody has been successful in the past with this type of use

>> case is I think irrelevant.
>>
>>
>>
>> The enterprise call centre use case is I think a vital use case=20
>> because it is a scenario in which one user is only concerned that=20
>> they can securely reach an organization/domain and is not concerned=20
>> about the individual within that domain  that they communicate with.

>> A suspect quite a large percentage of RTCWEB applications will be=20
>> like this and it is not covered in the current use case draft.
>
> I agree that this is a very useful use case and one I think is going=20
> to get a lot of traction. There is a very solid business case for=20
> this.  However, I have a fair amount of experience with a video call=20
> center for a client, and it is not as simple as it might seem.
>
> The essence of course is that you get the next available person, i.e.,

> it is anycast. Determining who the next available person is is not=20
> trivial, nor is error recovery. (If I call you, and you don't answer=20
> or the call drops or whatever,  I can leave a message or try later. If

> I call a help desk, and this happens, I want a new agent, ideally=20
> automatically.) Call forwarding (e.g., first tier to second tier=20
> technical support) is essential, and it may be anycast or directed.=20
> There are also some security oddities  - if I am connecting from home,

> I may need to authenticate, use a credit card, etc. If I am connecting

> from inside a store, and providing in store video technical support is

> big part of the market, then the store authenticates me off line and=20
> the call really should just be a button push, which implies that the=20
> store has previously authenticated some sort of master session. In=20
> addition, unlike most video calls, in the enterprise call center a=20
> supervisor may need to be able to monitor (i.e., watch) a call, and in

> some circumstances (financial or medical calls, for example) there=20
> will need to be third party recording. I believe that  these details=20
> would be different from the typical RTCWEB scenario.
>
> Also, there will be a temptation to do the anycasting by the=20
> techniques used to load balance servers in a data center, but I think=20
> that may not be sufficient. The call "center" may in fact be spread=20
> completely across the planet (daytime support in the US, nighttime=20
> support in India, for example) and be on multiple autonomous systems=20
> (and even from people's homes), which gives rise to some of the=20
> transport issues NVO3 may face, but without any opportunity for packet

> tagging. Plus, there will complicated rules about who can be selected=20
> next. RTCWEB shouldn't worry about the intricacies of bathroom break=20
> policies; these complexities should be dealt with by an=20
> enterprise-side database, which to me (together with some of the other

> issues above) suggests that this would probably benefit from API=20
> support.
>
> Regards Marshall
>
>
>>
>>
>>
>> So I think we need it.
>>
>>
>>
>> Regards
>>
>> Andy
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>> rtcweb@ietf.org
>>
>>
>> Subject: Re: [rtcweb] Use Case draft
>>
>>
>>
>> Without numbers it is impossible to argue, but, if we talk about the=20
>> perceived need, I disagree.  Think of the people who travel abroad=20
>> and cannot call the 800 number. (I routinely use Web interface for=20
>> calls when traveling.)
>>
>>
>>
>> I am all for  the use case, as described by Jim.
>>
>> Igor
>>
>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>
>> ...
>>
>> I can't tell you the actual numbers, but when presented with the=20
>> choice of calling a toll free number
>>
>> or clicking a button marked "free internet call" - almost no-one on a

>> real, busy site clicked the button.
>>
>> ( for every button click there were several orders of magnitude more=20
>> 0800 calls from that page).
>>
>>
>>
>>
>>
>> So from my perspective this is a legacy interop use case with almost=20
>> zero user acceptance.
>>
>>
>>
>> (as far as I can see no-one has made this use-case desirable in=20
>> practice yet.)
>>
>> Tim.
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>>
>> rtcweb mailing list
>>
>> rtcweb@ietf.org
>>
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>> _______________________________________________ rtcweb mailing list=20
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>
> _______________________________________________ rtcweb mailing list=20
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________ rtcweb 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

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On Wed, May 2, 2012 at 9:35 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> Especially, I am worried about what the state of the peerconnection has to
> be before it is cloned, what the state of the peerconnection is after it is
> cloned, which peerconnection owns the various resources allocated (ports
> are the obvious part of it), and which peerconnection any streams (local or
> remote) attached to the peerconnection before the clone are attached to
> after the clone.
>
>
I thought there was a discussion on this list about using the same set of
ports, ICE candidates and turn connection across multiple peerconnections.
I am actually curious if there is ever a down side of using the same set of
ICE candidates for all the streams within the all peerconnections for a
given web session. It should be possible to disambiguate those streams
based on remote IP/port and SSRC.

The state that must to be replicated with the peerconnection is the latest
offer information. It would probably be less disruptive if peerconnection
has some sort of clone method instead of using a factory. It should be
possible to clone the connection between the time offer is generated and
the final answer is received.
_____________
Roman Shpount

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<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 2, 20=
12 at 9:35 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:ha=
rald@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> w=
rote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000">Especially, I am worried about =
what the state of the peerconnection
    has to be before it is cloned, what the state of the peerconnection
    is after it is cloned, which peerconnection owns the various
    resources allocated (ports are the obvious part of it), and which
    peerconnection any streams (local or remote) attached to the
    peerconnection before the clone are attached to after the clone.<br>
    <br></div></blockquote><div><br>I thought there was a discussion on thi=
s list about using the same set of ports, ICE candidates and turn connectio=
n across multiple peerconnections. I am actually curious if there is ever a=
 down side of using the same set of ICE candidates for all the streams with=
in the all peerconnections for a given web session. It should be possible t=
o disambiguate those streams based on remote IP/port and SSRC. <br>
<br>The state that must to be replicated with the peerconnection is the lat=
est offer information. It would probably be less disruptive if peerconnecti=
on has some sort of clone method instead of using a factory. It should be p=
ossible to clone the connection between the time offer is generated and the=
 final answer is received.<br>
</div></div>_____________<br>Roman Shpount<br>
<br></div>

--0015175cd100951a0004bf0f808e--

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Subject: Re: [rtcweb] Use Case draft
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On 05/02/2012 04:39 PM, Jim Barnett wrote:
> When I say that this use case may not add further requirements, I mean
> that it looks like it would be possible to implement it given the
> current definitions of the protocols.  However, the current use cases
> are all written in terms of "the browser", which is not further defined.
> But if "browser" means Mozilla, Chrome, etc., then I think it is
> important to add a use case in which one of the end points is not a
> browser, but an enterprise gateway (which will route the call to an
> employee of its choice, and may record the call, etc.) It is important
> to note that this is not a peer-to-peer use case; the gateway is not the
> caller's peer.  The employee that the caller ends up talking to may be
> considered a peer, but the webRTC call does not extend all the way to
> that employee - it stops at the gateway.

I think all use cases in section 4.3 are browser - GW cases. That said, 
there may be good reasons for adding this one as well.

>
> This is a very different use case from any in the current document.
> That's why it's important to add it, even though (as far as I can tell)
> it doesn't require us to change any of the work we've done.
>
> - Jim
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Stefan Hakansson LK
> Sent: Wednesday, May 02, 2012 4:46 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>
> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>> One way to describe the use case is to let the contact center's media
>> server/gateway serve as the webRTC endpoint.  Then all the issues of
>> call delivery, call monitoring, etc. disappear.  They are handled by
>> application software that sits behind the webRTC endpoint.  The
>> company I work for makes a good living selling software that deals
>> with all these issues - including bathroom breaks - and that's how we
>> would tend to think of this case.  To us, it's a new kind of
>> call/connection coming into the contact center, which we translate
>> into SIP at the border and then handle normally.
>>
>> It's not clear to me if this use case adds any extra requirements.
>
> I think this is important to sort out. If the use case does not add any
> extra requirements, what's the point of adding it?
>
>> We would just have to be careful not to assume that a webRTC endpoint
>> is always a person/browser-based user agent.  It may seem a bit
>> unsettling that the webRTC endpoint can distribute the call somewhere
>> else and let others listen in, but as far as I can tell that is
>> already the case.  If Bob calls Alice with full authentication and
>> security, he can be sure that he is connected to Alice's user agent
>> and that no one in between can listen in, but there's nothing stopping
>
>> Alice from recording the audio, or forwarding it to a third party.  So
>
>> Bob could in fact be talking to Mary if that's how Alice wants to
>> arrange things (_behind_ her user agent).  In general, Bob is assured
>> only that he is talking to someone Alice wants him to talk to, and
>> that no one can snoop without Alice's permission.  That's very much
>> the way things work with the call center - you are sure that you are
>> 1) connected securely to your bank 2) talking to someone that the bank
>
>> wants you to talk to 3) being recorded or snooped on only when the
>> bank explicitly chooses to do so.
>>
>> - Jim
>>
>> -----Original Message----- From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>
>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>> Andrew<andrew.hutton@siemens-enterprise.com>   wrote:
>>> Whether anybody has been successful in the past with this type of use
>
>>> case is I think irrelevant.
>>>
>>>
>>>
>>> The enterprise call centre use case is I think a vital use case
>>> because it is a scenario in which one user is only concerned that
>>> they can securely reach an organization/domain and is not concerned
>>> about the individual within that domain  that they communicate with.
>
>>> A suspect quite a large percentage of RTCWEB applications will be
>>> like this and it is not covered in the current use case draft.
>>
>> I agree that this is a very useful use case and one I think is going
>> to get a lot of traction. There is a very solid business case for
>> this.  However, I have a fair amount of experience with a video call
>> center for a client, and it is not as simple as it might seem.
>>
>> The essence of course is that you get the next available person, i.e.,
>
>> it is anycast. Determining who the next available person is is not
>> trivial, nor is error recovery. (If I call you, and you don't answer
>> or the call drops or whatever,  I can leave a message or try later. If
>
>> I call a help desk, and this happens, I want a new agent, ideally
>> automatically.) Call forwarding (e.g., first tier to second tier
>> technical support) is essential, and it may be anycast or directed.
>> There are also some security oddities  - if I am connecting from home,
>
>> I may need to authenticate, use a credit card, etc. If I am connecting
>
>> from inside a store, and providing in store video technical support is
>
>> big part of the market, then the store authenticates me off line and
>> the call really should just be a button push, which implies that the
>> store has previously authenticated some sort of master session. In
>> addition, unlike most video calls, in the enterprise call center a
>> supervisor may need to be able to monitor (i.e., watch) a call, and in
>
>> some circumstances (financial or medical calls, for example) there
>> will need to be third party recording. I believe that  these details
>> would be different from the typical RTCWEB scenario.
>>
>> Also, there will be a temptation to do the anycasting by the
>> techniques used to load balance servers in a data center, but I think
>> that may not be sufficient. The call "center" may in fact be spread
>> completely across the planet (daytime support in the US, nighttime
>> support in India, for example) and be on multiple autonomous systems
>> (and even from people's homes), which gives rise to some of the
>> transport issues NVO3 may face, but without any opportunity for packet
>
>> tagging. Plus, there will complicated rules about who can be selected
>> next. RTCWEB shouldn't worry about the intricacies of bathroom break
>> policies; these complexities should be dealt with by an
>> enterprise-side database, which to me (together with some of the other
>
>> issues above) suggests that this would probably benefit from API
>> support.
>>
>> Regards Marshall
>>
>>
>>>
>>>
>>>
>>> So I think we need it.
>>>
>>>
>>>
>>> Regards
>>>
>>> Andy
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>> rtcweb@ietf.org
>>>
>>>
>>> Subject: Re: [rtcweb] Use Case draft
>>>
>>>
>>>
>>> Without numbers it is impossible to argue, but, if we talk about the
>>> perceived need, I disagree.  Think of the people who travel abroad
>>> and cannot call the 800 number. (I routinely use Web interface for
>>> calls when traveling.)
>>>
>>>
>>>
>>> I am all for  the use case, as described by Jim.
>>>
>>> Igor
>>>
>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>
>>> ...
>>>
>>> I can't tell you the actual numbers, but when presented with the
>>> choice of calling a toll free number
>>>
>>> or clicking a button marked "free internet call" - almost no-one on a
>
>>> real, busy site clicked the button.
>>>
>>> ( for every button click there were several orders of magnitude more
>>> 0800 calls from that page).
>>>
>>>
>>>
>>>
>>>
>>> So from my perspective this is a legacy interop use case with almost
>>> zero user acceptance.
>>>
>>>
>>>
>>> (as far as I can see no-one has made this use-case desirable in
>>> practice yet.)
>>>
>>> Tim.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> 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
>> _______________________________________________ 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 05/02/2012 05:40 PM, Roman Shpount wrote:
>
> On Wed, May 2, 2012 at 9:35 AM, Harald Alvestrand 
> <harald@alvestrand.no <mailto:harald@alvestrand.no>> wrote:
>
>     Especially, I am worried about what the state of the
>     peerconnection has to be before it is cloned, what the state of
>     the peerconnection is after it is cloned, which peerconnection
>     owns the various resources allocated (ports are the obvious part
>     of it), and which peerconnection any streams (local or remote)
>     attached to the peerconnection before the clone are attached to
>     after the clone.
>
>
> I thought there was a discussion on this list about using the same set 
> of ports, ICE candidates and turn connection across multiple 
> peerconnections. I am actually curious if there is ever a down side of 
> using the same set of ICE candidates for all the streams within the 
> all peerconnections for a given web session. It should be possible to 
> disambiguate those streams based on remote IP/port and SSRC.
It does mean that the implementation will have to do reference counting 
to know when it can close the port - if one clones the socket and binds 
the clones to different remote ports, I think the OS will take care of 
it on Unix, I'm not sure how it goes for other OSes.

Does someone know what the semantic of bind() is here - whether one 
needs to have an unbound port to fork from in order to bind to different 
remote addresses? I've not tried this myself.

In the case of hardware codecs that must be allocated to a specific 
stream .... if one supports forking, which connection (if any) gets the 
codec? The first one to start using it? What happens to the second one?

>
> The state that must to be replicated with the peerconnection is the 
> latest offer information. It would probably be less disruptive if 
> peerconnection has some sort of clone method instead of using a 
> factory. It should be possible to clone the connection between the 
> time offer is generated and the final answer is received.
I kind of think it's less disruptive to the people who don't want to 
fork stuff if you must instantiate a different object in order to 
support forking. Then implementations that don't support forking can 
simply not offer a constructor for that object.






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    On 05/02/2012 05:40 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com"
      type="cite">
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Wed, May 2, 2012 at 9:35 AM, Harald
          Alvestrand <span dir="ltr">&lt;<a moz-do-not-send="true"
              href="mailto:harald@alvestrand.no" target="_blank">harald@alvestrand.no</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            <div bgcolor="#FFFFFF" text="#000000">Especially, I am
              worried about what the state of the peerconnection has to
              be before it is cloned, what the state of the
              peerconnection is after it is cloned, which peerconnection
              owns the various resources allocated (ports are the
              obvious part of it), and which peerconnection any streams
              (local or remote) attached to the peerconnection before
              the clone are attached to after the clone.<br>
              <br>
            </div>
          </blockquote>
          <div><br>
            I thought there was a discussion on this list about using
            the same set of ports, ICE candidates and turn connection
            across multiple peerconnections. I am actually curious if
            there is ever a down side of using the same set of ICE
            candidates for all the streams within the all
            peerconnections for a given web session. It should be
            possible to disambiguate those streams based on remote
            IP/port and SSRC. <br>
          </div>
        </div>
      </div>
    </blockquote>
    It does mean that the implementation will have to do reference
    counting to know when it can close the port - if one clones the
    socket and binds the clones to different remote ports, I think the
    OS will take care of it on Unix, I'm not sure how it goes for other
    OSes.<br>
    <br>
    Does someone know what the semantic of bind() is here - whether one
    needs to have an unbound port to fork from in order to bind to
    different remote addresses? I've not tried this myself.<br>
    <br>
    In the case of hardware codecs that must be allocated to a specific
    stream .... if one supports forking, which connection (if any) gets
    the codec? The first one to start using it? What happens to the
    second one?<br>
    <br>
    <blockquote
cite="mid:CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com"
      type="cite">
      <div class="gmail_extra">
        <div class="gmail_quote">
          <div>
            <br>
            The state that must to be replicated with the peerconnection
            is the latest offer information. It would probably be less
            disruptive if peerconnection has some sort of clone method
            instead of using a factory. It should be possible to clone
            the connection between the time offer is generated and the
            final answer is received.<br>
          </div>
        </div>
      </div>
    </blockquote>
    I kind of think it's less disruptive to the people who don't want to
    fork stuff if you must instantiate a different object in order to
    support forking. Then implementations that don't support forking can
    simply not offer a constructor for that object.<br>
    <br>
    <br>
    <br>
    <br>
    <br>
  </body>
</html>

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From roman@telurix.com  Wed May  2 09:03:50 2012
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From: Roman Shpount <roman@telurix.com>
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Subject: [rtcweb] Final plea about SRTP
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I know there was a consensus call on this list that SRTP shall be used for
all the calls in WebRTC, but I still do not understand the justification
for this requirement for WebRTC applications delivered over HTTP with no
identity. For such scenarios SRTP (even DTLS-SRTP) serves almost no
purpose. If application is delivered over HTTP attacker can spoof the
entire web site. It is trivial if the attacker is on the communications
path. If attacker is seating in the airport using the same network, it can
put itself on the communications path using arp cache poisoning. Once the
web site is spoofed, any type of man in the middle attack can be
implemented. If DTLS-SRTP is used user can detect the attack by checking
the key signature, but in reality very few people will do this.

The main argument to require SRTP everywhere was that it does not break
anything. But neither would naming all the API methods in High Elfish.
Either requirement does not break things, but make working with WebRTC
harder then it should. At the same time both of those requirements are
completely unjustified.

Furthermore, assumption on this list that most of the WebRTC use would be
peer-to-peer communications between browsers with all the rest of the
communication modes, such as calling automated services or PSTN being
insignificant. I simply do not agree to this point of view. I expect that
communication with automated services, such as video greeting cards or
voice blogging, would be a significant portion of WebRTC user base. If such
automated service is deployed as a plain HTTP web site, it should be able
to communicate with web browsers using RTP. SRTP in such case would serve
no purpose.

Finally, requiring secure communications for everything is going against
the way most of the web works. Most of it is not secured and only requires
secure communications when secure (HTTPS) web site is accessed. I think it
should be the same for WebRTC, with DTLS-SRTP required when connected to
HTTPS web site and plain RTP allowed when connected to plan HTTP.
_____________
Roman Shpount

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I know there was a consensus call on this list that SRTP shall be used for =
all the calls in WebRTC, but I still do not understand the justification fo=
r this requirement for WebRTC applications delivered over HTTP with no iden=
tity. For such scenarios SRTP (even DTLS-SRTP) serves almost no purpose. If=
 application is delivered over HTTP attacker can spoof the entire web site.=
 It is trivial if the attacker is on the communications path. If attacker i=
s seating in the airport using the same network, it can put itself on the c=
ommunications path using arp cache poisoning. Once the web site is spoofed,=
 any type of man in the middle attack can be implemented. If DTLS-SRTP is u=
sed user can detect the attack by checking the key signature, but in realit=
y very few people will do this.<br>
<br>The main argument to require SRTP everywhere was that it does not break=
 anything. But neither would naming all the API methods in High Elfish. Eit=
her requirement does not break things, but make working with WebRTC harder =
then it should. At the same time both of those requirements are completely =
unjustified.<br>
<br>Furthermore, assumption on this list that most of the WebRTC use would =
be peer-to-peer communications between browsers with all the rest of the co=
mmunication modes, such as calling automated services or PSTN being insigni=
ficant. I simply do not agree to this point of view. I expect that communic=
ation with automated services, such as video greeting cards or voice bloggi=
ng, would be a significant portion of WebRTC user base. If such automated s=
ervice is deployed as a plain HTTP web site, it should be able to communica=
te with web browsers using RTP. SRTP in such case would serve no purpose.<b=
r>
<br>Finally, requiring secure communications for everything is going agains=
t the way most of the web works. Most of it is not secured and only require=
s secure communications when secure (HTTPS) web site is accessed. I think i=
t should be the same for WebRTC, with DTLS-SRTP required when connected to =
HTTPS web site and plain RTP allowed when connected to plan HTTP.<br clear=
=3D"all">
_____________<br>Roman Shpount<br>

--0015175cd0da06691004bf0fd55e--

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Subject: Re: [rtcweb] Use Case draft
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I am afraid I don't understand this.

For the call center case, when I "call" company X, I would like to make 
sure that I am connected to an agent representing X, and so the identity 
should looke like something "X_Agent_707," not "Anonymous."

Igor

On 5/2/2012 5:05 AM, Ravindran, Parthasarathi wrote:
> Stefan,
>
> This usecase add its own requirements from identity perspective as
> I mentioned earlier.
>
> WebRTC MUST allow "Anonymous" users secure session for call center
> usecase. "Anonymous" User may be agent in the call center side or
> customer who does not require Identity to start the session.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Stefan Hakansson LK
>> Sent: Wednesday, May 02, 2012 2:16 PM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use Case draft
>>
>> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>>> One way to describe the use case is to let the contact center's media
>>> server/gateway serve as the webRTC endpoint.  Then all the issues of
>>> call delivery, call monitoring, etc. disappear.  They are handled by
>>> application software that sits behind the webRTC endpoint.  The
>>> company I work for makes a good living selling software that deals
>>> with all these issues - including bathroom breaks - and that's how we
>>> would tend to think of this case.  To us, it's a new kind of
>>> call/connection coming into the contact center, which we translate
>>> into SIP at the border and then handle normally.
>>>
>>> It's not clear to me if this use case adds any extra requirements.
>> I think this is important to sort out. If the use case does not add any
>> extra requirements, what's the point of adding it?
>>
>>> We would just have to be careful not to assume that a webRTC endpoint
>>> is always a person/browser-based user agent.  It may seem a bit
>>> unsettling that the webRTC endpoint can distribute the call somewhere
>>> else and let others listen in, but as far as I can tell that is
>>> already the case.  If Bob calls Alice with full authentication and
>>> security, he can be sure that he is connected to Alice's user agent
>>> and that no one in between can listen in, but there's nothing stopping
>>> Alice from recording the audio, or forwarding it to a third party.  So
>>> Bob could in fact be talking to Mary if that's how Alice wants to
>>> arrange things (_behind_ her user agent).  In general, Bob is assured
>>> only that he is talking to someone Alice wants him to talk to, and
>>> that no one can snoop without Alice's permission.  That's very much
>>> the way things work with the call center - you are sure that you are
>>> 1) connected securely to your bank 2) talking to someone that the bank
>>> wants you to talk to 3) being recorded or snooped on only when the
>>> bank explicitly chooses to do so.
>>>
>>> - Jim
>>>
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>>
>>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>>> Andrew<andrew.hutton@siemens-enterprise.com>   wrote:
>>>> Whether anybody has been successful in the past with this type of use
>>>> case is I think irrelevant.
>>>>
>>>>
>>>>
>>>> The enterprise call centre use case is I think a vital use case
>>>> because it is a scenario in which one user is only concerned that
>>>> they can securely reach an organization/domain and is not concerned
>>>> about the individual within that domain  that they communicate with.
>>>> A suspect quite a large percentage of RTCWEB applications will be
>>>> like this and it is not covered in the current use case draft.
>>> I agree that this is a very useful use case and one I think is going
>>> to get a lot of traction. There is a very solid business case for
>>> this.  However, I have a fair amount of experience with a video call
>>> center for a client, and it is not as simple as it might seem.
>>>
>>> The essence of course is that you get the next available person, i.e.,
>>> it is anycast. Determining who the next available person is is not
>>> trivial, nor is error recovery. (If I call you, and you don't answer
>>> or the call drops or whatever,  I can leave a message or try later. If
>>> I call a help desk, and this happens, I want a new agent, ideally
>>> automatically.) Call forwarding (e.g., first tier to second tier
>>> technical support) is essential, and it may be anycast or directed.
>>> There are also some security oddities  - if I am connecting from home,
>>> I may need to authenticate, use a credit card, etc. If I am connecting
>>> from inside a store, and providing in store video technical support is
>>> big part of the market, then the store authenticates me off line and
>>> the call really should just be a button push, which implies that the
>>> store has previously authenticated some sort of master session. In
>>> addition, unlike most video calls, in the enterprise call center a
>>> supervisor may need to be able to monitor (i.e., watch) a call, and in
>>> some circumstances (financial or medical calls, for example) there
>>> will need to be third party recording. I believe that  these details
>>> would be different from the typical RTCWEB scenario.
>>>
>>> Also, there will be a temptation to do the anycasting by the
>>> techniques used to load balance servers in a data center, but I think
>>> that may not be sufficient. The call "center" may in fact be spread
>>> completely across the planet (daytime support in the US, nighttime
>>> support in India, for example) and be on multiple autonomous systems
>>> (and even from people's homes), which gives rise to some of the
>>> transport issues NVO3 may face, but without any opportunity for packet
>>> tagging. Plus, there will complicated rules about who can be selected
>>> next. RTCWEB shouldn't worry about the intricacies of bathroom break
>>> policies; these complexities should be dealt with by an
>>> enterprise-side database, which to me (together with some of the other
>>> issues above) suggests that this would probably benefit from API
>>> support.
>>>
>>> Regards Marshall
>>>
>>>
>>>>
>>>>
>>>> So I think we need it.
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Andy
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>>> rtcweb@ietf.org
>>>>
>>>>
>>>> Subject: Re: [rtcweb] Use Case draft
>>>>
>>>>
>>>>
>>>> Without numbers it is impossible to argue, but, if we talk about the
>>>> perceived need, I disagree.  Think of the people who travel abroad
>>>> and cannot call the 800 number. (I routinely use Web interface for
>>>> calls when traveling.)
>>>>
>>>>
>>>>
>>>> I am all for  the use case, as described by Jim.
>>>>
>>>> Igor
>>>>
>>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>>
>>>> ...
>>>>
>>>> I can't tell you the actual numbers, but when presented with the
>>>> choice of calling a toll free number
>>>>
>>>> or clicking a button marked "free internet call" - almost no-one on a
>>>> real, busy site clicked the button.
>>>>
>>>> ( for every button click there were several orders of magnitude more
>>>> 0800 calls from that page).
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> So from my perspective this is a legacy interop use case with almost
>>>> zero user acceptance.
>>>>
>>>>
>>>>
>>>> (as far as I can see no-one has made this use-case desirable in
>>>> practice yet.)
>>>>
>>>> Tim.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> 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
>>> _______________________________________________ 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
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From christer.holmberg@ericsson.com  Wed May  2 09:17:49 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, Roman Shpount <roman@telurix.com>
Date: Wed, 2 May 2012 18:13:17 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

I thought that we had more or less agreed on the fact that, eventhough we w=
ould support forking, we would only support one "active" remote location - =
and typically that would be the location from where you got the previous an=
swer.

So, if you receive an answer from R1, that becomes "active". Then, if you r=
eceive an answer from R2, that becomes "active".

Then, if you receive an UPDATE from R1, you would reject it, since R2 is "a=
ctive".

...or, alternatively you could switch back to R1, and accept the UPDATE. Th=
at is implementation dependent, and nothing we need to specify.

Do people think that is not enough, and that we would need to support multi=
ple remote locations simultanously (no matter whether it's implemented usin=
g cloning or something else)?

Regards,

Christer



________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Harald=
 Alvestrand [harald@alvestrand.no]
Sent: Wednesday, May 02, 2012 6:54 PM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

On 05/02/2012 05:40 PM, Roman Shpount wrote:

On Wed, May 2, 2012 at 9:35 AM, Harald Alvestrand <harald@alvestrand.no<mai=
lto:harald@alvestrand.no>> wrote:
Especially, I am worried about what the state of the peerconnection has to =
be before it is cloned, what the state of the peerconnection is after it is=
 cloned, which peerconnection owns the various resources allocated (ports a=
re the obvious part of it), and which peerconnection any streams (local or =
remote) attached to the peerconnection before the clone are attached to aft=
er the clone.


I thought there was a discussion on this list about using the same set of p=
orts, ICE candidates and turn connection across multiple peerconnections. I=
 am actually curious if there is ever a down side of using the same set of =
ICE candidates for all the streams within the all peerconnections for a giv=
en web session. It should be possible to disambiguate those streams based o=
n remote IP/port and SSRC.
It does mean that the implementation will have to do reference counting to =
know when it can close the port - if one clones the socket and binds the cl=
ones to different remote ports, I think the OS will take care of it on Unix=
, I'm not sure how it goes for other OSes.

Does someone know what the semantic of bind() is here - whether one needs t=
o have an unbound port to fork from in order to bind to different remote ad=
dresses? I've not tried this myself.

In the case of hardware codecs that must be allocated to a specific stream =
.... if one supports forking, which connection (if any) gets the codec? The=
 first one to start using it? What happens to the second one?


The state that must to be replicated with the peerconnection is the latest =
offer information. It would probably be less disruptive if peerconnection h=
as some sort of clone method instead of using a factory. It should be possi=
ble to clone the connection between the time offer is generated and the fin=
al answer is received.
I kind of think it's less disruptive to the people who don't want to fork s=
tuff if you must instantiate a different object in order to support forking=
. Then implementations that don't support forking can simply not offer a co=
nstructor for that object.






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--0015175cdc56ab6ae104bf101b98
Content-Type: text/plain; charset=ISO-8859-1

On Wed, May 2, 2012 at 11:54 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  It does mean that the implementation will have to do reference counting
> to know when it can close the port - if one clones the socket and binds the
> clones to different remote ports, I think the OS will take care of it on
> Unix, I'm not sure how it goes for other OSes.
>
> Does someone know what the semantic of bind() is here - whether one needs
> to have an unbound port to fork from in order to bind to different remote
> addresses? I've not tried this myself.
>

I actually do not think this cloning the socket will work properly even on
Unix. You can connect a cloned UDP socket to a different remote IP/port but
both sockets will continue to receive packets from ALL the remote IP/ports.

Even if it does work, this does not prevent the implementation from
reference counting, since the same TURN allocation would need to be reused
across multiple peerconnections, with a new set of permissions/channels
created for each connection.

In the case of hardware codecs that must be allocated to a specific stream
> .... if one supports forking, which connection (if any) gets the codec? The
> first one to start using it? What happens to the second one?
>

My understanding was that you do not really need to allocated hardware
resources until the answer arrives and ICE handshake completes. So, my
assumption was that in case of forking, original connection does not
allocate any codec related resources. If the connection is cloned, then it
does not allocation any codec related resources either and stores the
reference for the sockets and TURN allocations from the original
connection. When either connection receives an answer, it conducts the ICE
handshake and allocates resources as needed.


>  I kind of think it's less disruptive to the people who don't want to fork
> stuff if you must instantiate a different object in order to support
> forking. Then implementations that don't support forking can simply not
> offer a constructor for that object.
>

Alternatively clone can return and error/null if cloning is not supported.
_____________
Roman Shpount

--0015175cdc56ab6ae104bf101b98
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Wed, May 2, 2012 a=
t 11:54 AM, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:haral=
d@alvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrot=
e:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex">
<div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"im">
 =20
   =20
 =20
 =20
          </div>
    It does mean that the implementation will have to do reference
    counting to know when it can close the port - if one clones the
    socket and binds the clones to different remote ports, I think the
    OS will take care of it on Unix, I&#39;m not sure how it goes for other
    OSes.<br>
    <br>
    Does someone know what the semantic of bind() is here - whether one
    needs to have an unbound port to fork from in order to bind to
    different remote addresses? I&#39;ve not tried this myself.<br></div></=
blockquote><div>=A0<br>I actually do not think this cloning the socket will=
 work properly even on Unix. You can connect a cloned UDP socket to a diffe=
rent remote IP/port but both sockets will continue to receive packets from =
ALL the remote IP/ports.<br>
<br>Even if it does work, this does not prevent the implementation from ref=
erence counting, since the same TURN allocation would need to be reused acr=
oss multiple peerconnections, with a new set of permissions/channels create=
d for each connection.<br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div bgcolor=3D=
"#FFFFFF" text=3D"#000000">
    In the case of hardware codecs that must be allocated to a specific
    stream .... if one supports forking, which connection (if any) gets
    the codec? The first one to start using it? What happens to the
    second one?<div class=3D"im"></div></div></blockquote><div><br>My under=
standing was that you do not really need to allocated hardware resources un=
til the answer arrives and ICE handshake completes. So, my assumption was t=
hat in case of forking, original connection does not allocate any codec rel=
ated resources. If the connection is cloned, then it does not allocation an=
y codec related resources either and stores the reference for the sockets a=
nd TURN allocations from the original connection. When either connection re=
ceives an answer, it conducts the ICE handshake and allocates resources as =
needed.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div bgcolor=3D"=
#FFFFFF" text=3D"#000000"><div class=3D"im">
   =20
    </div>
    I kind of think it&#39;s less disruptive to the people who don&#39;t wa=
nt to
    fork stuff if you must instantiate a different object in order to
    support forking. Then implementations that don&#39;t support forking ca=
n
    simply not offer a constructor for that object.<br></div></blockquote><=
div><br>Alternatively clone can return and error/null if cloning is not sup=
ported. <br></div></div>_____________<br>Roman Shpount<br>
<br></div>

--0015175cdc56ab6ae104bf101b98--

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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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On Wed, May 2, 2012 at 12:13 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Do people think that is not enough, and that we would need to support
> multiple remote locations simultanously (no matter whether it's implemented
> using cloning or something else)?
>
>
I never thought that was enough if we are to interop with anything that
support forking. Latest example from Partha shows that this is indeed the
case.

I actually think that adding support for forking is trivial (order of
magnitude less complcated then DTLS-SRTP or identity) and would simplify
interop as well as enable numerous additional calling scenarios.
_____________
Roman Shpount

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<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 2, 20=
12 at 12:13 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:c=
hrister.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@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">
Do people think that is not enough, and that we would need to support multi=
ple remote locations simultanously (no matter whether it&#39;s implemented =
using cloning or something else)?<br>
<br></blockquote><div><br>I never thought that was enough if we are to inte=
rop with anything that support forking. Latest example from Partha shows th=
at this is indeed the case.<br><br>I actually think that adding support for=
 forking is trivial (order of magnitude less complcated then DTLS-SRTP or i=
dentity) and would simplify interop as well as enable numerous additional c=
alling scenarios.<br>
</div></div>_____________<br>Roman Shpount<br>
<br></div>

--0015175cfd18593ca904bf10296a--

From christer.holmberg@ericsson.com  Wed May  2 09:31:47 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Date: Wed, 2 May 2012 18:28:32 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>> Do people think that is not enough, and that we would need to support mu=
ltiple remote locations=20
>> simultanously (no matter whether it's implemented using cloning or somet=
hing else)?
>
> I never thought that was enough if we are to interop with anything that s=
upport forking. Latest example from > Partha shows that this is indeed the =
case.
>
> I actually think that adding support for forking is trivial (order of mag=
nitude less complcated then DTLS-
> SRTP or identity) and would simplify interop as well as enable numerous a=
dditional calling scenarios.

It's not about forking as such - I also agree we need to support that.

The question is whether we need to *simultanously* need to support multiple=
 remote locations, or whether we only have one "active" remote location, bu=
t we allow switching from one to another (based on where the previous answe=
r can from, or whatever policy).

Regards,

Christer=

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This is a multi-part message in MIME format.
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On 05/02/2012 06:27 PM, Roman Shpount wrote:
>
> On Wed, May 2, 2012 at 12:13 PM, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Do people think that is not enough, and that we would need to
>     support multiple remote locations simultanously (no matter whether
>     it's implemented using cloning or something else)?
>
>
> I never thought that was enough if we are to interop with anything 
> that support forking. Latest example from Partha shows that this is 
> indeed the case.
>
> I actually think that adding support for forking is trivial (order of 
> magnitude less complcated then DTLS-SRTP or identity) and would 
> simplify interop as well as enable numerous additional calling scenarios.

I'm looking forward to reviewing your detailed proposal and your 
implementation.


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    On 05/02/2012 06:27 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com"
      type="cite">
      <div class="gmail_extra"><br>
        <div class="gmail_quote">On Wed, May 2, 2012 at 12:13 PM,
          Christer Holmberg <span dir="ltr">&lt;<a
              moz-do-not-send="true"
              href="mailto:christer.holmberg@ericsson.com"
              target="_blank">christer.holmberg@ericsson.com</a>&gt;</span>
          wrote:<br>
          <blockquote class="gmail_quote" style="margin:0 0 0
            .8ex;border-left:1px #ccc solid;padding-left:1ex">
            Do people think that is not enough, and that we would need
            to support multiple remote locations simultanously (no
            matter whether it's implemented using cloning or something
            else)?<br>
            <br>
          </blockquote>
          <div><br>
            I never thought that was enough if we are to interop with
            anything that support forking. Latest example from Partha
            shows that this is indeed the case.<br>
            <br>
            I actually think that adding support for forking is trivial
            (order of magnitude less complcated then DTLS-SRTP or
            identity) and would simplify interop as well as enable
            numerous additional calling scenarios.<br>
          </div>
        </div>
      </div>
    </blockquote>
    <br>
    I'm looking forward to reviewing your detailed proposal and your
    implementation.<br>
    <br>
  </body>
</html>

--------------090903080506070707010605--

From roman@telurix.com  Wed May  2 09:51:01 2012
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Subject: Re: [rtcweb] Use Case draft -- Enterprise Policies
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Couple more use cases that I've mentioned on this least before had to do
with enforcing enterprise policies on  Web RTC clients at the enterprise
location. Keep in mind that I am not looking for a way to disable WebRTC in
the location completely, but to restrict its use of functionality. I see
the following scenarios:

1. Enterprise would like to limit the amount of bandwidth available for
WebRTC communications per location and per user.

This can be addressed by setting up enterprise network not to route to a
public internet, setting up an HTTP proxy and TURN server on the enterprise
network, and allowing to overwrite TURN server settings for enterprise
users. This way no unauthorized WebRTC communications would be allowed and
bandwidth restrictions can be enforced by the TURN server. TURN user
credentials can be used to restrict bandwidth per user or to assign user
bandwidth quotas. This provides an additional requirements that TURN server
configuration provided in the browser overwrites TURN/STUN configuration
provided by WebRTC API.

2. Enterprise would like to limit WebRTC communications (but not the other
communications such as HTTP)  to particular networks.

This can be addressed using the premise based TURN server, as in 1.

3. Enterprise would like to limit certain types of communications, ie
enable audio, but disable video and data for all the WebRTC applications on
its premises.

This currently has no solution, since there is no consistent way to know
which remote IP/port/SSRC is used for what type of communication. To enable
this, some sort of centralized SDP monitoring/modification service needs to
be deployed. Proposed solution of using HTTP proxy to modify WebRTC
application does not seem to be very stable and can be easily circumvented.

4. Enterprise would like to limit external communications only to
destinations signed by a specified list of identity providers, ie users are
allowed to communicate with anybody with identity at acme.com, but not with
user with identity at socialnetwork.com

This currently has no solution, since there is no consistent way to know
which remote IP/port is used by what remote identity. To enable this, some
sort of centralized SDP monitoring/modification service needs to be
deployed as in 3.

5. Enterprise would like to enable only communications that are recorded to
leave its premises.

This currently has no solution, since there is no consistent way to know if
communication with particular remote IP/port is recorded. This does not
fall under RAVEN since it is consensual by the user. Using solutions such
as SIP recording or installing some sort of desktop based recording
solutions does not solve this scenario since it provides no way to enforce
that only recorded communications are allowed to leave the premises. Also
note that on premise communications do not have to be recorded. Once again,
some sort for SDP monitoring/modification service needs to be implemented
to support this.
_____________
Roman Shpount

--0015175d67b49a44b904bf107d46
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Couple more use cases that I&#39;ve mentioned on this least before had to d=
o with enforcing enterprise policies on=A0 Web RTC clients at the enterpris=
e location. Keep in mind that I am not looking for a way to disable WebRTC =
in the location completely, but to restrict its use of functionality. I see=
 the following scenarios:<br>
<br>1. Enterprise would like to limit the amount of bandwidth available for=
 WebRTC communications per location and per user.<br><br>This can be addres=
sed by setting up enterprise network not to route to a public internet, set=
ting up an HTTP proxy and TURN server on the enterprise network, and allowi=
ng to overwrite TURN server settings for enterprise users. This way no unau=
thorized WebRTC communications would be allowed and bandwidth restrictions =
can be enforced by the TURN server. TURN user credentials can be used to re=
strict bandwidth per user or to assign user bandwidth quotas. This provides=
 an additional requirements that TURN server configuration provided in the =
browser overwrites TURN/STUN configuration provided by WebRTC API.<br>
<br>2. Enterprise would like to limit WebRTC communications (but not the ot=
her communications such as HTTP)=A0 to particular networks.<br><br>This can=
 be addressed using the premise based TURN server, as in 1.<br><br>3. Enter=
prise would like to limit certain types of communications, ie enable audio,=
 but disable video and data for all the WebRTC applications on its premises=
.<br>
<br>This currently has no solution, since there is no consistent way to kno=
w which remote IP/port/SSRC is used for what type of communication. To enab=
le this, some sort of centralized SDP monitoring/modification service needs=
 to be deployed. Proposed solution of using HTTP proxy to modify WebRTC app=
lication does not seem to be very stable and can be easily circumvented.<br=
>
<br>4. Enterprise would like to limit external communications only to desti=
nations signed by a specified list of identity providers, ie users are allo=
wed to communicate with anybody with identity at <a href=3D"http://acme.com=
">acme.com</a>, but not with user with identity at <a href=3D"http://social=
network.com">socialnetwork.com</a><br>

<br>
This currently has no solution, since there is no consistent way to know
 which remote IP/port is used by what remote identity. To=20
enable this, some sort of centralized SDP monitoring/modification=20
service needs to be deployed as in 3.<br>
<br>5. Enterprise would like to enable only communications that are recorde=
d to leave its premises.<br><br>This currently has no solution, since there=
 is no consistent way to know if communication with particular remote IP/po=
rt is recorded. This does not fall under RAVEN since it is consensual by th=
e user. Using solutions such as SIP recording or installing some sort of de=
sktop based recording solutions does not solve this scenario since it provi=
des no way to enforce that only recorded communications are allowed to leav=
e the premises. Also note that on premise communications do not have to be =
recorded. Once again, some sort for  SDP monitoring/modification=20
service needs to be implemented to support this.<br clear=3D"all">_________=
____<br>Roman Shpount<br>
<div class=3D"gmail_extra"><br><br></div>

--0015175d67b49a44b904bf107d46--

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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Use Case draft -- Enterprise Policies
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On Wed, May 2, 2012 at 9:50 AM, Roman Shpount <roman@telurix.com> wrote:
> Couple more use cases that I've mentioned on this least before had to do
> with enforcing enterprise policies on=A0 Web RTC clients at the enterpris=
e
> location. Keep in mind that I am not looking for a way to disable WebRTC =
in
> the location completely, but to restrict its use of functionality. I see =
the
> following scenarios:
>
> 1. Enterprise would like to limit the amount of bandwidth available for
> WebRTC communications per location and per user.
>

It's a bit hard to tease this apart from general per-user fairness methods
and methods that would generally provide (or limit) capabilities for types =
of
flows.  Why would an enterprise apply different limits because they were
signalled with WebRTC?

regards,

Ted

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--0015175cdc56b8d9ec04bf112213
Content-Type: text/plain; charset=ISO-8859-1

On Wed, May 2, 2012 at 12:57 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Wed, May 2, 2012 at 9:50 AM, Roman Shpount <roman@telurix.com> wrote:
> > 1. Enterprise would like to limit the amount of bandwidth available for
> > WebRTC communications per location and per user.
> >
>
> It's a bit hard to tease this apart from general per-user fairness methods
> and methods that would generally provide (or limit) capabilities for types
> of
> flows.  Why would an enterprise apply different limits because they were
> signalled with WebRTC?
>
>
I was trying to follow enterprise policies that are used in conjunction
with IP-PBX deployments in enterprises. Typically you would see the
location level limit. Once this location limit is reached, new real time
communications are not allowed, unless they are from high priority caller.
In this case, one of the flows from the low priority caller is disconnected
and high priority call is connected.

With WebRTC and variable rate codecs you can actually implement much better
bandwidth control schemes, but having an enterprise controlled TURN server
that manages bandwidth and QOS parameters and can attribute which
enterprise user this bandwidth is being used by can be very useful.
_____________
Roman Shpount

--0015175cdc56b8d9ec04bf112213
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<br><div class=3D"gmail_extra"><div class=3D"gmail_quote">On Wed, May 2, 20=
12 at 12:57 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf=
@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">
<div class=3D"im">On Wed, May 2, 2012 at 9:50 AM, Roman Shpount &lt;<a href=
=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt; 1. Enterprise would like to limit the amount of bandwidth available fo=
r<br>
&gt; WebRTC communications per location and per user.<br>
&gt;<br>
<br>
</div>It&#39;s a bit hard to tease this apart from general per-user fairnes=
s methods<br>
and methods that would generally provide (or limit) capabilities for types =
of<br>
flows. =A0Why would an enterprise apply different limits because they were<=
br>
signalled with WebRTC?<br>
<br></blockquote><div>=A0</div><div>I was trying to follow enterprise polic=
ies that are used in conjunction with IP-PBX deployments in enterprises. Ty=
pically you would see the location level limit. Once this location limit is=
 reached, new real time communications are not allowed, unless they are fro=
m high priority caller. In this case, one of the flows from the low priorit=
y caller is disconnected and high priority call is connected. <br>
<br>With WebRTC and variable rate codecs you can actually implement much be=
tter bandwidth control schemes, but having an enterprise controlled TURN se=
rver that manages bandwidth and QOS parameters and can attribute which ente=
rprise user this bandwidth is being used by can be very useful.<br>
</div></div>_____________<br>Roman Shpount<br>
<br></div>

--0015175cdc56b8d9ec04bf112213--

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To: "'Jim Barnett'" <Jim.Barnett@genesyslab.com>, "'Stefan Hakansson LK'" <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com><BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl><387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com><2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com><387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com><E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com><4F9EC0B2.10903@alcatel-lucent.com><101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net><CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com>	<4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com>
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Subject: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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> -----Original Message-----
> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> Behalf Of Jim Barnett
> Sent: Wednesday, May 02, 2012 7:39 AM
> To: Stefan Hakansson LK; WEBRTC@ietf.org
> Subject: Re: [WEBRTC] Use Case draft
> 
> When I say that this use case may not add further requirements, I mean
> that it looks like it would be possible to implement it given the
> current definitions of the protocols.  However, the current use cases
> are all written in terms of "the browser", which is not further
> defined.
> But if "browser" means Mozilla, Chrome, etc., then I think it is
> important to add a use case in which one of the end points is not a
> browser, but an enterprise gateway (which will route the call to an
> employee of its choice, and may record the call, etc.) It is important
> to note that this is not a peer-to-peer use case; the gateway is not
> the
> caller's peer.  The employee that the caller ends up talking to may be
> considered a peer, but the webRTC call does not extend all the way to
> that employee - it stops at the gateway.
> 
> This is a very different use case from any in the current document.
> That's why it's important to add it, even though (as far as I can tell)
> it doesn't require us to change any of the work we've done.

Somewhere, we need consensus on a model for interworking WEBRTC 
endpoints with non-WEBRTC endpoints.  

The decision comes down to this:  

  1. encumber WEBRTC endpoints with the interworking 
     effort, or 
  2. encumber a separate interworking device with the 
     interworking effort.

I believe we have a better chance of success with (2), where 
possible to do (2).

For some decisions, such as Consent Freshness (previously called Voice 
Hammer Attack in http://tools.ietf.org/html/rfc5245#section-18.5.1),
non-WEBRTC endpoints need to respond to those ICE connectivity
checks or have a gateway in front of them that responds to those
connectivity checks on their behalf.  This means that WEBRTC
cannot work directly with some existing SIP equipment (because
a lot of SIP equipment does not support ICE).

For other decisions, such as if we disallow un-encrypted RTP by
WEBRTC endpoints, we create a requirement that some device does
the interworking between WEBRTC endpoints (which do only SRTP) 
and non-WEBRTC endpoints (which do RTP).  That means, for that
interworking, we would adopt the interworking model on slide 7 
that I presented at IETF83,
http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf

However, when I presented slide 7, there were objections at the 
microphone that this model 'is broken'.  I would like to understand 
the objections so we can reach consensus on how interworking from
WEBRTC to non-WEBRTC is expected to occur.

-d


> - Jim
> -----Original Message-----
> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> Behalf
> Of Stefan Hakansson LK
> Sent: Wednesday, May 02, 2012 4:46 AM
> To: WEBRTC@ietf.org
> Subject: Re: [WEBRTC] Use Case draft
> 
> On 05/01/2012 02:05 PM, Jim Barnett wrote:
> > One way to describe the use case is to let the contact center's media
> > server/gateway serve as the webRTC endpoint.  Then all the issues of
> > call delivery, call monitoring, etc. disappear.  They are handled by
> > application software that sits behind the webRTC endpoint.  The
> > company I work for makes a good living selling software that deals
> > with all these issues - including bathroom breaks - and that's how we
> > would tend to think of this case.  To us, it's a new kind of
> > call/connection coming into the contact center, which we translate
> > into SIP at the border and then handle normally.
> >
> > It's not clear to me if this use case adds any extra requirements.
> 
> I think this is important to sort out. If the use case does not add any
> extra requirements, what's the point of adding it?
> 
> > We would just have to be careful not to assume that a webRTC endpoint
> > is always a person/browser-based user agent.  It may seem a bit
> > unsettling that the webRTC endpoint can distribute the call somewhere
> > else and let others listen in, but as far as I can tell that is
> > already the case.  If Bob calls Alice with full authentication and
> > security, he can be sure that he is connected to Alice's user agent
> > and that no one in between can listen in, but there's nothing
> stopping
> 
> > Alice from recording the audio, or forwarding it to a third party.
> So
> 
> > Bob could in fact be talking to Mary if that's how Alice wants to
> > arrange things (_behind_ her user agent).  In general, Bob is assured
> > only that he is talking to someone Alice wants him to talk to, and
> > that no one can snoop without Alice's permission.  That's very much
> > the way things work with the call center - you are sure that you are
> > 1) connected securely to your bank 2) talking to someone that the
> bank
> 
> > wants you to talk to 3) being recorded or snooped on only when the
> > bank explicitly chooses to do so.
> >
> > - Jim
> >
> > -----Original Message----- From: WEBRTC-bounces@ietf.org
> > [mailto:WEBRTC-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
> > Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
> > WEBRTC@ietf.org Subject: Re: [WEBRTC] Use Case draft
> >
> > On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
> > Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
> >> Whether anybody has been successful in the past with this type of
> use
> 
> >> case is I think irrelevant.
> >>
> >>
> >>
> >> The enterprise call centre use case is I think a vital use case
> >> because it is a scenario in which one user is only concerned that
> >> they can securely reach an organization/domain and is not concerned
> >> about the individual within that domain  that they communicate with.
> 
> >> A suspect quite a large percentage of WEBRTC applications will be
> >> like this and it is not covered in the current use case draft.
> >
> > I agree that this is a very useful use case and one I think is going
> > to get a lot of traction. There is a very solid business case for
> > this.  However, I have a fair amount of experience with a video call
> > center for a client, and it is not as simple as it might seem.
> >
> > The essence of course is that you get the next available person,
> i.e.,
> 
> > it is anycast. Determining who the next available person is is not
> > trivial, nor is error recovery. (If I call you, and you don't answer
> > or the call drops or whatever,  I can leave a message or try later.
> If
> 
> > I call a help desk, and this happens, I want a new agent, ideally
> > automatically.) Call forwarding (e.g., first tier to second tier
> > technical support) is essential, and it may be anycast or directed.
> > There are also some security oddities  - if I am connecting from
> home,
> 
> > I may need to authenticate, use a credit card, etc. If I am
> connecting
> 
> > from inside a store, and providing in store video technical support
> is
> 
> > big part of the market, then the store authenticates me off line and
> > the call really should just be a button push, which implies that the
> > store has previously authenticated some sort of master session. In
> > addition, unlike most video calls, in the enterprise call center a
> > supervisor may need to be able to monitor (i.e., watch) a call, and
> in
> 
> > some circumstances (financial or medical calls, for example) there
> > will need to be third party recording. I believe that  these details
> > would be different from the typical WEBRTC scenario.
> >
> > Also, there will be a temptation to do the anycasting by the
> > techniques used to load balance servers in a data center, but I think
> > that may not be sufficient. The call "center" may in fact be spread
> > completely across the planet (daytime support in the US, nighttime
> > support in India, for example) and be on multiple autonomous systems
> > (and even from people's homes), which gives rise to some of the
> > transport issues NVO3 may face, but without any opportunity for
> packet
> 
> > tagging. Plus, there will complicated rules about who can be selected
> > next. WEBRTC shouldn't worry about the intricacies of bathroom break
> > policies; these complexities should be dealt with by an
> > enterprise-side database, which to me (together with some of the
> other
> 
> > issues above) suggests that this would probably benefit from API
> > support.
> >
> > Regards Marshall
> >
> >
> >>
> >>
> >>
> >> So I think we need it.
> >>
> >>
> >>
> >> Regards
> >>
> >> Andy
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> >> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
> >> WEBRTC@ietf.org
> >>
> >>
> >> Subject: Re: [WEBRTC] Use Case draft
> >>
> >>
> >>
> >> Without numbers it is impossible to argue, but, if we talk about the
> >> perceived need, I disagree.  Think of the people who travel abroad
> >> and cannot call the 800 number. (I routinely use Web interface for
> >> calls when traveling.)
> >>
> >>
> >>
> >> I am all for  the use case, as described by Jim.
> >>
> >> Igor
> >>
> >> On 4/30/2012 9:54 AM, Tim Panton wrote:
> >>
> >> ...
> >>
> >> I can't tell you the actual numbers, but when presented with the
> >> choice of calling a toll free number
> >>
> >> or clicking a button marked "free internet call" - almost no-one on
> a
> 
> >> real, busy site clicked the button.
> >>
> >> ( for every button click there were several orders of magnitude more
> >> 0800 calls from that page).
> >>
> >>
> >>
> >>
> >>
> >> So from my perspective this is a legacy interop use case with almost
> >> zero user acceptance.
> >>
> >>
> >>
> >> (as far as I can see no-one has made this use-case desirable in
> >> practice yet.)
> >>
> >> Tim.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >>
> >> WEBRTC mailing list
> >>
> >> WEBRTC@ietf.org
> >>
> >> https://www.ietf.org/mailman/listinfo/WEBRTC
> >>
> >>
> >> _______________________________________________ WEBRTC mailing list
> >> WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> >>
> > _______________________________________________ WEBRTC mailing list
> > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> > _______________________________________________ WEBRTC mailing list
> > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> 
> _______________________________________________
> WEBRTC mailing list
> WEBRTC@ietf.org
> https://www.ietf.org/mailman/listinfo/WEBRTC
> _______________________________________________
> WEBRTC mailing list
> WEBRTC@ietf.org
> https://www.ietf.org/mailman/listinfo/WEBRTC


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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Wed, 2 May 2012 19:54:03 +0200
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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2012/5/2 Dan Wing <dwing@cisco.com>:
> For other decisions, such as if we disallow un-encrypted RTP by
> WEBRTC endpoints, we create a requirement that some device does
> the interworking between WEBRTC endpoints (which do only SRTP)
> and non-WEBRTC endpoints (which do RTP). =C2=A0That means, for that
> interworking, we would adopt the interworking model on slide 7
> that I presented at IETF83,
> http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf

Hi, the link is broken (404). Could you please point to a working one?

Thanks a lot.

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

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From: Marshall Eubanks <marshall.eubanks@gmail.com>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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On Wed, May 2, 2012 at 1:54 PM, I=F1aki Baz Castillo <ibc@aliax.net> wrote:
> 2012/5/2 Dan Wing <dwing@cisco.com>:
>> For other decisions, such as if we disallow un-encrypted RTP by
>> WEBRTC endpoints, we create a requirement that some device does
>> the interworking between WEBRTC endpoints (which do only SRTP)
>> and non-WEBRTC endpoints (which do RTP). =A0That means, for that
>> interworking, we would adopt the interworking model on slide 7
>> that I presented at IETF83,
>> http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
>
> Hi, the link is broken (404). Could you please point to a working one?
>

http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf

> However, when I presented slide 7, there were objections at the microphon=
e that this model 'is broken'.  I would like to understand the objections s=
o we can reach consensus on how interworking from WEBRTC to non-WEBRTC is e=
xpected to occur.

My objection is that the proposed system will require middleware to
interoperate with the
vast number of videoconferencing sessions out there, most of which use
RTP. From the
standpoint of a video service provider, buying hardware to support
video to laptops is likely to
lead to requests that participants download some other software which
interoperates natively.

This is an existing business with a fairly large scale and installed
base. Not operating the way that they do is not likely to go over
well.

Regards
Marshall

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

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Date: Wed, 2 May 2012 20:20:12 +0200
From: Lorenzo Miniero <lorenzo@meetecho.com>
To: =?UTF-8?B?ScOxYWtp?= Baz Castillo <ibc@aliax.net>
Message-ID: <20120502202012.387fcff0@rainpc>
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <013101cd288c$09328250$1b9786f0$@com> <CALiegf=RsHHf9jCBhE55t7qFUpts8yJ1c8qUX12nc_vd4vgjSQ@mail.gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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On Wed, 2 May 2012 19:54:03 +0200
I=C3=B1aki Baz Castillo <ibc@aliax.net> wrote:

> 2012/5/2 Dan Wing <dwing@cisco.com>:
> > For other decisions, such as if we disallow un-encrypted RTP by
> > WEBRTC endpoints, we create a requirement that some device does
> > the interworking between WEBRTC endpoints (which do only SRTP)
> > and non-WEBRTC endpoints (which do RTP). =C2=A0That means, for that
> > interworking, we would adopt the interworking model on slide 7
> > that I presented at IETF83,
> > http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
>=20
> Hi, the link is broken (404). Could you please point to a working one?
>=20
> Thanks a lot.
>=20


This one seems to work (just replace WEBRTC with rtcweb):

http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf

L.

> --=20
> I=C3=B1aki Baz Castillo
> <ibc@aliax.net>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

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Date: Wed, 2 May 2012 15:27:16 -0500
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: Dan Wing <dwing@cisco.com>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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I agree with you in general, however, the link to your slides seems to be
broken.

Mary.

On Wed, May 2, 2012 at 12:50 PM, Dan Wing <dwing@cisco.com> wrote:

> > -----Original Message-----
> > From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> > Behalf Of Jim Barnett
> > Sent: Wednesday, May 02, 2012 7:39 AM
> > To: Stefan Hakansson LK; WEBRTC@ietf.org
> > Subject: Re: [WEBRTC] Use Case draft
> >
> > When I say that this use case may not add further requirements, I mean
> > that it looks like it would be possible to implement it given the
> > current definitions of the protocols.  However, the current use cases
> > are all written in terms of "the browser", which is not further
> > defined.
> > But if "browser" means Mozilla, Chrome, etc., then I think it is
> > important to add a use case in which one of the end points is not a
> > browser, but an enterprise gateway (which will route the call to an
> > employee of its choice, and may record the call, etc.) It is important
> > to note that this is not a peer-to-peer use case; the gateway is not
> > the
> > caller's peer.  The employee that the caller ends up talking to may be
> > considered a peer, but the webRTC call does not extend all the way to
> > that employee - it stops at the gateway.
> >
> > This is a very different use case from any in the current document.
> > That's why it's important to add it, even though (as far as I can tell)
> > it doesn't require us to change any of the work we've done.
>
> Somewhere, we need consensus on a model for interworking WEBRTC
> endpoints with non-WEBRTC endpoints.
>
> The decision comes down to this:
>
>  1. encumber WEBRTC endpoints with the interworking
>     effort, or
>  2. encumber a separate interworking device with the
>     interworking effort.
>
> I believe we have a better chance of success with (2), where
> possible to do (2).
>
> For some decisions, such as Consent Freshness (previously called Voice
> Hammer Attack in http://tools.ietf.org/html/rfc5245#section-18.5.1),
> non-WEBRTC endpoints need to respond to those ICE connectivity
> checks or have a gateway in front of them that responds to those
> connectivity checks on their behalf.  This means that WEBRTC
> cannot work directly with some existing SIP equipment (because
> a lot of SIP equipment does not support ICE).
>
> For other decisions, such as if we disallow un-encrypted RTP by
> WEBRTC endpoints, we create a requirement that some device does
> the interworking between WEBRTC endpoints (which do only SRTP)
> and non-WEBRTC endpoints (which do RTP).  That means, for that
> interworking, we would adopt the interworking model on slide 7
> that I presented at IETF83,
> http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
>
> However, when I presented slide 7, there were objections at the
> microphone that this model 'is broken'.  I would like to understand
> the objections so we can reach consensus on how interworking from
> WEBRTC to non-WEBRTC is expected to occur.
>
> -d
>
>
> > - Jim
> > -----Original Message-----
> > From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> > Behalf
> > Of Stefan Hakansson LK
> > Sent: Wednesday, May 02, 2012 4:46 AM
> > To: WEBRTC@ietf.org
> > Subject: Re: [WEBRTC] Use Case draft
> >
> > On 05/01/2012 02:05 PM, Jim Barnett wrote:
> > > One way to describe the use case is to let the contact center's media
> > > server/gateway serve as the webRTC endpoint.  Then all the issues of
> > > call delivery, call monitoring, etc. disappear.  They are handled by
> > > application software that sits behind the webRTC endpoint.  The
> > > company I work for makes a good living selling software that deals
> > > with all these issues - including bathroom breaks - and that's how we
> > > would tend to think of this case.  To us, it's a new kind of
> > > call/connection coming into the contact center, which we translate
> > > into SIP at the border and then handle normally.
> > >
> > > It's not clear to me if this use case adds any extra requirements.
> >
> > I think this is important to sort out. If the use case does not add any
> > extra requirements, what's the point of adding it?
> >
> > > We would just have to be careful not to assume that a webRTC endpoint
> > > is always a person/browser-based user agent.  It may seem a bit
> > > unsettling that the webRTC endpoint can distribute the call somewhere
> > > else and let others listen in, but as far as I can tell that is
> > > already the case.  If Bob calls Alice with full authentication and
> > > security, he can be sure that he is connected to Alice's user agent
> > > and that no one in between can listen in, but there's nothing
> > stopping
> >
> > > Alice from recording the audio, or forwarding it to a third party.
> > So
> >
> > > Bob could in fact be talking to Mary if that's how Alice wants to
> > > arrange things (_behind_ her user agent).  In general, Bob is assured
> > > only that he is talking to someone Alice wants him to talk to, and
> > > that no one can snoop without Alice's permission.  That's very much
> > > the way things work with the call center - you are sure that you are
> > > 1) connected securely to your bank 2) talking to someone that the
> > bank
> >
> > > wants you to talk to 3) being recorded or snooped on only when the
> > > bank explicitly chooses to do so.
> > >
> > > - Jim
> > >
> > > -----Original Message----- From: WEBRTC-bounces@ietf.org
> > > [mailto:WEBRTC-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
> > > Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
> > > WEBRTC@ietf.org Subject: Re: [WEBRTC] Use Case draft
> > >
> > > On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
> > > Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
> > >> Whether anybody has been successful in the past with this type of
> > use
> >
> > >> case is I think irrelevant.
> > >>
> > >>
> > >>
> > >> The enterprise call centre use case is I think a vital use case
> > >> because it is a scenario in which one user is only concerned that
> > >> they can securely reach an organization/domain and is not concerned
> > >> about the individual within that domain  that they communicate with.
> >
> > >> A suspect quite a large percentage of WEBRTC applications will be
> > >> like this and it is not covered in the current use case draft.
> > >
> > > I agree that this is a very useful use case and one I think is going
> > > to get a lot of traction. There is a very solid business case for
> > > this.  However, I have a fair amount of experience with a video call
> > > center for a client, and it is not as simple as it might seem.
> > >
> > > The essence of course is that you get the next available person,
> > i.e.,
> >
> > > it is anycast. Determining who the next available person is is not
> > > trivial, nor is error recovery. (If I call you, and you don't answer
> > > or the call drops or whatever,  I can leave a message or try later.
> > If
> >
> > > I call a help desk, and this happens, I want a new agent, ideally
> > > automatically.) Call forwarding (e.g., first tier to second tier
> > > technical support) is essential, and it may be anycast or directed.
> > > There are also some security oddities  - if I am connecting from
> > home,
> >
> > > I may need to authenticate, use a credit card, etc. If I am
> > connecting
> >
> > > from inside a store, and providing in store video technical support
> > is
> >
> > > big part of the market, then the store authenticates me off line and
> > > the call really should just be a button push, which implies that the
> > > store has previously authenticated some sort of master session. In
> > > addition, unlike most video calls, in the enterprise call center a
> > > supervisor may need to be able to monitor (i.e., watch) a call, and
> > in
> >
> > > some circumstances (financial or medical calls, for example) there
> > > will need to be third party recording. I believe that  these details
> > > would be different from the typical WEBRTC scenario.
> > >
> > > Also, there will be a temptation to do the anycasting by the
> > > techniques used to load balance servers in a data center, but I think
> > > that may not be sufficient. The call "center" may in fact be spread
> > > completely across the planet (daytime support in the US, nighttime
> > > support in India, for example) and be on multiple autonomous systems
> > > (and even from people's homes), which gives rise to some of the
> > > transport issues NVO3 may face, but without any opportunity for
> > packet
> >
> > > tagging. Plus, there will complicated rules about who can be selected
> > > next. WEBRTC shouldn't worry about the intricacies of bathroom break
> > > policies; these complexities should be dealt with by an
> > > enterprise-side database, which to me (together with some of the
> > other
> >
> > > issues above) suggests that this would probably benefit from API
> > > support.
> > >
> > > Regards Marshall
> > >
> > >
> > >>
> > >>
> > >>
> > >> So I think we need it.
> > >>
> > >>
> > >>
> > >> Regards
> > >>
> > >> Andy
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
> > >> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
> > >> WEBRTC@ietf.org
> > >>
> > >>
> > >> Subject: Re: [WEBRTC] Use Case draft
> > >>
> > >>
> > >>
> > >> Without numbers it is impossible to argue, but, if we talk about the
> > >> perceived need, I disagree.  Think of the people who travel abroad
> > >> and cannot call the 800 number. (I routinely use Web interface for
> > >> calls when traveling.)
> > >>
> > >>
> > >>
> > >> I am all for  the use case, as described by Jim.
> > >>
> > >> Igor
> > >>
> > >> On 4/30/2012 9:54 AM, Tim Panton wrote:
> > >>
> > >> ...
> > >>
> > >> I can't tell you the actual numbers, but when presented with the
> > >> choice of calling a toll free number
> > >>
> > >> or clicking a button marked "free internet call" - almost no-one on
> > a
> >
> > >> real, busy site clicked the button.
> > >>
> > >> ( for every button click there were several orders of magnitude more
> > >> 0800 calls from that page).
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> So from my perspective this is a legacy interop use case with almost
> > >> zero user acceptance.
> > >>
> > >>
> > >>
> > >> (as far as I can see no-one has made this use-case desirable in
> > >> practice yet.)
> > >>
> > >> Tim.
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> _______________________________________________
> > >>
> > >> WEBRTC mailing list
> > >>
> > >> WEBRTC@ietf.org
> > >>
> > >> https://www.ietf.org/mailman/listinfo/WEBRTC
> > >>
> > >>
> > >> _______________________________________________ WEBRTC mailing list
> > >> WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> > >>
> > > _______________________________________________ WEBRTC mailing list
> > > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> > > _______________________________________________ WEBRTC mailing list
> > > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> >
> > _______________________________________________
> > WEBRTC mailing list
> > WEBRTC@ietf.org
> > https://www.ietf.org/mailman/listinfo/WEBRTC
> > _______________________________________________
> > WEBRTC mailing list
> > WEBRTC@ietf.org
> > https://www.ietf.org/mailman/listinfo/WEBRTC
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d043c7c78625ca904bf1383c4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

I agree with you in general, however, the link to your slides seems to be b=
roken.<div><br></div><div>Mary.=A0<br><br><div class=3D"gmail_quote">On Wed=
, May 2, 2012 at 12:50 PM, Dan Wing <span dir=3D"ltr">&lt;<a href=3D"mailto=
:dwing@cisco.com" target=3D"_blank">dwing@cisco.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">&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounces@ie=
tf.org</a>] On<br>
&gt; Behalf Of Jim Barnett<br>
&gt; Sent: Wednesday, May 02, 2012 7:39 AM<br>
&gt; To: Stefan Hakansson LK; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@iet=
f.org</a><br>
&gt; Subject: Re: [WEBRTC] Use Case draft<br>
&gt;<br>
&gt; When I say that this use case may not add further requirements, I mean=
<br>
&gt; that it looks like it would be possible to implement it given the<br>
&gt; current definitions of the protocols. =A0However, the current use case=
s<br>
&gt; are all written in terms of &quot;the browser&quot;, which is not furt=
her<br>
&gt; defined.<br>
&gt; But if &quot;browser&quot; means Mozilla, Chrome, etc., then I think i=
t is<br>
&gt; important to add a use case in which one of the end points is not a<br=
>
&gt; browser, but an enterprise gateway (which will route the call to an<br=
>
&gt; employee of its choice, and may record the call, etc.) It is important=
<br>
&gt; to note that this is not a peer-to-peer use case; the gateway is not<b=
r>
&gt; the<br>
&gt; caller&#39;s peer. =A0The employee that the caller ends up talking to =
may be<br>
&gt; considered a peer, but the webRTC call does not extend all the way to<=
br>
&gt; that employee - it stops at the gateway.<br>
&gt;<br>
&gt; This is a very different use case from any in the current document.<br=
>
&gt; That&#39;s why it&#39;s important to add it, even though (as far as I =
can tell)<br>
&gt; it doesn&#39;t require us to change any of the work we&#39;ve done.<br=
>
<br>
Somewhere, we need consensus on a model for interworking WEBRTC<br>
endpoints with non-WEBRTC endpoints.<br>
<br>
The decision comes down to this:<br>
<br>
 =A01. encumber WEBRTC endpoints with the interworking<br>
 =A0 =A0 effort, or<br>
 =A02. encumber a separate interworking device with the<br>
 =A0 =A0 interworking effort.<br>
<br>
I believe we have a better chance of success with (2), where<br>
possible to do (2).<br>
<br>
For some decisions, such as Consent Freshness (previously called Voice<br>
Hammer Attack in <a href=3D"http://tools.ietf.org/html/rfc5245#section-18.5=
.1" target=3D"_blank">http://tools.ietf.org/html/rfc5245#section-18.5.1</a>=
),<br>
non-WEBRTC endpoints need to respond to those ICE connectivity<br>
checks or have a gateway in front of them that responds to those<br>
connectivity checks on their behalf. =A0This means that WEBRTC<br>
cannot work directly with some existing SIP equipment (because<br>
a lot of SIP equipment does not support ICE).<br>
<br>
For other decisions, such as if we disallow un-encrypted RTP by<br>
WEBRTC endpoints, we create a requirement that some device does<br>
the interworking between WEBRTC endpoints (which do only SRTP)<br>
and non-WEBRTC endpoints (which do RTP). =A0That means, for that<br>
interworking, we would adopt the interworking model on slide 7<br>
that I presented at IETF83,<br>
<a href=3D"http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf=
" target=3D"_blank">http://www.ietf.org/proceedings/83/slides/slides-83-WEB=
RTC-3.pdf</a><br>
<br>
However, when I presented slide 7, there were objections at the<br>
microphone that this model &#39;is broken&#39;. =A0I would like to understa=
nd<br>
the objections so we can reach consensus on how interworking from<br>
WEBRTC to non-WEBRTC is expected to occur.<br>
<br>
-d<br>
<br>
<br>
&gt; - Jim<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounces@ie=
tf.org</a>] On<br>
&gt; Behalf<br>
&gt; Of Stefan Hakansson LK<br>
&gt; Sent: Wednesday, May 02, 2012 4:46 AM<br>
&gt; To: <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a><br>
&gt; Subject: Re: [WEBRTC] Use Case draft<br>
&gt;<br>
&gt; On 05/01/2012 02:05 PM, Jim Barnett wrote:<br>
&gt; &gt; One way to describe the use case is to let the contact center&#39=
;s media<br>
&gt; &gt; server/gateway serve as the webRTC endpoint. =A0Then all the issu=
es of<br>
&gt; &gt; call delivery, call monitoring, etc. disappear. =A0They are handl=
ed by<br>
&gt; &gt; application software that sits behind the webRTC endpoint. =A0The=
<br>
&gt; &gt; company I work for makes a good living selling software that deal=
s<br>
&gt; &gt; with all these issues - including bathroom breaks - and that&#39;=
s how we<br>
&gt; &gt; would tend to think of this case. =A0To us, it&#39;s a new kind o=
f<br>
&gt; &gt; call/connection coming into the contact center, which we translat=
e<br>
&gt; &gt; into SIP at the border and then handle normally.<br>
&gt; &gt;<br>
&gt; &gt; It&#39;s not clear to me if this use case adds any extra requirem=
ents.<br>
&gt;<br>
&gt; I think this is important to sort out. If the use case does not add an=
y<br>
&gt; extra requirements, what&#39;s the point of adding it?<br>
&gt;<br>
&gt; &gt; We would just have to be careful not to assume that a webRTC endp=
oint<br>
&gt; &gt; is always a person/browser-based user agent. =A0It may seem a bit=
<br>
&gt; &gt; unsettling that the webRTC endpoint can distribute the call somew=
here<br>
&gt; &gt; else and let others listen in, but as far as I can tell that is<b=
r>
&gt; &gt; already the case. =A0If Bob calls Alice with full authentication =
and<br>
&gt; &gt; security, he can be sure that he is connected to Alice&#39;s user=
 agent<br>
&gt; &gt; and that no one in between can listen in, but there&#39;s nothing=
<br>
&gt; stopping<br>
&gt;<br>
&gt; &gt; Alice from recording the audio, or forwarding it to a third party=
.<br>
&gt; So<br>
&gt;<br>
&gt; &gt; Bob could in fact be talking to Mary if that&#39;s how Alice want=
s to<br>
&gt; &gt; arrange things (_behind_ her user agent). =A0In general, Bob is a=
ssured<br>
&gt; &gt; only that he is talking to someone Alice wants him to talk to, an=
d<br>
&gt; &gt; that no one can snoop without Alice&#39;s permission. =A0That&#39=
;s very much<br>
&gt; &gt; the way things work with the call center - you are sure that you =
are<br>
&gt; &gt; 1) connected securely to your bank 2) talking to someone that the=
<br>
&gt; bank<br>
&gt;<br>
&gt; &gt; wants you to talk to 3) being recorded or snooped on only when th=
e<br>
&gt; &gt; bank explicitly chooses to do so.<br>
&gt; &gt;<br>
&gt; &gt; - Jim<br>
&gt; &gt;<br>
&gt; &gt; -----Original Message----- From: <a href=3D"mailto:WEBRTC-bounces=
@ietf.org">WEBRTC-bounces@ietf.org</a><br>
&gt; &gt; [mailto:<a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounces=
@ietf.org</a>] On Behalf Of Marshall Eubanks Sent:<br>
&gt; &gt; Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:<br>
&gt; &gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a> Subject: R=
e: [WEBRTC] Use Case draft<br>
&gt; &gt;<br>
&gt; &gt; On Mon, Apr 30, 2012 at 2:31 PM, Hutton,<br>
&gt; &gt; Andrew&lt;<a href=3D"mailto:andrew.hutton@siemens-enterprise.com"=
>andrew.hutton@siemens-enterprise.com</a>&gt; =A0wrote:<br>
&gt; &gt;&gt; Whether anybody has been successful in the past with this typ=
e of<br>
&gt; use<br>
&gt;<br>
&gt; &gt;&gt; case is I think irrelevant.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; The enterprise call centre use case is I think a vital use ca=
se<br>
&gt; &gt;&gt; because it is a scenario in which one user is only concerned =
that<br>
&gt; &gt;&gt; they can securely reach an organization/domain and is not con=
cerned<br>
&gt; &gt;&gt; about the individual within that domain =A0that they communic=
ate with.<br>
&gt;<br>
&gt; &gt;&gt; A suspect quite a large percentage of WEBRTC applications wil=
l be<br>
&gt; &gt;&gt; like this and it is not covered in the current use case draft=
.<br>
&gt; &gt;<br>
&gt; &gt; I agree that this is a very useful use case and one I think is go=
ing<br>
&gt; &gt; to get a lot of traction. There is a very solid business case for=
<br>
&gt; &gt; this. =A0However, I have a fair amount of experience with a video=
 call<br>
&gt; &gt; center for a client, and it is not as simple as it might seem.<br=
>
&gt; &gt;<br>
&gt; &gt; The essence of course is that you get the next available person,<=
br>
&gt; i.e.,<br>
&gt;<br>
&gt; &gt; it is anycast. Determining who the next available person is is no=
t<br>
&gt; &gt; trivial, nor is error recovery. (If I call you, and you don&#39;t=
 answer<br>
&gt; &gt; or the call drops or whatever, =A0I can leave a message or try la=
ter.<br>
&gt; If<br>
&gt;<br>
&gt; &gt; I call a help desk, and this happens, I want a new agent, ideally=
<br>
&gt; &gt; automatically.) Call forwarding (e.g., first tier to second tier<=
br>
&gt; &gt; technical support) is essential, and it may be anycast or directe=
d.<br>
&gt; &gt; There are also some security oddities =A0- if I am connecting fro=
m<br>
&gt; home,<br>
&gt;<br>
&gt; &gt; I may need to authenticate, use a credit card, etc. If I am<br>
&gt; connecting<br>
&gt;<br>
&gt; &gt; from inside a store, and providing in store video technical suppo=
rt<br>
&gt; is<br>
&gt;<br>
&gt; &gt; big part of the market, then the store authenticates me off line =
and<br>
&gt; &gt; the call really should just be a button push, which implies that =
the<br>
&gt; &gt; store has previously authenticated some sort of master session. I=
n<br>
&gt; &gt; addition, unlike most video calls, in the enterprise call center =
a<br>
&gt; &gt; supervisor may need to be able to monitor (i.e., watch) a call, a=
nd<br>
&gt; in<br>
&gt;<br>
&gt; &gt; some circumstances (financial or medical calls, for example) ther=
e<br>
&gt; &gt; will need to be third party recording. I believe that =A0these de=
tails<br>
&gt; &gt; would be different from the typical WEBRTC scenario.<br>
&gt; &gt;<br>
&gt; &gt; Also, there will be a temptation to do the anycasting by the<br>
&gt; &gt; techniques used to load balance servers in a data center, but I t=
hink<br>
&gt; &gt; that may not be sufficient. The call &quot;center&quot; may in fa=
ct be spread<br>
&gt; &gt; completely across the planet (daytime support in the US, nighttim=
e<br>
&gt; &gt; support in India, for example) and be on multiple autonomous syst=
ems<br>
&gt; &gt; (and even from people&#39;s homes), which gives rise to some of t=
he<br>
&gt; &gt; transport issues NVO3 may face, but without any opportunity for<b=
r>
&gt; packet<br>
&gt;<br>
&gt; &gt; tagging. Plus, there will complicated rules about who can be sele=
cted<br>
&gt; &gt; next. WEBRTC shouldn&#39;t worry about the intricacies of bathroo=
m break<br>
&gt; &gt; policies; these complexities should be dealt with by an<br>
&gt; &gt; enterprise-side database, which to me (together with some of the<=
br>
&gt; other<br>
&gt;<br>
&gt; &gt; issues above) suggests that this would probably benefit from API<=
br>
&gt; &gt; support.<br>
&gt; &gt;<br>
&gt; &gt; Regards Marshall<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; So I think we need it.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Regards<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Andy<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; From: <a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-bounc=
es@ietf.org</a> [mailto:<a href=3D"mailto:WEBRTC-bounces@ietf.org">WEBRTC-b=
ounces@ietf.org</a>] On<br>
&gt; &gt;&gt; Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:<br>
&gt; &gt;&gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a><br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Subject: Re: [WEBRTC] Use Case draft<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Without numbers it is impossible to argue, but, if we talk ab=
out the<br>
&gt; &gt;&gt; perceived need, I disagree. =A0Think of the people who travel=
 abroad<br>
&gt; &gt;&gt; and cannot call the 800 number. (I routinely use Web interfac=
e for<br>
&gt; &gt;&gt; calls when traveling.)<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I am all for =A0the use case, as described by Jim.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Igor<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; On 4/30/2012 9:54 AM, Tim Panton wrote:<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; ...<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I can&#39;t tell you the actual numbers, but when presented w=
ith the<br>
&gt; &gt;&gt; choice of calling a toll free number<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; or clicking a button marked &quot;free internet call&quot; - =
almost no-one on<br>
&gt; a<br>
&gt;<br>
&gt; &gt;&gt; real, busy site clicked the button.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; ( for every button click there were several orders of magnitu=
de more<br>
&gt; &gt;&gt; 0800 calls from that page).<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; So from my perspective this is a legacy interop use case with=
 almost<br>
&gt; &gt;&gt; zero user acceptance.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; (as far as I can see no-one has made this use-case desirable =
in<br>
&gt; &gt;&gt; practice yet.)<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Tim.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; _______________________________________________<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; WEBRTC mailing list<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a><br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/WEBRTC" targ=
et=3D"_blank">https://www.ietf.org/mailman/listinfo/WEBRTC</a><br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; _______________________________________________ WEBRTC mailin=
g list<br>
&gt; &gt;&gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a> <a hre=
f=3D"https://www.ietf.org/mailman/listinfo/WEBRTC" target=3D"_blank">https:=
//www.ietf.org/mailman/listinfo/WEBRTC</a><br>
&gt; &gt;&gt;<br>
&gt; &gt; _______________________________________________ WEBRTC mailing li=
st<br>
&gt; &gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a> <a href=3D=
"https://www.ietf.org/mailman/listinfo/WEBRTC" target=3D"_blank">https://ww=
w.ietf.org/mailman/listinfo/WEBRTC</a><br>
&gt; &gt; _______________________________________________ WEBRTC mailing li=
st<br>
&gt; &gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a> <a href=3D=
"https://www.ietf.org/mailman/listinfo/WEBRTC" target=3D"_blank">https://ww=
w.ietf.org/mailman/listinfo/WEBRTC</a><br>
&gt;<br>
&gt; _______________________________________________<br>
&gt; WEBRTC mailing list<br>
&gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/WEBRTC" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/WEBRTC</a><br>
&gt; _______________________________________________<br>
&gt; WEBRTC mailing list<br>
&gt; <a href=3D"mailto:WEBRTC@ietf.org">WEBRTC@ietf.org</a><br>
&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/WEBRTC" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/WEBRTC</a><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>
</blockquote></div><br></div>

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To: Christer Holmberg <christer.holmberg@ericsson.com>, Roman Shpount <roman@telurix.com>
Date: Wed, 2 May 2012 15:27:53 -0500
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Christer,
Regarding whether we need to be able to simultaneously support multiple rem=
ote locations (forking), I have reluctantly concluded "yes" after thinking =
about it for a bit.

The UPDATE scenario from Partha is not the only reason we need the concept =
of peer connection cloning (or something similar), although it is sufficien=
t in itself.  As soon as you get multiple answers from different forked tar=
gets, it is necessary to separately keep track of ICE and DTLS progress wit=
h each target, not to mention RTCP.  Otherwise ICE and DTLS results from on=
e target will get overwritten by those from another and then need to be red=
one again if an earlier target is selected as the winner.  To me this is no=
t acceptable, hence my reluctant conclusion.

Maybe it could work if we simply delay ICE, DTLS, RTCP and media until we d=
ecide on a final ANSWER, and any nested offer/answer transactions (i.e., Pa=
rtha's example) are handled at the JS layer only without interaction with t=
he peer connection.  Does anyone think this is an acceptable approach?  Not=
 I.

Richard

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: Wednesday, May 02, 2012 11:29 AM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

Hi,

>> Do people think that is not enough, and that we would need to support mu=
ltiple remote locations=20
>> simultanously (no matter whether it's implemented using cloning or somet=
hing else)?
>
> I never thought that was enough if we are to interop with anything that s=
upport forking. Latest example from > Partha shows that this is indeed the =
case.
>
> I actually think that adding support for forking is trivial (order of mag=
nitude less complcated then DTLS-
> SRTP or identity) and would simplify interop as well as enable numerous a=
dditional calling scenarios.

It's not about forking as such - I also agree we need to support that.

The question is whether we need to *simultanously* need to support multiple=
 remote locations, or whether we only have one "active" remote location, bu=
t we allow switching from one to another (based on where the previous answe=
r can from, or whatever policy).

Regards,

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

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 2 May 2012 15:34:22 -0500
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Harald,
Thanks for the challenge.

I do agree with Roman that there is considerable overlap between forking an=
d bundle that we should be able to leverage in a solution.

I need a few days to think this through before I am ready to propose anythi=
ng specific.

BR, Richard

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Wednesday, May 02, 2012 8:35 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

On 05/01/2012 03:57 PM, Ejzak, Richard P (Richard) wrote:
Hi Justin, Partha,
The call flow from Partha is a very reasonable one in the presence of SIP f=
orking.  Let's assume that a first target responded with an SDP answer in a=
 183 that must be treated as a PRANSWER since other targets have not yet ha=
d a chance to respond.  Then the first target sends an UPDATE with offer2.

As far as SIP is concerned, there is no such thing as a PRANSWER.  The firs=
t target already responded with an SDP ANSWER and wants to UPDATE with a ne=
w offer.
In that case, PRANSWER seems like the wrong semantic.... if you want to sup=
port multiple answers, shouldn't you do the cloning before you start accept=
ing answers?

 This is perfectly legitimate and common SIP usage.  PRANSWER is a local co=
nstruct that should be usable by the application to support more complex of=
fer/answer cases like forking.

You must respond to this offer by cloning the peer connection, closing the =
clone with an ANSWER identical to the previous PRANSWER and then applying o=
ffer2 to the clone (or just applying the offer2 directly with the understan=
ding that the previous transaction is closed).  The original peer connectio=
n must revert to the original OFFER state to allow for other targets to res=
pond.

Alternately, the original peer connection can support the first target and =
a new clone created if there is the potential for other forked responses to=
 the original offer.  Either way can be made to work.

This is a perfectly reasonable SIP use case that explains why we need to be=
 able to clone the peer connection.

Hmm....
could you please propose a semantic for "clone" that allows us to evaluate =
whether/how it can be implemented?

Especially, I am worried about what the state of the peerconnection has to =
be before it is cloned, what the state of the peerconnection is after it is=
 cloned, which peerconnection owns the various resources allocated (ports a=
re the obvious part of it), and which peerconnection any streams (local or =
remote) attached to the peerconnection before the clone are attached to aft=
er the clone.

In an earlier thread, I suggested a peerconnection factory that could gener=
ate an offer, but required you to manufacture a real peerconnection before =
applying an answer (any answer).... since I'm not that interested in suppor=
ting forking, I haven't pursued that further. Others might want to.

I don't think any of this is impossible. I do think it requires rather prec=
ise definitions of what we want before we ask browser developers to do it.

           Harald

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Harald,<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks for the challenge.<o:p></o:p></=
span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I do agree with Roman that there is
considerable overlap between forking and bundle that we should be able to
leverage in a solution.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I need a few days to think this throug=
h before
I am ready to propose anything specific.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>BR, Richard<o:p></o:p></span></font></=
p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span style=3D'font-size:12.0pt;colo=
r:windowtext'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span style=3D'font-size:10.0pt;font-f=
amily:Tahoma;
color:windowtext'> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]=
 <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Harald Alvestrand<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Wednesday, May 02, 201=
2 8:35
AM<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> Re: [rtcweb] SDP_PR=
ANSWER
followed by SDP_OFFER scenario in JSEP</span></font><font color=3Dblack><sp=
an
style=3D'color:windowtext'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>On 05/01/2012 03:57 PM, Ejzak, Richard P (Richar=
d)
wrote: <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><!--[if gte mso 9]><xml>
 <u1:shapedefaults u2:ext=3D"edit" spidmax=3D"1026"/>
</xml><![endif]--><!--[if gte mso 9]><xml>
 <u3:shapelayout u4:ext=3D"edit">
  <u3:idmap u4:ext=3D"edit" data=3D"1"/>
 </u3:shapelayout>
</xml><![endif]-->Hi Justin, Partha,<u5:p></u5:p></span></font><o:p></o:p><=
/p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The call flow from Partha is a very
reasonable one in the presence of SIP forking. &nbsp;Let&#8217;s assume tha=
t a
first target responded with an SDP answer in a 183 that must be treated as =
a
PRANSWER since other targets have not yet had a chance to respond.&nbsp; Th=
en
the first target sends an UPDATE with offer2.<u5:p></u5:p></span></font><o:=
p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><u5:p>&nbsp;</u5:p></span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>As far as SIP is concerned, there is n=
o
such thing as a PRANSWER. &nbsp;The first target already responded with an =
SDP
ANSWER and wants to UPDATE with a new offer.</span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt'>In that case, PRANSWER seems like the wrong
semantic.... if you want to support multiple answers, shouldn't you do the
cloning before you start accepting answers?<br>
<br>
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>&nbsp;This is perfectly legitimate and=
 common
SIP usage.&nbsp; PRANSWER is a local construct that should be usable by the
application to support more complex offer/answer cases like forking.<u5:p><=
/u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><u5:p>&nbsp;</u5:p></span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>You must respond to this offer by clon=
ing
the peer connection, closing the clone with an ANSWER identical to the prev=
ious
PRANSWER and then applying offer2 to the clone (or just applying the offer2
directly with the understanding that the previous transaction is closed).
&nbsp;The original peer connection must revert to the original OFFER state =
to
allow for other targets to respond.<u5:p></u5:p></span></font><o:p></o:p></=
p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><u5:p>&nbsp;</u5:p></span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Alternately, the original peer connect=
ion
can support the first target and a new clone created if there is the potent=
ial
for other forked responses to the original offer. &nbsp;Either way can be m=
ade
to work.<u5:p></u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><u5:p>&nbsp;</u5:p></span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>This is a perfectly reasonable SIP use
case that explains why we need to be able to clone the peer connection.</sp=
an></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 color=3D=
black
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br>
Hmm....<br>
could you please propose a semantic for &quot;clone&quot; that allows us to
evaluate whether/how it can be implemented?<br>
<br>
Especially, I am worried about what the state of the peerconnection has to =
be
before it is cloned, what the state of the peerconnection is after it is
cloned, which peerconnection owns the various resources allocated (ports ar=
e
the obvious part of it), and which peerconnection any streams (local or rem=
ote)
attached to the peerconnection before the clone are attached to after the
clone.<br>
<br>
In an earlier thread, I suggested a peerconnection factory that could gener=
ate
an offer, but required you to manufacture a real peerconnection before appl=
ying
an answer (any answer).... since I'm not that interested in supporting fork=
ing,
I haven't pursued that further. Others might want to.<br>
<br>
I don't think any of this is impossible. I do think it requires rather prec=
ise
definitions of what we want before we ask browser developers to do it.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<o:p></o=
:p></span></font></p>

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Date: Wed, 2 May 2012 16:35:18 -0400
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From: Marshall Eubanks <marshall.eubanks@gmail.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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On Wed, May 2, 2012 at 4:27 PM, Mary Barnes <mary.ietf.barnes@gmail.com> wr=
ote:
> I agree with you in general, however, the link to your slides seems to be
> broken.
>

Mary;

http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf

Marshall

> Mary.
>
>
> On Wed, May 2, 2012 at 12:50 PM, Dan Wing <dwing@cisco.com> wrote:
>>
>> > -----Original Message-----
>> > From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
>> > Behalf Of Jim Barnett
>> > Sent: Wednesday, May 02, 2012 7:39 AM
>> > To: Stefan Hakansson LK; WEBRTC@ietf.org
>> > Subject: Re: [WEBRTC] Use Case draft
>> >
>> > When I say that this use case may not add further requirements, I mean
>> > that it looks like it would be possible to implement it given the
>> > current definitions of the protocols. =A0However, the current use case=
s
>> > are all written in terms of "the browser", which is not further
>> > defined.
>> > But if "browser" means Mozilla, Chrome, etc., then I think it is
>> > important to add a use case in which one of the end points is not a
>> > browser, but an enterprise gateway (which will route the call to an
>> > employee of its choice, and may record the call, etc.) It is important
>> > to note that this is not a peer-to-peer use case; the gateway is not
>> > the
>> > caller's peer. =A0The employee that the caller ends up talking to may =
be
>> > considered a peer, but the webRTC call does not extend all the way to
>> > that employee - it stops at the gateway.
>> >
>> > This is a very different use case from any in the current document.
>> > That's why it's important to add it, even though (as far as I can tell=
)
>> > it doesn't require us to change any of the work we've done.
>>
>> Somewhere, we need consensus on a model for interworking WEBRTC
>> endpoints with non-WEBRTC endpoints.
>>
>> The decision comes down to this:
>>
>> =A01. encumber WEBRTC endpoints with the interworking
>> =A0 =A0 effort, or
>> =A02. encumber a separate interworking device with the
>> =A0 =A0 interworking effort.
>>
>> I believe we have a better chance of success with (2), where
>> possible to do (2).
>>
>> For some decisions, such as Consent Freshness (previously called Voice
>> Hammer Attack in http://tools.ietf.org/html/rfc5245#section-18.5.1),
>> non-WEBRTC endpoints need to respond to those ICE connectivity
>> checks or have a gateway in front of them that responds to those
>> connectivity checks on their behalf. =A0This means that WEBRTC
>> cannot work directly with some existing SIP equipment (because
>> a lot of SIP equipment does not support ICE).
>>
>> For other decisions, such as if we disallow un-encrypted RTP by
>> WEBRTC endpoints, we create a requirement that some device does
>> the interworking between WEBRTC endpoints (which do only SRTP)
>> and non-WEBRTC endpoints (which do RTP). =A0That means, for that
>> interworking, we would adopt the interworking model on slide 7
>> that I presented at IETF83,
>> http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
>>
>> However, when I presented slide 7, there were objections at the
>> microphone that this model 'is broken'. =A0I would like to understand
>> the objections so we can reach consensus on how interworking from
>> WEBRTC to non-WEBRTC is expected to occur.
>>
>> -d
>>
>>
>> > - Jim
>> > -----Original Message-----
>> > From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
>> > Behalf
>> > Of Stefan Hakansson LK
>> > Sent: Wednesday, May 02, 2012 4:46 AM
>> > To: WEBRTC@ietf.org
>> > Subject: Re: [WEBRTC] Use Case draft
>> >
>> > On 05/01/2012 02:05 PM, Jim Barnett wrote:
>> > > One way to describe the use case is to let the contact center's medi=
a
>> > > server/gateway serve as the webRTC endpoint. =A0Then all the issues =
of
>> > > call delivery, call monitoring, etc. disappear. =A0They are handled =
by
>> > > application software that sits behind the webRTC endpoint. =A0The
>> > > company I work for makes a good living selling software that deals
>> > > with all these issues - including bathroom breaks - and that's how w=
e
>> > > would tend to think of this case. =A0To us, it's a new kind of
>> > > call/connection coming into the contact center, which we translate
>> > > into SIP at the border and then handle normally.
>> > >
>> > > It's not clear to me if this use case adds any extra requirements.
>> >
>> > I think this is important to sort out. If the use case does not add an=
y
>> > extra requirements, what's the point of adding it?
>> >
>> > > We would just have to be careful not to assume that a webRTC endpoin=
t
>> > > is always a person/browser-based user agent. =A0It may seem a bit
>> > > unsettling that the webRTC endpoint can distribute the call somewher=
e
>> > > else and let others listen in, but as far as I can tell that is
>> > > already the case. =A0If Bob calls Alice with full authentication and
>> > > security, he can be sure that he is connected to Alice's user agent
>> > > and that no one in between can listen in, but there's nothing
>> > stopping
>> >
>> > > Alice from recording the audio, or forwarding it to a third party.
>> > So
>> >
>> > > Bob could in fact be talking to Mary if that's how Alice wants to
>> > > arrange things (_behind_ her user agent). =A0In general, Bob is assu=
red
>> > > only that he is talking to someone Alice wants him to talk to, and
>> > > that no one can snoop without Alice's permission. =A0That's very muc=
h
>> > > the way things work with the call center - you are sure that you are
>> > > 1) connected securely to your bank 2) talking to someone that the
>> > bank
>> >
>> > > wants you to talk to 3) being recorded or snooped on only when the
>> > > bank explicitly chooses to do so.
>> > >
>> > > - Jim
>> > >
>> > > -----Original Message----- From: WEBRTC-bounces@ietf.org
>> > > [mailto:WEBRTC-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>> > > Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>> > > WEBRTC@ietf.org Subject: Re: [WEBRTC] Use Case draft
>> > >
>> > > On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>> > > Andrew<andrew.hutton@siemens-enterprise.com> =A0wrote:
>> > >> Whether anybody has been successful in the past with this type of
>> > use
>> >
>> > >> case is I think irrelevant.
>> > >>
>> > >>
>> > >>
>> > >> The enterprise call centre use case is I think a vital use case
>> > >> because it is a scenario in which one user is only concerned that
>> > >> they can securely reach an organization/domain and is not concerned
>> > >> about the individual within that domain =A0that they communicate wi=
th.
>> >
>> > >> A suspect quite a large percentage of WEBRTC applications will be
>> > >> like this and it is not covered in the current use case draft.
>> > >
>> > > I agree that this is a very useful use case and one I think is going
>> > > to get a lot of traction. There is a very solid business case for
>> > > this. =A0However, I have a fair amount of experience with a video ca=
ll
>> > > center for a client, and it is not as simple as it might seem.
>> > >
>> > > The essence of course is that you get the next available person,
>> > i.e.,
>> >
>> > > it is anycast. Determining who the next available person is is not
>> > > trivial, nor is error recovery. (If I call you, and you don't answer
>> > > or the call drops or whatever, =A0I can leave a message or try later=
.
>> > If
>> >
>> > > I call a help desk, and this happens, I want a new agent, ideally
>> > > automatically.) Call forwarding (e.g., first tier to second tier
>> > > technical support) is essential, and it may be anycast or directed.
>> > > There are also some security oddities =A0- if I am connecting from
>> > home,
>> >
>> > > I may need to authenticate, use a credit card, etc. If I am
>> > connecting
>> >
>> > > from inside a store, and providing in store video technical support
>> > is
>> >
>> > > big part of the market, then the store authenticates me off line and
>> > > the call really should just be a button push, which implies that the
>> > > store has previously authenticated some sort of master session. In
>> > > addition, unlike most video calls, in the enterprise call center a
>> > > supervisor may need to be able to monitor (i.e., watch) a call, and
>> > in
>> >
>> > > some circumstances (financial or medical calls, for example) there
>> > > will need to be third party recording. I believe that =A0these detai=
ls
>> > > would be different from the typical WEBRTC scenario.
>> > >
>> > > Also, there will be a temptation to do the anycasting by the
>> > > techniques used to load balance servers in a data center, but I thin=
k
>> > > that may not be sufficient. The call "center" may in fact be spread
>> > > completely across the planet (daytime support in the US, nighttime
>> > > support in India, for example) and be on multiple autonomous systems
>> > > (and even from people's homes), which gives rise to some of the
>> > > transport issues NVO3 may face, but without any opportunity for
>> > packet
>> >
>> > > tagging. Plus, there will complicated rules about who can be selecte=
d
>> > > next. WEBRTC shouldn't worry about the intricacies of bathroom break
>> > > policies; these complexities should be dealt with by an
>> > > enterprise-side database, which to me (together with some of the
>> > other
>> >
>> > > issues above) suggests that this would probably benefit from API
>> > > support.
>> > >
>> > > Regards Marshall
>> > >
>> > >
>> > >>
>> > >>
>> > >>
>> > >> So I think we need it.
>> > >>
>> > >>
>> > >>
>> > >> Regards
>> > >>
>> > >> Andy
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org] On
>> > >> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>> > >> WEBRTC@ietf.org
>> > >>
>> > >>
>> > >> Subject: Re: [WEBRTC] Use Case draft
>> > >>
>> > >>
>> > >>
>> > >> Without numbers it is impossible to argue, but, if we talk about th=
e
>> > >> perceived need, I disagree. =A0Think of the people who travel abroa=
d
>> > >> and cannot call the 800 number. (I routinely use Web interface for
>> > >> calls when traveling.)
>> > >>
>> > >>
>> > >>
>> > >> I am all for =A0the use case, as described by Jim.
>> > >>
>> > >> Igor
>> > >>
>> > >> On 4/30/2012 9:54 AM, Tim Panton wrote:
>> > >>
>> > >> ...
>> > >>
>> > >> I can't tell you the actual numbers, but when presented with the
>> > >> choice of calling a toll free number
>> > >>
>> > >> or clicking a button marked "free internet call" - almost no-one on
>> > a
>> >
>> > >> real, busy site clicked the button.
>> > >>
>> > >> ( for every button click there were several orders of magnitude mor=
e
>> > >> 0800 calls from that page).
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >> So from my perspective this is a legacy interop use case with almos=
t
>> > >> zero user acceptance.
>> > >>
>> > >>
>> > >>
>> > >> (as far as I can see no-one has made this use-case desirable in
>> > >> practice yet.)
>> > >>
>> > >> Tim.
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >>
>> > >> _______________________________________________
>> > >>
>> > >> WEBRTC mailing list
>> > >>
>> > >> WEBRTC@ietf.org
>> > >>
>> > >> https://www.ietf.org/mailman/listinfo/WEBRTC
>> > >>
>> > >>
>> > >> _______________________________________________ WEBRTC mailing list
>> > >> WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
>> > >>
>> > > _______________________________________________ WEBRTC mailing list
>> > > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
>> > > _______________________________________________ WEBRTC mailing list
>> > > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
>> >
>> > _______________________________________________
>> > WEBRTC mailing list
>> > WEBRTC@ietf.org
>> > https://www.ietf.org/mailman/listinfo/WEBRTC
>> > _______________________________________________
>> > WEBRTC mailing list
>> > WEBRTC@ietf.org
>> > https://www.ietf.org/mailman/listinfo/WEBRTC
>>
>> _______________________________________________
>> 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  Wed May  2 13:41:38 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>, Roman Shpount <roman@telurix.com>
Date: Wed, 2 May 2012 22:41:22 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>Regarding whether we need to be able to simultaneously support multiple re=
mote locations (forking), I have >reluctantly concluded "yes" after thinkin=
g about it for a bit.
>
>The UPDATE scenario from Partha is not the only reason we need the concept=
 of peer connection cloning (or >something similar), although it is suffici=
ent in itself.  As soon as you get multiple answers from different forked >=
targets, it is necessary to separately keep track of ICE and DTLS progress =
with each target, not to mention >RTCP.

...unless you, when you receive an answer from a new target, terminate the =
leg with the previous forked target.

>Otherwise ICE and DTLS results from one target will get overwritten by tho=
se from another and then need to be >redone again if an earlier target is s=
elected as the winner.=20

I think that is unlikely to happen, as forking is normally done in a serial=
 manner (to avoid other issues associated with parallel forking).

I am not saying that there aren't use-cases that would break, but my questi=
on is still whether we can live with that restriction.

Regards,

Christer



-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: Wednesday, May 02, 2012 11:29 AM
To: Roman Shpount
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

Hi,

>> Do people think that is not enough, and that we would need to support mu=
ltiple remote locations
>> simultanously (no matter whether it's implemented using cloning or somet=
hing else)?
>
> I never thought that was enough if we are to interop with anything that s=
upport forking. Latest example from > Partha shows that this is indeed the =
case.
>
> I actually think that adding support for forking is trivial (order of mag=
nitude less complcated then DTLS-
> SRTP or identity) and would simplify interop as well as enable numerous a=
dditional calling scenarios.

It's not about forking as such - I also agree we need to support that.

The question is whether we need to *simultanously* need to support multiple=
 remote locations, or whether we only have one "active" remote location, bu=
t we allow switching from one to another (based on where the previous answe=
r can from, or whatever policy).

Regards,

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

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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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On 5/2/12 7:50 PM, Dan Wing wrote:
http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf
> However, when I presented slide 7, there were objections at the 
> microphone that this model 'is broken'.  I would like to understand 
> the objections so we can reach consensus on how interworking from
> WEBRTC to non-WEBRTC is expected to occur.

IMHO it would be much easier, as described in tons of previous email, to
use different Key Management for different requirements:

- SDES + SRTP for end-to-site (peer to gateway)
- DTLS-SRTP for end-to-end (peer to peer)

It would be a simpler approach for:

- Security (the user know the security level)
- Interoperability (Use standard protocols for the need to interoperate)

That way the "simpler, legacy, standard" technology would be used for
all voip "server applications" while the new burning (yet not used by
anyone) DTLS-SRTP will be used for peer-to-peer calls.

Please DO NOT reinvent the wheel for what's already existing and deployed.

Doing so, it's like going against the natural human behavior, it would
just represent a failure.

Fabio

From roman@telurix.com  Wed May  2 14:13:24 2012
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--e89a8ff24a9b2f833304bf14286e
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On Wed, May 2, 2012 at 4:41 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:


> I think that is unlikely to happen, as forking is normally done in a
> serial manner (to avoid other issues associated with parallel forking).
>

I am not sure why you would say that most forking is serial. It is quite
often parallel. What is commonly done to avoid support for multiple
parallel media streams is playing media from the newest received RTP
stream. In either case there is no guarantee that you picked the winning
stream, and you always need to make sure that you use the SDP received in
the dialog that got 200 OK and that media playback switches to RTP stream
that remains. Making this work is often problematic in SIP since there is
no way to associate received RTP with received SDP information due to RTP
being sent from a different IP/port from the one used in SDP. It should be
easier with WebRTC since ICE support is required and remote party IP/port
can be determined for each flow.

Finally, when you are dealing with forking you need to keep track of the
remote IP/ports and ICE candidates for each dialog. You can play media from
only one of them, but this remote connection state is required to complete
the call setup when 200 OK is received.
_____________
Roman Shpount

--e89a8ff24a9b2f833304bf14286e
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<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Wed, May 2, 2012 a=
t 4:41 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christ=
er.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com<=
/a>&gt;</span> wrote:<br>
<div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt=
 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I think tha=
t is unlikely to happen, as forking is normally done in a serial manner (to=
 avoid other issues associated with parallel forking).<br>
</blockquote></div><br>I am not sure why you would say that most forking is=
 serial. It is quite often parallel. What is commonly done to avoid support=
 for multiple parallel media streams is playing media from the newest recei=
ved RTP stream. In either case there is no guarantee that you picked the wi=
nning stream, and you always need to make sure that you use the SDP receive=
d in the dialog that got 200 OK and that media playback switches to RTP str=
eam that remains. Making this work is often problematic in SIP since there =
is no way to associate received RTP with received SDP information due to RT=
P being sent from a different IP/port from the one used in SDP. It should b=
e easier with WebRTC since ICE support is required and remote party IP/port=
 can be determined for each flow.<br>
<br>Finally, when you are dealing with forking you need to keep track of th=
e remote IP/ports and ICE candidates for each dialog. You can play media fr=
om only one of them, but this remote connection state is required to comple=
te the call setup when 200 OK is received.<br>
_____________<br>Roman Shpount<br>
<br><br></div>

--e89a8ff24a9b2f833304bf14286e--

From fluffy@iii.ca  Wed May  2 14:32:48 2012
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Subject: Re: [rtcweb] Final plea about SRTP
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Roman,=20

One comment on this - I think people understand there could be services =
with no security requirements that could run over RTP, and HTTP, with no =
identity. But we need to have a secure solution for some other services. =
The questions is once you have a secure solution, what is the incentive =
to also support an insecure solution - so far no one has come up with a =
super compelling story about dealing with the bid down and I suspect =
that lots of people did not view the overhead of running the secure =
version as all that high. I suspect that is part of why the decisions =
went the way it did - basically people agreed we needed a secure =
solution, and when they considered also having an insecure solution, =
they saw lots of complications of doing both and not much gain in the =
insecure solution over the secure solution.=20

Cullen



On May 2, 2012, at 10:03 AM, Roman Shpount wrote:

> I know there was a consensus call on this list that SRTP shall be used =
for all the calls in WebRTC, but I still do not understand the =
justification for this requirement for WebRTC applications delivered =
over HTTP with no identity. For such scenarios SRTP (even DTLS-SRTP) =
serves almost no purpose. If application is delivered over HTTP attacker =
can spoof the entire web site. It is trivial if the attacker is on the =
communications path. If attacker is seating in the airport using the =
same network, it can put itself on the communications path using arp =
cache poisoning. Once the web site is spoofed, any type of man in the =
middle attack can be implemented. If DTLS-SRTP is used user can detect =
the attack by checking the key signature, but in reality very few people =
will do this.
>=20
> The main argument to require SRTP everywhere was that it does not =
break anything. But neither would naming all the API methods in High =
Elfish. Either requirement does not break things, but make working with =
WebRTC harder then it should. At the same time both of those =
requirements are completely unjustified.
>=20
> Furthermore, assumption on this list that most of the WebRTC use would =
be peer-to-peer communications between browsers with all the rest of the =
communication modes, such as calling automated services or PSTN being =
insignificant. I simply do not agree to this point of view. I expect =
that communication with automated services, such as video greeting cards =
or voice blogging, would be a significant portion of WebRTC user base. =
If such automated service is deployed as a plain HTTP web site, it =
should be able to communicate with web browsers using RTP. SRTP in such =
case would serve no purpose.
>=20
> Finally, requiring secure communications for everything is going =
against the way most of the web works. Most of it is not secured and =
only requires secure communications when secure (HTTPS) web site is =
accessed. I think it should be the same for WebRTC, with DTLS-SRTP =
required when connected to HTTPS web site and plain RTP allowed when =
connected to plan HTTP.
> _____________
> Roman Shpount
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From roman@telurix.com  Wed May  2 15:16:19 2012
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Date: Wed, 2 May 2012 18:16:14 -0400
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To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] Final plea about SRTP
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Cullen,

I do not think bid down is an issue. It should be clearly specified when
the peer connection is created that it is SRTP or RTP and never should one
become another. And if key exchange mechanism selection is not enforced, I
would consider bid down from DTLS to SDES SRTP just as much of an issue.

Numerous reasons for RTP support have been mentioned on this list (legacy
interop, easier to implement, easier to debug, quicker to setup, can be
used from a location from which encrypted communications are not allowed)
but as far as I know none of them have been considered compelling by the
people on this list. As everyone have mentioned before, requiring SRTP does
not really break anything, but, from my point of view, it just makes
implementation of some services more difficult for no apparent reason. I
simply see this as a general design flaw introduced based on FUD.
_____________
Roman Shpount


On Wed, May 2, 2012 at 5:32 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Roman,
>
> One comment on this - I think people understand there could be services
> with no security requirements that could run over RTP, and HTTP, with no
> identity. But we need to have a secure solution for some other services.
> The questions is once you have a secure solution, what is the incentive to
> also support an insecure solution - so far no one has come up with a super
> compelling story about dealing with the bid down and I suspect that lots of
> people did not view the overhead of running the secure version as all that
> high. I suspect that is part of why the decisions went the way it did -
> basically people agreed we needed a secure solution, and when they
> considered also having an insecure solution, they saw lots of complications
> of doing both and not much gain in the insecure solution over the secure
> solution.
>
> Cullen
>
>
>
> On May 2, 2012, at 10:03 AM, Roman Shpount wrote:
>
> > I know there was a consensus call on this list that SRTP shall be used
> for all the calls in WebRTC, but I still do not understand the
> justification for this requirement for WebRTC applications delivered over
> HTTP with no identity. For such scenarios SRTP (even DTLS-SRTP) serves
> almost no purpose. If application is delivered over HTTP attacker can spoof
> the entire web site. It is trivial if the attacker is on the communications
> path. If attacker is seating in the airport using the same network, it can
> put itself on the communications path using arp cache poisoning. Once the
> web site is spoofed, any type of man in the middle attack can be
> implemented. If DTLS-SRTP is used user can detect the attack by checking
> the key signature, but in reality very few people will do this.
> >
> > The main argument to require SRTP everywhere was that it does not break
> anything. But neither would naming all the API methods in High Elfish.
> Either requirement does not break things, but make working with WebRTC
> harder then it should. At the same time both of those requirements are
> completely unjustified.
> >
> > Furthermore, assumption on this list that most of the WebRTC use would
> be peer-to-peer communications between browsers with all the rest of the
> communication modes, such as calling automated services or PSTN being
> insignificant. I simply do not agree to this point of view. I expect that
> communication with automated services, such as video greeting cards or
> voice blogging, would be a significant portion of WebRTC user base. If such
> automated service is deployed as a plain HTTP web site, it should be able
> to communicate with web browsers using RTP. SRTP in such case would serve
> no purpose.
> >
> > Finally, requiring secure communications for everything is going against
> the way most of the web works. Most of it is not secured and only requires
> secure communications when secure (HTTPS) web site is accessed. I think it
> should be the same for WebRTC, with DTLS-SRTP required when connected to
> HTTPS web site and plain RTP allowed when connected to plan HTTP.
> > _____________
> > Roman Shpount
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001517592bfc1890f704bf1509ac
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

Cullen,<br><br>I do not think bid down is an issue. It should be clearly sp=
ecified when the peer connection is created that it is SRTP or RTP and neve=
r should one become another. And if key exchange mechanism selection is not=
 enforced, I would consider bid down from DTLS to SDES SRTP just as much of=
 an issue.<br>
<br>Numerous reasons for RTP support have been mentioned on this list (lega=
cy interop, easier to implement, easier to debug, quicker to setup, can be =
used from a location from which encrypted communications are not allowed) b=
ut as far as I know none of them have been considered compelling by the peo=
ple on this list. As everyone have mentioned before, requiring SRTP does no=
t really break anything, but, from my point of view, it just makes implemen=
tation of some services more difficult for no apparent reason. I simply see=
 this as a general design flaw introduced based on FUD.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Wed, May 2=
, 2012 at 5:32 PM, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:=
fluffy@iii.ca" target=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><bl=
ockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #=
ccc solid;padding-left:1ex">
<br>
Roman,<br>
<br>
One comment on this - I think people understand there could be services wit=
h no security requirements that could run over RTP, and HTTP, with no ident=
ity. But we need to have a secure solution for some other services. The que=
stions is once you have a secure solution, what is the incentive to also su=
pport an insecure solution - so far no one has come up with a super compell=
ing story about dealing with the bid down and I suspect that lots of people=
 did not view the overhead of running the secure version as all that high. =
I suspect that is part of why the decisions went the way it did - basically=
 people agreed we needed a secure solution, and when they considered also h=
aving an insecure solution, they saw lots of complications of doing both an=
d not much gain in the insecure solution over the secure solution.<br>

<br>
Cullen<br>
<div><div class=3D"h5"><br>
<br>
<br>
On May 2, 2012, at 10:03 AM, Roman Shpount wrote:<br>
<br>
&gt; I know there was a consensus call on this list that SRTP shall be used=
 for all the calls in WebRTC, but I still do not understand the justificati=
on for this requirement for WebRTC applications delivered over HTTP with no=
 identity. For such scenarios SRTP (even DTLS-SRTP) serves almost no purpos=
e. If application is delivered over HTTP attacker can spoof the entire web =
site. It is trivial if the attacker is on the communications path. If attac=
ker is seating in the airport using the same network, it can put itself on =
the communications path using arp cache poisoning. Once the web site is spo=
ofed, any type of man in the middle attack can be implemented. If DTLS-SRTP=
 is used user can detect the attack by checking the key signature, but in r=
eality very few people will do this.<br>

&gt;<br>
&gt; The main argument to require SRTP everywhere was that it does not brea=
k anything. But neither would naming all the API methods in High Elfish. Ei=
ther requirement does not break things, but make working with WebRTC harder=
 then it should. At the same time both of those requirements are completely=
 unjustified.<br>

&gt;<br>
&gt; Furthermore, assumption on this list that most of the WebRTC use would=
 be peer-to-peer communications between browsers with all the rest of the c=
ommunication modes, such as calling automated services or PSTN being insign=
ificant. I simply do not agree to this point of view. I expect that communi=
cation with automated services, such as video greeting cards or voice blogg=
ing, would be a significant portion of WebRTC user base. If such automated =
service is deployed as a plain HTTP web site, it should be able to communic=
ate with web browsers using RTP. SRTP in such case would serve no purpose.<=
br>

&gt;<br>
&gt; Finally, requiring secure communications for everything is going again=
st the way most of the web works. Most of it is not secured and only requir=
es secure communications when secure (HTTPS) web site is accessed. I think =
it should be the same for WebRTC, with DTLS-SRTP required when connected to=
 HTTPS web site and plain RTP allowed when connected to plan HTTP.<br>

&gt; _____________<br>
&gt; Roman Shpount<br>
</div></div>&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>
</blockquote></div><br></div>

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From: "Dan Wing" <dwing@cisco.com>
To: "'Mary Barnes'" <mary.ietf.barnes@gmail.com>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>	<E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com>	<BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl>	<387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com>	<2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com>	<387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com>	<E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com>	<4F9EC0B2.10903@alcatel-lucent.com>	<101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net>	<CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com>	<E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com>	<4FA0F43E.4020308@ericsson.com>	<E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com>	<013101cd288c$09328250$1b9786f0$@com> <CAHBDyN51L=1Khiy1Kv45F_i9RD8kwWi4eNfcQD2bT6_9y2=GeQ@mail.gmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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> -----Original Message-----
> From: Mary Barnes [mailto:mary.ietf.barnes@gmail.com]
> Sent: Wednesday, May 02, 2012 1:27 PM
> To: Dan Wing
> Cc: Jim Barnett; Stefan Hakansson LK; rtcweb@ietf.org
> Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE:
> Use Case draft]
> 
> I agree with you in general, however, the link to your slides seems to
> be broken.

http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf

-d

> 
> Mary.
> 
> 
> On Wed, May 2, 2012 at 12:50 PM, Dan Wing <dwing@cisco.com> wrote:
> 
> 
> 	> -----Original Message-----
> 	> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org]
> On
> 	> Behalf Of Jim Barnett
> 	> Sent: Wednesday, May 02, 2012 7:39 AM
> 	> To: Stefan Hakansson LK; WEBRTC@ietf.org
> 	> Subject: Re: [WEBRTC] Use Case draft
> 	>
> 	> When I say that this use case may not add further requirements,
> I mean
> 	> that it looks like it would be possible to implement it given
> the
> 	> current definitions of the protocols.  However, the current use
> cases
> 	> are all written in terms of "the browser", which is not further
> 	> defined.
> 	> But if "browser" means Mozilla, Chrome, etc., then I think it
> is
> 	> important to add a use case in which one of the end points is
> not a
> 	> browser, but an enterprise gateway (which will route the call
> to an
> 	> employee of its choice, and may record the call, etc.) It is
> important
> 	> to note that this is not a peer-to-peer use case; the gateway
> is not
> 	> the
> 	> caller's peer.  The employee that the caller ends up talking to
> may be
> 	> considered a peer, but the webRTC call does not extend all the
> way to
> 	> that employee - it stops at the gateway.
> 	>
> 	> This is a very different use case from any in the current
> document.
> 	> That's why it's important to add it, even though (as far as I
> can tell)
> 	> it doesn't require us to change any of the work we've done.
> 
> 	Somewhere, we need consensus on a model for interworking WEBRTC
> 	endpoints with non-WEBRTC endpoints.
> 
> 	The decision comes down to this:
> 
> 	 1. encumber WEBRTC endpoints with the interworking
> 	    effort, or
> 	 2. encumber a separate interworking device with the
> 	    interworking effort.
> 
> 	I believe we have a better chance of success with (2), where
> 	possible to do (2).
> 
> 	For some decisions, such as Consent Freshness (previously called
> Voice
> 	Hammer Attack in http://tools.ietf.org/html/rfc5245#section-
> 18.5.1),
> 	non-WEBRTC endpoints need to respond to those ICE connectivity
> 	checks or have a gateway in front of them that responds to those
> 	connectivity checks on their behalf.  This means that WEBRTC
> 	cannot work directly with some existing SIP equipment (because
> 	a lot of SIP equipment does not support ICE).
> 
> 	For other decisions, such as if we disallow un-encrypted RTP by
> 	WEBRTC endpoints, we create a requirement that some device does
> 	the interworking between WEBRTC endpoints (which do only SRTP)
> 	and non-WEBRTC endpoints (which do RTP).  That means, for that
> 	interworking, we would adopt the interworking model on slide 7
> 	that I presented at IETF83,
> 	http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
> 
> 	However, when I presented slide 7, there were objections at the
> 	microphone that this model 'is broken'.  I would like to
> understand
> 	the objections so we can reach consensus on how interworking from
> 	WEBRTC to non-WEBRTC is expected to occur.
> 
> 	-d
> 
> 
> 	> - Jim
> 	> -----Original Message-----
> 	> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-bounces@ietf.org]
> On
> 	> Behalf
> 	> Of Stefan Hakansson LK
> 	> Sent: Wednesday, May 02, 2012 4:46 AM
> 	> To: WEBRTC@ietf.org
> 	> Subject: Re: [WEBRTC] Use Case draft
> 	>
> 	> On 05/01/2012 02:05 PM, Jim Barnett wrote:
> 	> > One way to describe the use case is to let the contact
> center's media
> 	> > server/gateway serve as the webRTC endpoint.  Then all the
> issues of
> 	> > call delivery, call monitoring, etc. disappear.  They are
> handled by
> 	> > application software that sits behind the webRTC endpoint.
> The
> 	> > company I work for makes a good living selling software that
> deals
> 	> > with all these issues - including bathroom breaks - and
> that's how we
> 	> > would tend to think of this case.  To us, it's a new kind of
> 	> > call/connection coming into the contact center, which we
> translate
> 	> > into SIP at the border and then handle normally.
> 	> >
> 	> > It's not clear to me if this use case adds any extra
> requirements.
> 	>
> 	> I think this is important to sort out. If the use case does not
> add any
> 	> extra requirements, what's the point of adding it?
> 	>
> 	> > We would just have to be careful not to assume that a webRTC
> endpoint
> 	> > is always a person/browser-based user agent.  It may seem a
> bit
> 	> > unsettling that the webRTC endpoint can distribute the call
> somewhere
> 	> > else and let others listen in, but as far as I can tell that
> is
> 	> > already the case.  If Bob calls Alice with full
> authentication and
> 	> > security, he can be sure that he is connected to Alice's user
> agent
> 	> > and that no one in between can listen in, but there's nothing
> 	> stopping
> 	>
> 	> > Alice from recording the audio, or forwarding it to a third
> party.
> 	> So
> 	>
> 	> > Bob could in fact be talking to Mary if that's how Alice
> wants to
> 	> > arrange things (_behind_ her user agent).  In general, Bob is
> assured
> 	> > only that he is talking to someone Alice wants him to talk
> to, and
> 	> > that no one can snoop without Alice's permission.  That's
> very much
> 	> > the way things work with the call center - you are sure that
> you are
> 	> > 1) connected securely to your bank 2) talking to someone that
> the
> 	> bank
> 	>
> 	> > wants you to talk to 3) being recorded or snooped on only
> when the
> 	> > bank explicitly chooses to do so.
> 	> >
> 	> > - Jim
> 	> >
> 	> > -----Original Message----- From: WEBRTC-bounces@ietf.org
> 	> > [mailto:WEBRTC-bounces@ietf.org] On Behalf Of Marshall
> Eubanks Sent:
> 	> > Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
> 	> > WEBRTC@ietf.org Subject: Re: [WEBRTC] Use Case draft
> 	> >
> 	> > On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
> 	> > Andrew<andrew.hutton@siemens-enterprise.com>  wrote:
> 	> >> Whether anybody has been successful in the past with this
> type of
> 	> use
> 	>
> 	> >> case is I think irrelevant.
> 	> >>
> 	> >>
> 	> >>
> 	> >> The enterprise call centre use case is I think a vital use
> case
> 	> >> because it is a scenario in which one user is only concerned
> that
> 	> >> they can securely reach an organization/domain and is not
> concerned
> 	> >> about the individual within that domain  that they
> communicate with.
> 	>
> 	> >> A suspect quite a large percentage of WEBRTC applications
> will be
> 	> >> like this and it is not covered in the current use case
> draft.
> 	> >
> 	> > I agree that this is a very useful use case and one I think
> is going
> 	> > to get a lot of traction. There is a very solid business case
> for
> 	> > this.  However, I have a fair amount of experience with a
> video call
> 	> > center for a client, and it is not as simple as it might
> seem.
> 	> >
> 	> > The essence of course is that you get the next available
> person,
> 	> i.e.,
> 	>
> 	> > it is anycast. Determining who the next available person is
> is not
> 	> > trivial, nor is error recovery. (If I call you, and you don't
> answer
> 	> > or the call drops or whatever,  I can leave a message or try
> later.
> 	> If
> 	>
> 	> > I call a help desk, and this happens, I want a new agent,
> ideally
> 	> > automatically.) Call forwarding (e.g., first tier to second
> tier
> 	> > technical support) is essential, and it may be anycast or
> directed.
> 	> > There are also some security oddities  - if I am connecting
> from
> 	> home,
> 	>
> 	> > I may need to authenticate, use a credit card, etc. If I am
> 	> connecting
> 	>
> 	> > from inside a store, and providing in store video technical
> support
> 	> is
> 	>
> 	> > big part of the market, then the store authenticates me off
> line and
> 	> > the call really should just be a button push, which implies
> that the
> 	> > store has previously authenticated some sort of master
> session. In
> 	> > addition, unlike most video calls, in the enterprise call
> center a
> 	> > supervisor may need to be able to monitor (i.e., watch) a
> call, and
> 	> in
> 	>
> 	> > some circumstances (financial or medical calls, for example)
> there
> 	> > will need to be third party recording. I believe that  these
> details
> 	> > would be different from the typical WEBRTC scenario.
> 	> >
> 	> > Also, there will be a temptation to do the anycasting by the
> 	> > techniques used to load balance servers in a data center, but
> I think
> 	> > that may not be sufficient. The call "center" may in fact be
> spread
> 	> > completely across the planet (daytime support in the US,
> nighttime
> 	> > support in India, for example) and be on multiple autonomous
> systems
> 	> > (and even from people's homes), which gives rise to some of
> the
> 	> > transport issues NVO3 may face, but without any opportunity
> for
> 	> packet
> 	>
> 	> > tagging. Plus, there will complicated rules about who can be
> selected
> 	> > next. WEBRTC shouldn't worry about the intricacies of
> bathroom break
> 	> > policies; these complexities should be dealt with by an
> 	> > enterprise-side database, which to me (together with some of
> the
> 	> other
> 	>
> 	> > issues above) suggests that this would probably benefit from
> API
> 	> > support.
> 	> >
> 	> > Regards Marshall
> 	> >
> 	> >
> 	> >>
> 	> >>
> 	> >>
> 	> >> So I think we need it.
> 	> >>
> 	> >>
> 	> >>
> 	> >> Regards
> 	> >>
> 	> >> Andy
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >> From: WEBRTC-bounces@ietf.org [mailto:WEBRTC-
> bounces@ietf.org] On
> 	> >> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
> 	> >> WEBRTC@ietf.org
> 	> >>
> 	> >>
> 	> >> Subject: Re: [WEBRTC] Use Case draft
> 	> >>
> 	> >>
> 	> >>
> 	> >> Without numbers it is impossible to argue, but, if we talk
> about the
> 	> >> perceived need, I disagree.  Think of the people who travel
> abroad
> 	> >> and cannot call the 800 number. (I routinely use Web
> interface for
> 	> >> calls when traveling.)
> 	> >>
> 	> >>
> 	> >>
> 	> >> I am all for  the use case, as described by Jim.
> 	> >>
> 	> >> Igor
> 	> >>
> 	> >> On 4/30/2012 9:54 AM, Tim Panton wrote:
> 	> >>
> 	> >> ...
> 	> >>
> 	> >> I can't tell you the actual numbers, but when presented with
> the
> 	> >> choice of calling a toll free number
> 	> >>
> 	> >> or clicking a button marked "free internet call" - almost
> no-one on
> 	> a
> 	>
> 	> >> real, busy site clicked the button.
> 	> >>
> 	> >> ( for every button click there were several orders of
> magnitude more
> 	> >> 0800 calls from that page).
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >> So from my perspective this is a legacy interop use case
> with almost
> 	> >> zero user acceptance.
> 	> >>
> 	> >>
> 	> >>
> 	> >> (as far as I can see no-one has made this use-case desirable
> in
> 	> >> practice yet.)
> 	> >>
> 	> >> Tim.
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >>
> 	> >> _______________________________________________
> 	> >>
> 	> >> WEBRTC mailing list
> 	> >>
> 	> >> WEBRTC@ietf.org
> 	> >>
> 	> >> https://www.ietf.org/mailman/listinfo/WEBRTC
> 	> >>
> 	> >>
> 	> >> _______________________________________________ WEBRTC
> mailing list
> 	> >> WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> 	> >>
> 	> > _______________________________________________ WEBRTC
> mailing list
> 	> > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> 	> > _______________________________________________ WEBRTC
> mailing list
> 	> > WEBRTC@ietf.org https://www.ietf.org/mailman/listinfo/WEBRTC
> 	>
> 	> _______________________________________________
> 	> WEBRTC mailing list
> 	> WEBRTC@ietf.org
> 	> https://www.ietf.org/mailman/listinfo/WEBRTC
> 	> _______________________________________________
> 	> WEBRTC mailing list
> 	> WEBRTC@ietf.org
> 	> https://www.ietf.org/mailman/listinfo/WEBRTC
> 
> 	_______________________________________________
> 	rtcweb mailing list
> 	rtcweb@ietf.org
> 	https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 



From bernard_aboba@hotmail.com  Wed May  2 17:41:44 2012
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At IETF 83,  Dan showed some slides relating to SRTP/RTP gateways that
seemed to cover the "extreme legacy" case quite well. 

In those slides, the "RTCWEB" components were still talking SRTP, so they
could be said to be compliant with a "mandatory to implement, mandatory to
use" posture.   Yet, the circa 2005 IP phone still got to receive and send
RTP.  

Since the "extreme legacy" cases are solvable via gateways, and the
performance impact and cost on virtually any new hardware (PC/tablet/mobile)
is minimal, it seems to me that making SRTP mandatory to implement and use
has little downside.

A few years ago, the thought of turning on SRTP by default was a bit scary
(mostly because of potential interop issues, not cost).  However, today
turning it on by default "just works" with minimal performance impact or
other hassles (other than occasional interop gremlins).  By the time RTCWEB
is widely deployed any argument against SRTP will probably be vestigial.

Given this, it seems to me that the "right thing" is for SRTP to be
mandatory to implement and use, especially if SDES is available, so the
likelihood of interoperability will be high.   

On Wed, May 2, 2012 at 5:32 PM, Cullen Jennings <fluffy@iii.ca> wrote:


Roman,

One comment on this - I think people understand there could be services with
no security requirements that could run over RTP, and HTTP, with no
identity. But we need to have a secure solution for some other services. The
questions is once you have a secure solution, what is the incentive to also
support an insecure solution - so far no one has come up with a super
compelling story about dealing with the bid down and I suspect that lots of
people did not view the overhead of running the secure version as all that
high. I suspect that is part of why the decisions went the way it did -
basically people agreed we needed a secure solution, and when they
considered also having an insecure solution, they saw lots of complications
of doing both and not much gain in the insecure solution over the secure
solution.

Cullen




On May 2, 2012, at 10:03 AM, Roman Shpount wrote:

> I know there was a consensus call on this list that SRTP shall be used for
all the calls in WebRTC, but I still do not understand the justification for
this requirement for WebRTC applications delivered over HTTP with no
identity. For such scenarios SRTP (even DTLS-SRTP) serves almost no purpose.
If application is delivered over HTTP attacker can spoof the entire web
site. It is trivial if the attacker is on the communications path. If
attacker is seating in the airport using the same network, it can put itself
on the communications path using arp cache poisoning. Once the web site is
spoofed, any type of man in the middle attack can be implemented. If
DTLS-SRTP is used user can detect the attack by checking the key signature,
but in reality very few people will do this.
>
> The main argument to require SRTP everywhere was that it does not break
anything. But neither would naming all the API methods in High Elfish.
Either requirement does not break things, but make working with WebRTC
harder then it should. At the same time both of those requirements are
completely unjustified.
>
> Furthermore, assumption on this list that most of the WebRTC use would be
peer-to-peer communications between browsers with all the rest of the
communication modes, such as calling automated services or PSTN being
insignificant. I simply do not agree to this point of view. I expect that
communication with automated services, such as video greeting cards or voice
blogging, would be a significant portion of WebRTC user base. If such
automated service is deployed as a plain HTTP web site, it should be able to
communicate with web browsers using RTP. SRTP in such case would serve no
purpose.
>
> Finally, requiring secure communications for everything is going against
the way most of the web works. Most of it is not secured and only requires
secure communications when secure (HTTPS) web site is accessed. I think it
should be the same for WebRTC, with DTLS-SRTP required when connected to
HTTPS web site and plain RTP allowed when connected to plan HTTP.
> _____________
> Roman Shpount

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

 


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that seemed to cover the &#8220;extreme legacy&#8221; case quite well. =
<o:p></o:p></span></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:#1F497D=
'>In those slides, the &#8220;RTCWEB&#8221; components were still =
talking SRTP, so they could be said to be compliant with a =
&#8220;mandatory to implement, mandatory to use&#8221; =
posture.&nbsp;&nbsp; Yet, the circa 2005 IP phone still got to receive =
and send RTP. &nbsp;<o:p></o:p></span></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:#1F497D=
'>Since the &#8220;extreme legacy&#8221; cases are solvable via =
gateways, and the performance impact and cost on virtually any new =
hardware (PC/tablet/mobile) is minimal, it seems to me that making SRTP =
mandatory to implement and use has little =
downside.<o:p></o:p></span></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:#1F497D=
'>A few years ago, the thought of turning on SRTP by default was a bit =
scary (mostly because of potential interop issues, not cost).&nbsp; =
However, today turning it on by default &#8220;just works&#8221; with =
minimal performance impact or other hassles (other than occasional =
interop gremlins).&nbsp; By the time RTCWEB is widely deployed any =
argument against SRTP will probably be =
vestigial.<o:p></o:p></span></b></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:#1F497D=
'>Given this, it seems to me that the &#8220;right thing&#8221; is for =
SRTP to be mandatory to implement and use, especially if SDES is =
available, so the likelihood of interoperability will be high. =
&nbsp;&nbsp;</span></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p></o:p></span></p><div><p class=3DMsoNormal>On Wed, May 2, 2012 =
at 5:32 PM, Cullen Jennings &lt;<a href=3D"mailto:fluffy@iii.ca" =
target=3D"_blank">fluffy@iii.ca</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal><br>Roman,<br><br>One comment on this - I think people =
understand there could be services with no security requirements that =
could run over RTP, and HTTP, with no identity. But we need to have a =
secure solution for some other services. The questions is once you have =
a secure solution, what is the incentive to also support an insecure =
solution - so far no one has come up with a super compelling story about =
dealing with the bid down and I suspect that lots of people did not view =
the overhead of running the secure version as all that high. I suspect =
that is part of why the decisions went the way it did - basically people =
agreed we needed a secure solution, and when they considered also having =
an insecure solution, they saw lots of complications of doing both and =
not much gain in the insecure solution over the secure =
solution.<br><br>Cullen<o:p></o:p></p><div><div><p =
class=3DMsoNormal><br><br><br>On May 2, 2012, at 10:03 AM, Roman Shpount =
wrote:<br><br>&gt; I know there was a consensus call on this list that =
SRTP shall be used for all the calls in WebRTC, but I still do not =
understand the justification for this requirement for WebRTC =
applications delivered over HTTP with no identity. For such scenarios =
SRTP (even DTLS-SRTP) serves almost no purpose. If application is =
delivered over HTTP attacker can spoof the entire web site. It is =
trivial if the attacker is on the communications path. If attacker is =
seating in the airport using the same network, it can put itself on the =
communications path using arp cache poisoning. Once the web site is =
spoofed, any type of man in the middle attack can be implemented. If =
DTLS-SRTP is used user can detect the attack by checking the key =
signature, but in reality very few people will do this.<br>&gt;<br>&gt; =
The main argument to require SRTP everywhere was that it does not break =
anything. But neither would naming all the API methods in High Elfish. =
Either requirement does not break things, but make working with WebRTC =
harder then it should. At the same time both of those requirements are =
completely unjustified.<br>&gt;<br>&gt; Furthermore, assumption on this =
list that most of the WebRTC use would be peer-to-peer communications =
between browsers with all the rest of the communication modes, such as =
calling automated services or PSTN being insignificant. I simply do not =
agree to this point of view. I expect that communication with automated =
services, such as video greeting cards or voice blogging, would be a =
significant portion of WebRTC user base. If such automated service is =
deployed as a plain HTTP web site, it should be able to communicate with =
web browsers using RTP. SRTP in such case would serve no =
purpose.<br>&gt;<br>&gt; Finally, requiring secure communications for =
everything is going against the way most of the web works. Most of it is =
not secured and only requires secure communications when secure (HTTPS) =
web site is accessed. I think it should be the same for WebRTC, with =
DTLS-SRTP required when connected to HTTPS web site and plain RTP =
allowed when connected to plan HTTP.<br>&gt; _____________<br>&gt; Roman =
Shpount<o:p></o:p></p></div></div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>&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"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></=
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class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></div></body></html>
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "igor.faynberg@alcatel-lucent.com" <igor.faynberg@alcatel-lucent.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft
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Date: Thu, 3 May 2012 05:05:52 +0000
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Subject: Re: [rtcweb] Use Case draft
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Igor,

When you call Company X, you need to assure that you are talking with Compa=
ny X agent and not required to distinguish the agent.

The identity could be just X (no userpart) or anonymous. There is no differ=
ence between anonymous and agent007@X as there is no means to route directl=
y agent007@x in the call centre scenario.

Thanks
Partha



>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Igor Faynberg
>Sent: Wednesday, May 02, 2012 9:39 PM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Use Case draft
>
>I am afraid I don't understand this.
>
>For the call center case, when I "call" company X, I would like to make
>sure that I am connected to an agent representing X, and so the identity
>should looke like something "X_Agent_707," not "Anonymous."
>
>Igor
>
>On 5/2/2012 5:05 AM, Ravindran, Parthasarathi wrote:
>> Stefan,
>>
>> This usecase add its own requirements from identity perspective as I
>> mentioned earlier.
>>
>> WebRTC MUST allow "Anonymous" users secure session for call center
>> usecase. "Anonymous" User may be agent in the call center side or
>> customer who does not require Identity to start the session.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Stefan Hakansson LK
>>> Sent: Wednesday, May 02, 2012 2:16 PM
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Use Case draft
>>>
>>> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>>>> One way to describe the use case is to let the contact center's
>>>> media server/gateway serve as the webRTC endpoint.  Then all the
>>>> issues of call delivery, call monitoring, etc. disappear.  They are
>>>> handled by application software that sits behind the webRTC
>>>> endpoint.  The company I work for makes a good living selling
>>>> software that deals with all these issues - including bathroom
>>>> breaks - and that's how we would tend to think of this case.  To us,
>>>> it's a new kind of call/connection coming into the contact center,
>>>> which we translate into SIP at the border and then handle normally.
>>>>
>>>> It's not clear to me if this use case adds any extra requirements.
>>> I think this is important to sort out. If the use case does not add
>>> any extra requirements, what's the point of adding it?
>>>
>>>> We would just have to be careful not to assume that a webRTC
>>>> endpoint is always a person/browser-based user agent.  It may seem a
>>>> bit unsettling that the webRTC endpoint can distribute the call
>>>> somewhere else and let others listen in, but as far as I can tell
>>>> that is already the case.  If Bob calls Alice with full
>>>> authentication and security, he can be sure that he is connected to
>>>> Alice's user agent and that no one in between can listen in, but
>>>> there's nothing stopping Alice from recording the audio, or
>>>> forwarding it to a third party.  So Bob could in fact be talking to
>>>> Mary if that's how Alice wants to arrange things (_behind_ her user
>>>> agent).  In general, Bob is assured only that he is talking to
>>>> someone Alice wants him to talk to, and that no one can snoop
>>>> without Alice's permission.  That's very much the way things work
>>>> with the call center - you are sure that you are
>>>> 1) connected securely to your bank 2) talking to someone that the
>>>> bank wants you to talk to 3) being recorded or snooped on only when
>>>> the bank explicitly chooses to do so.
>>>>
>>>> - Jim
>>>>
>>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>>>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>>>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>>>
>>>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>>>> Andrew<andrew.hutton@siemens-enterprise.com>   wrote:
>>>>> Whether anybody has been successful in the past with this type of
>>>>> use case is I think irrelevant.
>>>>>
>>>>>
>>>>>
>>>>> The enterprise call centre use case is I think a vital use case
>>>>> because it is a scenario in which one user is only concerned that
>>>>> they can securely reach an organization/domain and is not concerned
>>>>> about the individual within that domain  that they communicate
>with.
>>>>> A suspect quite a large percentage of RTCWEB applications will be
>>>>> like this and it is not covered in the current use case draft.
>>>> I agree that this is a very useful use case and one I think is going
>>>> to get a lot of traction. There is a very solid business case for
>>>> this.  However, I have a fair amount of experience with a video call
>>>> center for a client, and it is not as simple as it might seem.
>>>>
>>>> The essence of course is that you get the next available person,
>>>> i.e., it is anycast. Determining who the next available person is is
>>>> not trivial, nor is error recovery. (If I call you, and you don't
>>>> answer or the call drops or whatever,  I can leave a message or try
>>>> later. If I call a help desk, and this happens, I want a new agent,
>>>> ideally
>>>> automatically.) Call forwarding (e.g., first tier to second tier
>>>> technical support) is essential, and it may be anycast or directed.
>>>> There are also some security oddities  - if I am connecting from
>>>> home, I may need to authenticate, use a credit card, etc. If I am
>>>> connecting from inside a store, and providing in store video
>>>> technical support is big part of the market, then the store
>>>> authenticates me off line and the call really should just be a
>>>> button push, which implies that the store has previously
>>>> authenticated some sort of master session. In addition, unlike most
>>>> video calls, in the enterprise call center a supervisor may need to
>>>> be able to monitor (i.e., watch) a call, and in some circumstances
>>>> (financial or medical calls, for example) there will need to be
>>>> third party recording. I believe that  these details would be
>different from the typical RTCWEB scenario.
>>>>
>>>> Also, there will be a temptation to do the anycasting by the
>>>> techniques used to load balance servers in a data center, but I
>>>> think that may not be sufficient. The call "center" may in fact be
>>>> spread completely across the planet (daytime support in the US,
>>>> nighttime support in India, for example) and be on multiple
>>>> autonomous systems (and even from people's homes), which gives rise
>>>> to some of the transport issues NVO3 may face, but without any
>>>> opportunity for packet tagging. Plus, there will complicated rules
>>>> about who can be selected next. RTCWEB shouldn't worry about the
>>>> intricacies of bathroom break policies; these complexities should be
>>>> dealt with by an enterprise-side database, which to me (together
>>>> with some of the other issues above) suggests that this would
>>>> probably benefit from API support.
>>>>
>>>> Regards Marshall
>>>>
>>>>
>>>>>
>>>>>
>>>>> So I think we need it.
>>>>>
>>>>>
>>>>>
>>>>> Regards
>>>>>
>>>>> Andy
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>>>> rtcweb@ietf.org
>>>>>
>>>>>
>>>>> Subject: Re: [rtcweb] Use Case draft
>>>>>
>>>>>
>>>>>
>>>>> Without numbers it is impossible to argue, but, if we talk about
>>>>> the perceived need, I disagree.  Think of the people who travel
>>>>> abroad and cannot call the 800 number. (I routinely use Web
>>>>> interface for calls when traveling.)
>>>>>
>>>>>
>>>>>
>>>>> I am all for  the use case, as described by Jim.
>>>>>
>>>>> Igor
>>>>>
>>>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>>>
>>>>> ...
>>>>>
>>>>> I can't tell you the actual numbers, but when presented with the
>>>>> choice of calling a toll free number
>>>>>
>>>>> or clicking a button marked "free internet call" - almost no-one on
>>>>> a real, busy site clicked the button.
>>>>>
>>>>> ( for every button click there were several orders of magnitude
>>>>> more
>>>>> 0800 calls from that page).
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> So from my perspective this is a legacy interop use case with
>>>>> almost zero user acceptance.
>>>>>
>>>>>
>>>>>
>>>>> (as far as I can see no-one has made this use-case desirable in
>>>>> practice yet.)
>>>>>
>>>>> Tim.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>>
>>>>> 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
>>>> _______________________________________________ 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
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From pravindran@sonusnet.com  Wed May  2 22:23:27 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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Fabio,

Could you please explain how to differentiate through protocol means that t=
he peer is site (gateway) or endpoint (webbrowser).

Thanks
Partha=20

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Fabio Pietrosanti (naif)
>Sent: Thursday, May 03, 2012 2:24 AM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE:
>Use Case draft]
>
>On 5/2/12 7:50 PM, Dan Wing wrote:
>http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf
>> However, when I presented slide 7, there were objections at the
>> microphone that this model 'is broken'.  I would like to understand
>> the objections so we can reach consensus on how interworking from
>> WEBRTC to non-WEBRTC is expected to occur.
>
>IMHO it would be much easier, as described in tons of previous email, to
>use different Key Management for different requirements:
>
>- SDES + SRTP for end-to-site (peer to gateway)
>- DTLS-SRTP for end-to-end (peer to peer)
>
>It would be a simpler approach for:
>
>- Security (the user know the security level)
>- Interoperability (Use standard protocols for the need to interoperate)
>
>That way the "simpler, legacy, standard" technology would be used for
>all voip "server applications" while the new burning (yet not used by
>anyone) DTLS-SRTP will be used for peer-to-peer calls.
>
>Please DO NOT reinvent the wheel for what's already existing and
>deployed.
>
>Doing so, it's like going against the natural human behavior, it would
>just represent a failure.
>
>Fabio
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From christer.holmberg@ericsson.com  Wed May  2 23:02:49 2012
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Date: Thu, 3 May 2012 08:02:44 +0200
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>> I think that is unlikely to happen, as forking is normally done in a ser=
ial manner (to avoid other issues associated with parallel forking).
>
> I am not sure why you would say that most forking is serial. It is quite =
often parallel. What is commonly done to avoid support for multiple=20
> parallel media streams is playing media from the newest received RTP stre=
am. In either case there is no guarantee that you picked the=20
> winning stream, and you always need to make sure that you use the SDP rec=
eived in the dialog that got 200 OK and that media playback=20
> switches to RTP stream that remains. Making this work is often problemati=
c in SIP since there is no way to associate received RTP with=20
> received SDP information due to RTP being sent from a different IP/port f=
rom the one used in SDP. It should be easier with WebRTC=20
> since ICE support is required and remote party IP/port can be determined =
for each flow.
>
> Finally, when you are dealing with forking you need to keep track of the =
remote IP/ports and ICE candidates for each dialog. You can=20
> play media from only one of them, but this remote connection state is req=
uired to complete the call setup when 200 OK is received.

You are absolutely right. I still question whether we need to support paral=
lel forking. If you say it occurs often, I can't argue against you, eventho=
ugh my own experience is different :)

(Another option, if we want to support parallel forking, would of course be=
 to create a completely new PeerConnection, with a *new* local IP address:p=
ort. But, you would of course then have to provide that information to the =
remote peer.)

I have no strong feeling on whether we want to do cloning, but do people ag=
ree that, for a given PeerConnection, we only need to support a single remo=
te peer (which can be modified, though)?

Regards,

Christer



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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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On 05/02/2012 08:19 PM, Marshall Eubanks wrote:
> My objection is that the proposed system will require middleware to
> interoperate with the
> vast number of videoconferencing sessions out there, most of which use
> RTP. From the
> standpoint of a video service provider, buying hardware to support
> video to laptops is likely to
> lead to requests that participants download some other software which
> interoperates natively.
>
> This is an existing business with a fairly large scale and installed
> base. Not operating the way that they do is not likely to go over
> well.
>
Marshall,

I'd like to draw your attention to two numbers:

- Number of installed room videoconferencing units: On the order of 1 
million.

http://www.polycom.com/global/documents/company/video_conferencing_by_the_numbers.pdf

- Number of installed Chrome browsers: On the order of 200 million.

http://www.techradar.com/news/internet/google-chrome-browser-hits-200-million-users-1033951

(pulling out Chrome just because I know we've promised to ship this 
stuff. And we auto-update, which means most of the users WILL be running 
a WebRTC-compatible browser the week after we release it.)

I argue strongly for doing things in ways that we know work, which means 
not inventing stuff until we really have to. And I've even argued 
strongly for doing things in ways that *permit* interoperation with 
those older devices - but not in the cases where doing so risks harming 
the security, stability and operational complexity of the installed base 
that is to come.

                     Harald


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<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    On 05/02/2012 08:19 PM, Marshall Eubanks wrote:<br>
    <blockquote
cite="mid:CAJNg7VLCGgJGpV1+YTdBGjMOLBj4x=2-xyu8bpcjRt8Riyo6hA@mail.gmail.com"
      type="cite">
      <pre wrap="">
My objection is that the proposed system will require middleware to
interoperate with the
vast number of videoconferencing sessions out there, most of which use
RTP. From the
standpoint of a video service provider, buying hardware to support
video to laptops is likely to
lead to requests that participants download some other software which
interoperates natively.

This is an existing business with a fairly large scale and installed
base. Not operating the way that they do is not likely to go over
well.

</pre>
    </blockquote>
    Marshall,<br>
    <br>
    I'd like to draw your attention to two numbers:<br>
    <br>
    - Number of installed room videoconferencing units: On the order of
    1 million.<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="http://www.polycom.com/global/documents/company/video_conferencing_by_the_numbers.pdf">http://www.polycom.com/global/documents/company/video_conferencing_by_the_numbers.pdf</a><br>
    <br>
    - Number of installed Chrome browsers: On the order of 200 million.<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a
href="http://www.techradar.com/news/internet/google-chrome-browser-hits-200-million-users-1033951">http://www.techradar.com/news/internet/google-chrome-browser-hits-200-million-users-1033951</a><br>
    <br>
    (pulling out Chrome just because I know we've promised to ship this
    stuff. And we auto-update, which means most of the users WILL be
    running a WebRTC-compatible browser the week after we release it.)<br>
    <br>
    I argue strongly for doing things in ways that we know work, which
    means not inventing stuff until we really have to. And I've even
    argued strongly for doing things in ways that *permit*
    interoperation with those older devices - but not in the cases where
    doing so risks harming the security, stability and operational
    complexity of the installed base that is to come.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
  </body>
</html>

--------------010408090701000807030005--

From harald@alvestrand.no  Wed May  2 23:11:22 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Use Case draft
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On 05/03/2012 07:05 AM, Ravindran, Parthasarathi wrote:
> Igor,
>
> When you call Company X, you need to assure that you are talking with Company X agent and not required to distinguish the agent.
>
> The identity could be just X (no userpart) or anonymous. There is no difference between anonymous and agent007@X as there is no means to route directly agent007@x in the call centre scenario.
It' s easy to construct scenarios where having the ability to set up a 
connection directly to agent007@x is desirable - consider the case of 
anonymous counselling, where a call gets dropped in the middle of the 
conversation; while the caller wishes to know that the callee is really 
a representative of "X anonymous", and both parties wish to remain 
anonymous as persons, when the call gets dropped in the middle of the 
conversation, the calling party has a strong incentive to continue the 
conversation with the same party, if possible.

In this case, having a callee identity with a lifetime of "one 
conversation" seems highly desirable; any distributed ID system (such as 
1st party BrowserID) should be able to easily support that.

            Harald




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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints SDES-SRTP + DTLS-SRTP [was RE: Use Case draft]
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Well,

maybe it's just the "signaling provider" that will tell to the browser
what to use:

- simpler interoperable SDES/SRTP (saying to the user "send SRTP here
IP:Port with that Key)
- stronger non-interoperable DTLS-SRTP (saying to the user "go
peer-to-peer")

For example if the signaling provider know that it's providing:
a) A call to a center call
b) Call trough PSTN gateway (web call to PSTN)
c) Voicemail access
d) Financial call (stock brokerage)
e) Enterprise call (that would just need to proxy all calls)

then the signaling provider will notice to the user's WebRTC stack
SDES/SRTP in a *very simple* way without ICE/DTLS-SRTP and all that
protocol complexity.

Instead if that specific VoIP uses does not require a VoIP server to
provide value added services (transcoding, conferencing, bridging to
other telephony networks, recording, whatever) it would just goes as
DTLS-SRTP.

Some previous link on the topic:

* On DTLS-SRTP trust model (and consideration for SDES-SRTP)
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04007.html

* End-to-end encryption vs end-to-end authentication (DTLS-SRTP / SDES-SRTP)
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04032.html

*  DTLS-SRTP with end-to-end security: Short Authentication String
http://www.ietf.org/mail-archive/web/rtcweb/current/msg04123.html

Fabio

On 5/3/12 7:23 AM, Ravindran, Parthasarathi wrote:
> Fabio,
> 
> Could you please explain how to differentiate through protocol means that the peer is site (gateway) or endpoint (webbrowser).
> 
> Thanks
> Partha 

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Use Case draft
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Date: Thu, 3 May 2012 07:39:27 +0000
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com><BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl><387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com><2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com><387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com><E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com><4F9EC0B2.10903@alcatel-lucent.com><101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C148913C5@inba-mail02.sonusnet.com> <4FA15C18.6040509@alcatel-lucent.com> <387F9047F55E8C42850AD6B3A7A03C6C14894717@inba-mail02.sonusnet.com> <4FA22187.2000307@alvestrand.no>
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Subject: Re: [rtcweb] Use Case draft
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Harald,

It is one of the main reason, the reliability & QOS of the established sess=
ion is very important in the telecom/Enterprise world and the call disconne=
ct for the established session due to technical reason will taken as an esc=
alation :-(=20

There are lot of reasons, the majority of the call centre may not prefer to=
 provide the agent-id to the customer as it is not manageable solution. Dyn=
amically deciding the available agent helps for better agent time utilizati=
on in the call center compare to direct dialing to the agent. The agent may=
 work in shift (in the different timezone) wherein you will not be able to =
reach him in the same identity itself. In these type of call center, the ca=
se id or your existing some id is the way to continue your earlier conversa=
tion if it is recorded in some form.=20

As an individual user, I don't want to register as a user of each website b=
efore calling their site to know about their product information.

As you wish, The premium user of the call center will be given the agent id=
entity to contact. For example, you will be provided with Bank agent identi=
ty for your banking account related queries. Please note that you MUST be p=
remium user of the bank to get that privilege or have to live with case-id =
or start from IVR & reach some new agent to start with.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Harald Alvestrand
>Sent: Thursday, May 03, 2012 11:41 AM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Use Case draft
>
>On 05/03/2012 07:05 AM, Ravindran, Parthasarathi wrote:
>> Igor,
>>
>> When you call Company X, you need to assure that you are talking with
>Company X agent and not required to distinguish the agent.
>>
>> The identity could be just X (no userpart) or anonymous. There is no
>difference between anonymous and agent007@X as there is no means to
>route directly agent007@x in the call centre scenario.
>It' s easy to construct scenarios where having the ability to set up a
>connection directly to agent007@x is desirable - consider the case of
>anonymous counselling, where a call gets dropped in the middle of the
>conversation; while the caller wishes to know that the callee is really
>a representative of "X anonymous", and both parties wish to remain
>anonymous as persons, when the call gets dropped in the middle of the
>conversation, the calling party has a strong incentive to continue the
>conversation with the same party, if possible.
>
>In this case, having a callee identity with a lifetime of "one
>conversation" seems highly desirable; any distributed ID system (such as
>1st party BrowserID) should be able to easily support that.
>
>            Harald
>
>
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From harald@alvestrand.no  Thu May  3 01:25:37 2012
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To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com><BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl><387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com><2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com><387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com><E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com><4F9EC0B2.10903@alcatel-lucent.com><101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <387F9047F55E8C42850AD6B3A7A03C6C148913C5@inba-mail02.sonusnet.com> <4FA15C18.6040509@alcatel-lucent.com> <387F9047F55E8C42850AD6B3A7A03C6C14894717@inba-mail02.sonusnet.com> <4FA22187.2000307@alvestrand.no> <387F9047F55E8C42850AD6B3A7A03C6C148947B8@inba-mail02.sonusnet.com >
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Use Case draft
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On 05/03/2012 09:39 AM, Ravindran, Parthasarathi wrote:
> Harald,
>
> It is one of the main reason, the reliability&  QOS of the established session is very important in the telecom/Enterprise world and the call disconnect for the established session due to technical reason will taken as an escalation :-(
I was thinking of scenarios like "the phone slipped out of my hand" and 
"I tripped over the PC power cord", actually.... there are plenty of 
reasons why calls get disconnected.
>
> There are lot of reasons, the majority of the call centre may not prefer to provide the agent-id to the customer as it is not manageable solution. Dynamically deciding the available agent helps for better agent time utilization in the call center compare to direct dialing to the agent. The agent may work in shift (in the different timezone) wherein you will not be able to reach him in the same identity itself. In these type of call center, the case id or your existing some id is the way to continue your earlier conversation if it is recorded in some form.
My point was not to dispute any of those for the cases you describe. But 
those cases are not the only cases; I've re-called into call centers 
where I've said "I was talking to John", and the responder says "OK, I 
see he's free, I'll transfer you".

The desire for "anonymous agents" is not synonymous with "doesn't 
support re-dial" - that was really my only point.
>
> As an individual user, I don't want to register as a user of each website before calling their site to know about their product information.
>
> As you wish, The premium user of the call center will be given the agent identity to contact. For example, you will be provided with Bank agent identity for your banking account related queries. Please note that you MUST be premium user of the bank to get that privilege or have to live with case-id or start from IVR&  reach some new agent to start with.
It seems we agree that some people will want to build applications where 
this functionality is available. That was really all I was trying to 
say, so I'm happy now.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Harald Alvestrand
>> Sent: Thursday, May 03, 2012 11:41 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use Case draft
>>
>> On 05/03/2012 07:05 AM, Ravindran, Parthasarathi wrote:
>>> Igor,
>>>
>>> When you call Company X, you need to assure that you are talking with
>> Company X agent and not required to distinguish the agent.
>>> The identity could be just X (no userpart) or anonymous. There is no
>> difference between anonymous and agent007@X as there is no means to
>> route directly agent007@x in the call centre scenario.
>> It' s easy to construct scenarios where having the ability to set up a
>> connection directly to agent007@x is desirable - consider the case of
>> anonymous counselling, where a call gets dropped in the middle of the
>> conversation; while the caller wishes to know that the callee is really
>> a representative of "X anonymous", and both parties wish to remain
>> anonymous as persons, when the call gets dropped in the middle of the
>> conversation, the calling party has a strong incentive to continue the
>> conversation with the same party, if possible.
>>
>> In this case, having a callee identity with a lifetime of "one
>> conversation" seems highly desirable; any distributed ID system (such as
>> 1st party BrowserID) should be able to easily support that.
>>
>>             Harald
>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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Date: Thu, 3 May 2012 14:54:59 +0530
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Thread-Topic: [rtcweb] interworking with non-WEBRTC endpoints [was RE: UseCase draft]
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Marshall Eubanks" <marshall.eubanks@gmail.com>, =?iso-8859-1?Q?I=F1aki_Baz_Castillo?= <ibc@aliax.net>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: UseCase draft]
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|My objection is that the proposed system will require middleware
|to interoperate with the vast number of videoconferencing sessions
|out there, most of which use RTP. From the standpoint of a video
|service provider, buying hardware to support video to laptops is
|likely to lead to requests that participants download some other=20
|software which interoperates natively.

I find this argument nothing different from those for interoperating =
with legacy phones which probably would outnumber videoconferencing =
systems out there by a very large number. Given that video conferencing =
systems are fairly advanced, rather than requiring customers to use =
insecure communication, service providers could require those =
manufacturers to support secure communication at minimal cost. I don't =
see how insecure communication would go well with all those customers =
who would like to use web based communication over the Internet.

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Marshall Eubanks
|Sent: Wednesday, May 02, 2012 11:49 PM
|To: I=F1aki Baz Castillo
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: =
UseCase draft]
|
|On Wed, May 2, 2012 at 1:54 PM, I=F1aki Baz Castillo <ibc@aliax.net> =
wrote:
|> 2012/5/2 Dan Wing <dwing@cisco.com>:
|>> For other decisions, such as if we disallow un-encrypted RTP by
|>> WEBRTC endpoints, we create a requirement that some device does
|>> the interworking between WEBRTC endpoints (which do only SRTP)
|>> and non-WEBRTC endpoints (which do RTP). =A0That means, for that
|>> interworking, we would adopt the interworking model on slide 7
|>> that I presented at IETF83,
|>> http://www.ietf.org/proceedings/83/slides/slides-83-WEBRTC-3.pdf
|>
|> Hi, the link is broken (404). Could you please point to a working =
one?
|>
|
|http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf
|
|> However, when I presented slide 7, there were objections at the =
microphone that this model 'is
|broken'.  I would like to understand the objections so we can reach =
consensus on how interworking from
|WEBRTC to non-WEBRTC is expected to occur.
|
|My objection is that the proposed system will require middleware to
|interoperate with the
|vast number of videoconferencing sessions out there, most of which use
|RTP. From the
|standpoint of a video service provider, buying hardware to support
|video to laptops is likely to
|lead to requests that participants download some other software which
|interoperates natively.
|
|This is an existing business with a fairly large scale and installed
|base. Not operating the way that they do is not likely to go over
|well.
|
|Regards
|Marshall
|
|> Thanks a lot.
|>
|> --
|> I=F1aki Baz Castillo
|> <ibc@aliax.net>
|> _______________________________________________
|> 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 Jim.Barnett@genesyslab.com  Thu May  3 06:06:14 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Stefan Hakansson LK" <stefan.lk.hakansson@ericsson.com>
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Yes, the use cases in  4.3 are browser-GW, though none of them really
covers the call center case (routing to an arbitrary agent, recording
the call, etc.)

However, the requirements in section 5 don't seem to mention legacy
connectivity, unless that's what F27 is supposed to cover ("The browser
MUST be able to initiate and accept a media session where the data
needed for establishment can be carried in SIP.")  It might be good to
have a specific legacy requirement along the forms of "The Browser must
be able to communicate with legacy devices that have the following
characteristics ...." Of course, this gets into the whole
interoperability discussion. For example, should the Browser be able to
interoperate with devices that don't support  ICE?  (I have been
assuming that the call center gateway would  support ICE). =20

The enterprise use cases that Roman has mentioned are separate from
this, and would add new requirements, since they involve webRTC within
the enterprise connecting to external webRTC endpoints.  In the call
center use case, the webRTC call is terminated at the gateway. =20

- Jim

-----Original Message-----
From: Stefan Hakansson LK [mailto:stefan.lk.hakansson@ericsson.com]=20
Sent: Wednesday, May 02, 2012 11:49 AM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

On 05/02/2012 04:39 PM, Jim Barnett wrote:
> When I say that this use case may not add further requirements, I mean

> that it looks like it would be possible to implement it given the=20
> current definitions of the protocols.  However, the current use cases=20
> are all written in terms of "the browser", which is not further
defined.
> But if "browser" means Mozilla, Chrome, etc., then I think it is=20
> important to add a use case in which one of the end points is not a=20
> browser, but an enterprise gateway (which will route the call to an=20
> employee of its choice, and may record the call, etc.) It is important

> to note that this is not a peer-to-peer use case; the gateway is not=20
> the caller's peer.  The employee that the caller ends up talking to=20
> may be considered a peer, but the webRTC call does not extend all the=20
> way to that employee - it stops at the gateway.

I think all use cases in section 4.3 are browser - GW cases. That said,
there may be good reasons for adding this one as well.

>
> This is a very different use case from any in the current document.
> That's why it's important to add it, even though (as far as I can=20
> tell) it doesn't require us to change any of the work we've done.
>
> - Jim
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> Behalf Of Stefan Hakansson LK
> Sent: Wednesday, May 02, 2012 4:46 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>
> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>> One way to describe the use case is to let the contact center's media

>> server/gateway serve as the webRTC endpoint.  Then all the issues of=20
>> call delivery, call monitoring, etc. disappear.  They are handled by=20
>> application software that sits behind the webRTC endpoint.  The=20
>> company I work for makes a good living selling software that deals=20
>> with all these issues - including bathroom breaks - and that's how we

>> would tend to think of this case.  To us, it's a new kind of=20
>> call/connection coming into the contact center, which we translate=20
>> into SIP at the border and then handle normally.
>>
>> It's not clear to me if this use case adds any extra requirements.
>
> I think this is important to sort out. If the use case does not add=20
> any extra requirements, what's the point of adding it?
>
>> We would just have to be careful not to assume that a webRTC endpoint

>> is always a person/browser-based user agent.  It may seem a bit=20
>> unsettling that the webRTC endpoint can distribute the call somewhere

>> else and let others listen in, but as far as I can tell that is=20
>> already the case.  If Bob calls Alice with full authentication and=20
>> security, he can be sure that he is connected to Alice's user agent=20
>> and that no one in between can listen in, but there's nothing=20
>> stopping
>
>> Alice from recording the audio, or forwarding it to a third party. =20
>> So
>
>> Bob could in fact be talking to Mary if that's how Alice wants to=20
>> arrange things (_behind_ her user agent).  In general, Bob is assured

>> only that he is talking to someone Alice wants him to talk to, and=20
>> that no one can snoop without Alice's permission.  That's very much=20
>> the way things work with the call center - you are sure that you are
>> 1) connected securely to your bank 2) talking to someone that the=20
>> bank
>
>> wants you to talk to 3) being recorded or snooped on only when the=20
>> bank explicitly chooses to do so.
>>
>> - Jim
>>
>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>
>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>> Andrew<andrew.hutton@siemens-enterprise.com>   wrote:
>>> Whether anybody has been successful in the past with this type of=20
>>> use
>
>>> case is I think irrelevant.
>>>
>>>
>>>
>>> The enterprise call centre use case is I think a vital use case=20
>>> because it is a scenario in which one user is only concerned that=20
>>> they can securely reach an organization/domain and is not concerned=20
>>> about the individual within that domain  that they communicate with.
>
>>> A suspect quite a large percentage of RTCWEB applications will be=20
>>> like this and it is not covered in the current use case draft.
>>
>> I agree that this is a very useful use case and one I think is going=20
>> to get a lot of traction. There is a very solid business case for=20
>> this.  However, I have a fair amount of experience with a video call=20
>> center for a client, and it is not as simple as it might seem.
>>
>> The essence of course is that you get the next available person,=20
>> i.e.,
>
>> it is anycast. Determining who the next available person is is not=20
>> trivial, nor is error recovery. (If I call you, and you don't answer=20
>> or the call drops or whatever,  I can leave a message or try later.=20
>> If
>
>> I call a help desk, and this happens, I want a new agent, ideally
>> automatically.) Call forwarding (e.g., first tier to second tier=20
>> technical support) is essential, and it may be anycast or directed.
>> There are also some security oddities  - if I am connecting from=20
>> home,
>
>> I may need to authenticate, use a credit card, etc. If I am=20
>> connecting
>
>> from inside a store, and providing in store video technical support=20
>> is
>
>> big part of the market, then the store authenticates me off line and=20
>> the call really should just be a button push, which implies that the=20
>> store has previously authenticated some sort of master session. In=20
>> addition, unlike most video calls, in the enterprise call center a=20
>> supervisor may need to be able to monitor (i.e., watch) a call, and=20
>> in
>
>> some circumstances (financial or medical calls, for example) there=20
>> will need to be third party recording. I believe that  these details=20
>> would be different from the typical RTCWEB scenario.
>>
>> Also, there will be a temptation to do the anycasting by the=20
>> techniques used to load balance servers in a data center, but I think

>> that may not be sufficient. The call "center" may in fact be spread=20
>> completely across the planet (daytime support in the US, nighttime=20
>> support in India, for example) and be on multiple autonomous systems=20
>> (and even from people's homes), which gives rise to some of the=20
>> transport issues NVO3 may face, but without any opportunity for=20
>> packet
>
>> tagging. Plus, there will complicated rules about who can be selected

>> next. RTCWEB shouldn't worry about the intricacies of bathroom break=20
>> policies; these complexities should be dealt with by an=20
>> enterprise-side database, which to me (together with some of the=20
>> other
>
>> issues above) suggests that this would probably benefit from API=20
>> support.
>>
>> Regards Marshall
>>
>>
>>>
>>>
>>>
>>> So I think we need it.
>>>
>>>
>>>
>>> Regards
>>>
>>> Andy
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>> rtcweb@ietf.org
>>>
>>>
>>> Subject: Re: [rtcweb] Use Case draft
>>>
>>>
>>>
>>> Without numbers it is impossible to argue, but, if we talk about the

>>> perceived need, I disagree.  Think of the people who travel abroad=20
>>> and cannot call the 800 number. (I routinely use Web interface for=20
>>> calls when traveling.)
>>>
>>>
>>>
>>> I am all for  the use case, as described by Jim.
>>>
>>> Igor
>>>
>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>
>>> ...
>>>
>>> I can't tell you the actual numbers, but when presented with the=20
>>> choice of calling a toll free number
>>>
>>> or clicking a button marked "free internet call" - almost no-one on=20
>>> a
>
>>> real, busy site clicked the button.
>>>
>>> ( for every button click there were several orders of magnitude more
>>> 0800 calls from that page).
>>>
>>>
>>>
>>>
>>>
>>> So from my perspective this is a legacy interop use case with almost

>>> zero user acceptance.
>>>
>>>
>>>
>>> (as far as I can see no-one has made this use-case desirable in=20
>>> practice yet.)
>>>
>>> Tim.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>>
>>> rtcweb mailing list
>>>
>>> rtcweb@ietf.org
>>>
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>>
>>> _______________________________________________ rtcweb mailing list=20
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>> _______________________________________________ rtcweb mailing list=20
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>> _______________________________________________ rtcweb 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


From stefan.lk.hakansson@ericsson.com  Thu May  3 07:43:55 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Use Case draft
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On 05/03/2012 03:06 PM, Jim Barnett wrote:
> Yes, the use cases in  4.3 are browser-GW, though none of them really
> covers the call center case (routing to an arbitrary agent, recording
> the call, etc.)
>
> However, the requirements in section 5 don't seem to mention legacy
> connectivity, unless that's what F27 is supposed to cover ("The browser
> MUST be able to initiate and accept a media session where the data
> needed for establishment can be carried in SIP.")  It might be good to
> have a specific legacy requirement along the forms of "The Browser must
> be able to communicate with legacy devices that have the following
> characteristics ...." Of course, this gets into the whole
> interoperability discussion. For example, should the Browser be able to
> interoperate with devices that don't support  ICE?  (I have been
> assuming that the call center gateway would  support ICE).

I think reqs F21 and F22 are typical legacy device interop related 
requirements as well.

What we did when writing the document was to assume that the device in 
the other end of a connection would have to implement basically the same 
set of protocols as a browser, then we added some specific requirements 
to simplify/reduce cost (carry signaling in SIP, telco codec, DTMF).

I say this to explain why there is no specific requirement on "must be 
able to interop with a device having the following characteristics", not 
to say that such a req does not make sense.

That said, I'm afraid that the work required to specify the exact 
characteristics of devices that the browser must be able to interop with 
would be very big. Would not the discussion lead into every nitty gritty 
detail?

>
> The enterprise use cases that Roman has mentioned are separate from
> this, and would add new requirements, since they involve webRTC within
> the enterprise connecting to external webRTC endpoints.  In the call
> center use case, the webRTC call is terminated at the gateway.
>
> - Jim
>
> -----Original Message-----
> From: Stefan Hakansson LK [mailto:stefan.lk.hakansson@ericsson.com]
> Sent: Wednesday, May 02, 2012 11:49 AM
> To: Jim Barnett
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>
> On 05/02/2012 04:39 PM, Jim Barnett wrote:
>> When I say that this use case may not add further requirements, I mean
>
>> that it looks like it would be possible to implement it given the
>> current definitions of the protocols.  However, the current use cases
>> are all written in terms of "the browser", which is not further
> defined.
>> But if "browser" means Mozilla, Chrome, etc., then I think it is
>> important to add a use case in which one of the end points is not a
>> browser, but an enterprise gateway (which will route the call to an
>> employee of its choice, and may record the call, etc.) It is important
>
>> to note that this is not a peer-to-peer use case; the gateway is not
>> the caller's peer.  The employee that the caller ends up talking to
>> may be considered a peer, but the webRTC call does not extend all the
>> way to that employee - it stops at the gateway.
>
> I think all use cases in section 4.3 are browser - GW cases. That said,
> there may be good reasons for adding this one as well.
>
>>
>> This is a very different use case from any in the current document.
>> That's why it's important to add it, even though (as far as I can
>> tell) it doesn't require us to change any of the work we've done.
>>
>> - Jim
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Stefan Hakansson LK
>> Sent: Wednesday, May 02, 2012 4:46 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use Case draft
>>
>> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>>> One way to describe the use case is to let the contact center's media
>
>>> server/gateway serve as the webRTC endpoint.  Then all the issues of
>>> call delivery, call monitoring, etc. disappear.  They are handled by
>>> application software that sits behind the webRTC endpoint.  The
>>> company I work for makes a good living selling software that deals
>>> with all these issues - including bathroom breaks - and that's how we
>
>>> would tend to think of this case.  To us, it's a new kind of
>>> call/connection coming into the contact center, which we translate
>>> into SIP at the border and then handle normally.
>>>
>>> It's not clear to me if this use case adds any extra requirements.
>>
>> I think this is important to sort out. If the use case does not add
>> any extra requirements, what's the point of adding it?
>>
>>> We would just have to be careful not to assume that a webRTC endpoint
>
>>> is always a person/browser-based user agent.  It may seem a bit
>>> unsettling that the webRTC endpoint can distribute the call somewhere
>
>>> else and let others listen in, but as far as I can tell that is
>>> already the case.  If Bob calls Alice with full authentication and
>>> security, he can be sure that he is connected to Alice's user agent
>>> and that no one in between can listen in, but there's nothing
>>> stopping
>>
>>> Alice from recording the audio, or forwarding it to a third party.
>>> So
>>
>>> Bob could in fact be talking to Mary if that's how Alice wants to
>>> arrange things (_behind_ her user agent).  In general, Bob is assured
>
>>> only that he is talking to someone Alice wants him to talk to, and
>>> that no one can snoop without Alice's permission.  That's very much
>>> the way things work with the call center - you are sure that you are
>>> 1) connected securely to your bank 2) talking to someone that the
>>> bank
>>
>>> wants you to talk to 3) being recorded or snooped on only when the
>>> bank explicitly chooses to do so.
>>>
>>> - Jim
>>>
>>> -----Original Message----- From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>>
>>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>>> Andrew<andrew.hutton@siemens-enterprise.com>    wrote:
>>>> Whether anybody has been successful in the past with this type of
>>>> use
>>
>>>> case is I think irrelevant.
>>>>
>>>>
>>>>
>>>> The enterprise call centre use case is I think a vital use case
>>>> because it is a scenario in which one user is only concerned that
>>>> they can securely reach an organization/domain and is not concerned
>>>> about the individual within that domain  that they communicate with.
>>
>>>> A suspect quite a large percentage of RTCWEB applications will be
>>>> like this and it is not covered in the current use case draft.
>>>
>>> I agree that this is a very useful use case and one I think is going
>>> to get a lot of traction. There is a very solid business case for
>>> this.  However, I have a fair amount of experience with a video call
>>> center for a client, and it is not as simple as it might seem.
>>>
>>> The essence of course is that you get the next available person,
>>> i.e.,
>>
>>> it is anycast. Determining who the next available person is is not
>>> trivial, nor is error recovery. (If I call you, and you don't answer
>>> or the call drops or whatever,  I can leave a message or try later.
>>> If
>>
>>> I call a help desk, and this happens, I want a new agent, ideally
>>> automatically.) Call forwarding (e.g., first tier to second tier
>>> technical support) is essential, and it may be anycast or directed.
>>> There are also some security oddities  - if I am connecting from
>>> home,
>>
>>> I may need to authenticate, use a credit card, etc. If I am
>>> connecting
>>
>>> from inside a store, and providing in store video technical support
>>> is
>>
>>> big part of the market, then the store authenticates me off line and
>>> the call really should just be a button push, which implies that the
>>> store has previously authenticated some sort of master session. In
>>> addition, unlike most video calls, in the enterprise call center a
>>> supervisor may need to be able to monitor (i.e., watch) a call, and
>>> in
>>
>>> some circumstances (financial or medical calls, for example) there
>>> will need to be third party recording. I believe that  these details
>>> would be different from the typical RTCWEB scenario.
>>>
>>> Also, there will be a temptation to do the anycasting by the
>>> techniques used to load balance servers in a data center, but I think
>
>>> that may not be sufficient. The call "center" may in fact be spread
>>> completely across the planet (daytime support in the US, nighttime
>>> support in India, for example) and be on multiple autonomous systems
>>> (and even from people's homes), which gives rise to some of the
>>> transport issues NVO3 may face, but without any opportunity for
>>> packet
>>
>>> tagging. Plus, there will complicated rules about who can be selected
>
>>> next. RTCWEB shouldn't worry about the intricacies of bathroom break
>>> policies; these complexities should be dealt with by an
>>> enterprise-side database, which to me (together with some of the
>>> other
>>
>>> issues above) suggests that this would probably benefit from API
>>> support.
>>>
>>> Regards Marshall
>>>
>>>
>>>>
>>>>
>>>>
>>>> So I think we need it.
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Andy
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>>> rtcweb@ietf.org
>>>>
>>>>
>>>> Subject: Re: [rtcweb] Use Case draft
>>>>
>>>>
>>>>
>>>> Without numbers it is impossible to argue, but, if we talk about the
>
>>>> perceived need, I disagree.  Think of the people who travel abroad
>>>> and cannot call the 800 number. (I routinely use Web interface for
>>>> calls when traveling.)
>>>>
>>>>
>>>>
>>>> I am all for  the use case, as described by Jim.
>>>>
>>>> Igor
>>>>
>>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>>
>>>> ...
>>>>
>>>> I can't tell you the actual numbers, but when presented with the
>>>> choice of calling a toll free number
>>>>
>>>> or clicking a button marked "free internet call" - almost no-one on
>>>> a
>>
>>>> real, busy site clicked the button.
>>>>
>>>> ( for every button click there were several orders of magnitude more
>>>> 0800 calls from that page).
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> So from my perspective this is a legacy interop use case with almost
>
>>>> zero user acceptance.
>>>>
>>>>
>>>>
>>>> (as far as I can see no-one has made this use-case desirable in
>>>> practice yet.)
>>>>
>>>> Tim.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> 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
>>> _______________________________________________ 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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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com>, <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl>, <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com>, <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com>, <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com>, <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com>, <4F9EC0B2.10903@alcatel-lucent.com>, <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net>, <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com>, <4FA0F43E.4020308@ericsson.com>, <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com>, <013101cd288c$09328250$1b9786f0$@com>, <CALiegf=RsHHf9jCBhE55t7qFUpts8yJ1c8qUX12nc_vd4vgjSQ@mail.gmail.com>, <CAJNg7VLCGgJGpV1+YTdBGjMOLBj4x=2-xyu8bpcjRt8Riyo6hA@mail.gmail.com>, <4FA21FA3.1090702@alvestrand .no>
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I concur with Marshall's objections=2C based on my discussions with custome=
rs.=20

Video conferencing systems are frequently purchased with the expectation th=
at they will be
used to communicate outside the organization=2C as well as with other devic=
es=2C including PCs=2C
tablets and mobile devices.  =20

As an example=2C an enterprise with a video conferencing system may use it =
to communicate
with employees in the field who are using a notebook=2C tablet or mobile de=
vice.=20

Therefore=2C while an enterprise may have a finite number of video conferen=
cing units=2C  it will
often deploy them along with orders of magnitude more devices that interope=
rate with those
units. =20

The usual requirements for a desirable mobile or tablet implementation appl=
y here -- power
management (e.g. ability to utilize native encode/decode hardware) is an im=
portant capability.  Also=2C
the cost of supporting video transcoding for a large number of devices will=
 frequently be prohibitive=2C
so that the expectation is that videoconferencing systems and implementatio=
ns on PCs=2C tablets
and mobile devices will be able to negotiate a mutually supported codec.=20





 =20
   =20
 =20
 =20
    On 05/02/2012 08:19 PM=2C Marshall Eubanks wrote:

   =20
      My objection is that the proposed system will require middleware to
interoperate with the
vast number of videoconferencing sessions out there=2C most of which use
RTP. From the
standpoint of a video service provider=2C buying hardware to support
video to laptops is likely to
lead to requests that participants download some other software which
interoperates natively.

This is an existing business with a fairly large scale and installed
base. Not operating the way that they do is not likely to go over
well.


   =20
    Marshall=2C

   =20

    I'd like to draw your attention to two numbers:

   =20

    - Number of installed room videoconferencing units: On the order of
    1 million.

   =20

   =20
    http://www.polycom.com/global/documents/company/video_conferencing_by_t=
he_numbers.pdf

   =20

    - Number of installed Chrome browsers: On the order of 200 million.

   =20

   =20
    http://www.techradar.com/news/internet/google-chrome-browser-hits-200-m=
illion-users-1033951

   =20

    (pulling out Chrome just because I know we've promised to ship this
    stuff. And we auto-update=2C which means most of the users WILL be
    running a WebRTC-compatible browser the week after we release it.)

   =20

    I argue strongly for doing things in ways that we know work=2C which
    means not inventing stuff until we really have to. And I've even
    argued strongly for doing things in ways that *permit*
    interoperation with those older devices - but not in the cases where
    doing so risks harming the security=2C stability and operational
    complexity of the installed base that is to come.

   =20

                        Harald

   =20

 =20


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

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<html>
<head>
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<body class=3D'hmmessage'><div dir=3D'ltr'>
I concur with Marshall's objections=2C based on my discussions with custome=
rs. <br><br>Video conferencing systems are frequently purchased with the ex=
pectation that they will be<br>used to communicate outside the organization=
=2C as well as with other devices=2C including PCs=2C<br>tablets and mobile=
 devices.&nbsp=3B&nbsp=3B <br><br>As an example=2C an enterprise with a vid=
eo conferencing system may use it to communicate<br>with employees in the f=
ield who are using a notebook=2C tablet or mobile device. <br><br>Therefore=
=2C while an enterprise may have a finite number of video conferencing unit=
s=2C&nbsp=3B it will<br>often deploy them along with orders of magnitude mo=
re devices that interoperate with those<br>units.&nbsp=3B <br><br>The usual=
 requirements for a desirable mobile or tablet implementation apply here --=
 power<br>management (e.g. ability to utilize native encode/decode hardware=
) is an important capability.&nbsp=3B Also=2C<br>the cost of supporting vid=
eo transcoding for a large number of devices will frequently be prohibitive=
=2C<br>so that the expectation is that videoconferencing systems and implem=
entations on PCs=2C tablets<br>and mobile devices will be able to negotiate=
 a mutually supported codec. <br><br><br><br><div><br>
 =20
   =20
 =20
 =20
    On 05/02/2012 08:19 PM=2C Marshall Eubanks wrote:<br>
    <blockquote cite=3D"mid:CAJNg7VLCGgJGpV1+YTdBGjMOLBj4x=3D2-xyu8bpcjRt8R=
iyo6hA@mail.gmail.com">
      <pre>My objection is that the proposed system will require middleware=
 to
interoperate with the
vast number of videoconferencing sessions out there=2C most of which use
RTP. From the
standpoint of a video service provider=2C buying hardware to support
video to laptops is likely to
lead to requests that participants download some other software which
interoperates natively.

This is an existing business with a fairly large scale and installed
base. Not operating the way that they do is not likely to go over
well.

</pre>
    </blockquote>
    Marshall=2C<br>
    <br>
    I'd like to draw your attention to two numbers:<br>
    <br>
    - Number of installed room videoconferencing units: On the order of
    1 million.<br>
    <br>
   =20
    <a href=3D"http://www.polycom.com/global/documents/company/video_confer=
encing_by_the_numbers.pdf" target=3D"_blank">http://www.polycom.com/global/=
documents/company/video_conferencing_by_the_numbers.pdf</a><br>
    <br>
    - Number of installed Chrome browsers: On the order of 200 million.<br>
    <br>
   =20
    <a href=3D"http://www.techradar.com/news/internet/google-chrome-browser=
-hits-200-million-users-1033951" target=3D"_blank">http://www.techradar.com=
/news/internet/google-chrome-browser-hits-200-million-users-1033951</a><br>
    <br>
    (pulling out Chrome just because I know we've promised to ship this
    stuff. And we auto-update=2C which means most of the users WILL be
    running a WebRTC-compatible browser the week after we release it.)<br>
    <br>
    I argue strongly for doing things in ways that we know work=2C which
    means not inventing stuff until we really have to. And I've even
    argued strongly for doing things in ways that *permit*
    interoperation with those older devices - but not in the cases where
    doing so risks harming the security=2C stability and operational
    complexity of the installed base that is to come.<br>
    <br>
    &nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=
=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B&nbsp=3B=
&nbsp=3B Harald<br>
    <br>
 =20

<br>_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
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From Jim.Barnett@genesyslab.com  Thu May  3 07:52:45 2012
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com><BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl><387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com><2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com><387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com><E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com><4F9EC0B2.10903@alcatel-lucent.com><101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net><CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com><E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <4FA1575C.4050508@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F4BF@NAHALD.us.int.genesyslab.com> <4FA299A8.6080003@ericsson.com>
From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
To: "Stefan Hakansson LK" <stefan.lk.hakansson@ericsson.com>
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Yes, I think that you're right that specifying the exact set of legacy
devices would be a painful task.  It may be that the current
requirements are sufficient for the call center use case, as long as we
understand that in that use case it is the call center gateway, and not
the agent workstation/browser, that is the webRTC endpoint.  The call
center's requirements for routing, recording/monitoring, etc. are
awfully hard to combine with webRTC if we think of the webRTC call as
going all the way to the agent. =20

- Jim
P.S. Roman's enterprise  use cases are distinct from the call center
case, and do involve webRTC connections extending all the way into the
enterprise. =20

-----Original Message-----
From: Stefan Hakansson LK [mailto:stefan.lk.hakansson@ericsson.com]=20
Sent: Thursday, May 03, 2012 10:44 AM
To: Jim Barnett
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft

On 05/03/2012 03:06 PM, Jim Barnett wrote:
> Yes, the use cases in  4.3 are browser-GW, though none of them really=20
> covers the call center case (routing to an arbitrary agent, recording=20
> the call, etc.)
>
> However, the requirements in section 5 don't seem to mention legacy=20
> connectivity, unless that's what F27 is supposed to cover ("The=20
> browser MUST be able to initiate and accept a media session where the=20
> data needed for establishment can be carried in SIP.")  It might be=20
> good to have a specific legacy requirement along the forms of "The=20
> Browser must be able to communicate with legacy devices that have the=20
> following characteristics ...." Of course, this gets into the whole=20
> interoperability discussion. For example, should the Browser be able=20
> to interoperate with devices that don't support  ICE?  (I have been=20
> assuming that the call center gateway would  support ICE).

I think reqs F21 and F22 are typical legacy device interop related
requirements as well.

What we did when writing the document was to assume that the device in
the other end of a connection would have to implement basically the same
set of protocols as a browser, then we added some specific requirements
to simplify/reduce cost (carry signaling in SIP, telco codec, DTMF).

I say this to explain why there is no specific requirement on "must be
able to interop with a device having the following characteristics", not
to say that such a req does not make sense.

That said, I'm afraid that the work required to specify the exact
characteristics of devices that the browser must be able to interop with
would be very big. Would not the discussion lead into every nitty gritty
detail?

>
> The enterprise use cases that Roman has mentioned are separate from=20
> this, and would add new requirements, since they involve webRTC within

> the enterprise connecting to external webRTC endpoints.  In the call=20
> center use case, the webRTC call is terminated at the gateway.
>
> - Jim
>
> -----Original Message-----
> From: Stefan Hakansson LK [mailto:stefan.lk.hakansson@ericsson.com]
> Sent: Wednesday, May 02, 2012 11:49 AM
> To: Jim Barnett
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft
>
> On 05/02/2012 04:39 PM, Jim Barnett wrote:
>> When I say that this use case may not add further requirements, I=20
>> mean
>
>> that it looks like it would be possible to implement it given the=20
>> current definitions of the protocols.  However, the current use cases

>> are all written in terms of "the browser", which is not further
> defined.
>> But if "browser" means Mozilla, Chrome, etc., then I think it is=20
>> important to add a use case in which one of the end points is not a=20
>> browser, but an enterprise gateway (which will route the call to an=20
>> employee of its choice, and may record the call, etc.) It is=20
>> important
>
>> to note that this is not a peer-to-peer use case; the gateway is not=20
>> the caller's peer.  The employee that the caller ends up talking to=20
>> may be considered a peer, but the webRTC call does not extend all the

>> way to that employee - it stops at the gateway.
>
> I think all use cases in section 4.3 are browser - GW cases. That=20
> said, there may be good reasons for adding this one as well.
>
>>
>> This is a very different use case from any in the current document.
>> That's why it's important to add it, even though (as far as I can
>> tell) it doesn't require us to change any of the work we've done.
>>
>> - Jim
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>> Behalf Of Stefan Hakansson LK
>> Sent: Wednesday, May 02, 2012 4:46 AM
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use Case draft
>>
>> On 05/01/2012 02:05 PM, Jim Barnett wrote:
>>> One way to describe the use case is to let the contact center's=20
>>> media
>
>>> server/gateway serve as the webRTC endpoint.  Then all the issues of

>>> call delivery, call monitoring, etc. disappear.  They are handled by

>>> application software that sits behind the webRTC endpoint.  The=20
>>> company I work for makes a good living selling software that deals=20
>>> with all these issues - including bathroom breaks - and that's how=20
>>> we
>
>>> would tend to think of this case.  To us, it's a new kind of=20
>>> call/connection coming into the contact center, which we translate=20
>>> into SIP at the border and then handle normally.
>>>
>>> It's not clear to me if this use case adds any extra requirements.
>>
>> I think this is important to sort out. If the use case does not add=20
>> any extra requirements, what's the point of adding it?
>>
>>> We would just have to be careful not to assume that a webRTC=20
>>> endpoint
>
>>> is always a person/browser-based user agent.  It may seem a bit=20
>>> unsettling that the webRTC endpoint can distribute the call=20
>>> somewhere
>
>>> else and let others listen in, but as far as I can tell that is=20
>>> already the case.  If Bob calls Alice with full authentication and=20
>>> security, he can be sure that he is connected to Alice's user agent=20
>>> and that no one in between can listen in, but there's nothing=20
>>> stopping
>>
>>> Alice from recording the audio, or forwarding it to a third party.
>>> So
>>
>>> Bob could in fact be talking to Mary if that's how Alice wants to=20
>>> arrange things (_behind_ her user agent).  In general, Bob is=20
>>> assured
>
>>> only that he is talking to someone Alice wants him to talk to, and=20
>>> that no one can snoop without Alice's permission.  That's very much=20
>>> the way things work with the call center - you are sure that you are
>>> 1) connected securely to your bank 2) talking to someone that the=20
>>> bank
>>
>>> wants you to talk to 3) being recorded or snooped on only when the=20
>>> bank explicitly chooses to do so.
>>>
>>> - Jim
>>>
>>> -----Original Message----- From: rtcweb-bounces@ietf.org=20
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Marshall Eubanks Sent:
>>> Monday, April 30, 2012 11:42 PM To: Hutton, Andrew Cc:
>>> rtcweb@ietf.org Subject: Re: [rtcweb] Use Case draft
>>>
>>> On Mon, Apr 30, 2012 at 2:31 PM, Hutton,
>>> Andrew<andrew.hutton@siemens-enterprise.com>    wrote:
>>>> Whether anybody has been successful in the past with this type of=20
>>>> use
>>
>>>> case is I think irrelevant.
>>>>
>>>>
>>>>
>>>> The enterprise call centre use case is I think a vital use case=20
>>>> because it is a scenario in which one user is only concerned that=20
>>>> they can securely reach an organization/domain and is not concerned

>>>> about the individual within that domain  that they communicate
with.
>>
>>>> A suspect quite a large percentage of RTCWEB applications will be=20
>>>> like this and it is not covered in the current use case draft.
>>>
>>> I agree that this is a very useful use case and one I think is going

>>> to get a lot of traction. There is a very solid business case for=20
>>> this.  However, I have a fair amount of experience with a video call

>>> center for a client, and it is not as simple as it might seem.
>>>
>>> The essence of course is that you get the next available person,=20
>>> i.e.,
>>
>>> it is anycast. Determining who the next available person is is not=20
>>> trivial, nor is error recovery. (If I call you, and you don't answer

>>> or the call drops or whatever,  I can leave a message or try later.
>>> If
>>
>>> I call a help desk, and this happens, I want a new agent, ideally
>>> automatically.) Call forwarding (e.g., first tier to second tier=20
>>> technical support) is essential, and it may be anycast or directed.
>>> There are also some security oddities  - if I am connecting from=20
>>> home,
>>
>>> I may need to authenticate, use a credit card, etc. If I am=20
>>> connecting
>>
>>> from inside a store, and providing in store video technical support=20
>>> is
>>
>>> big part of the market, then the store authenticates me off line and

>>> the call really should just be a button push, which implies that the

>>> store has previously authenticated some sort of master session. In=20
>>> addition, unlike most video calls, in the enterprise call center a=20
>>> supervisor may need to be able to monitor (i.e., watch) a call, and=20
>>> in
>>
>>> some circumstances (financial or medical calls, for example) there=20
>>> will need to be third party recording. I believe that  these details

>>> would be different from the typical RTCWEB scenario.
>>>
>>> Also, there will be a temptation to do the anycasting by the=20
>>> techniques used to load balance servers in a data center, but I=20
>>> think
>
>>> that may not be sufficient. The call "center" may in fact be spread=20
>>> completely across the planet (daytime support in the US, nighttime=20
>>> support in India, for example) and be on multiple autonomous systems

>>> (and even from people's homes), which gives rise to some of the=20
>>> transport issues NVO3 may face, but without any opportunity for=20
>>> packet
>>
>>> tagging. Plus, there will complicated rules about who can be=20
>>> selected
>
>>> next. RTCWEB shouldn't worry about the intricacies of bathroom break

>>> policies; these complexities should be dealt with by an=20
>>> enterprise-side database, which to me (together with some of the=20
>>> other
>>
>>> issues above) suggests that this would probably benefit from API=20
>>> support.
>>>
>>> Regards Marshall
>>>
>>>
>>>>
>>>>
>>>>
>>>> So I think we need it.
>>>>
>>>>
>>>>
>>>> Regards
>>>>
>>>> Andy
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
>>>> Behalf Of Igor Faynberg Sent: 30 April 2012 17:41 To:
>>>> rtcweb@ietf.org
>>>>
>>>>
>>>> Subject: Re: [rtcweb] Use Case draft
>>>>
>>>>
>>>>
>>>> Without numbers it is impossible to argue, but, if we talk about=20
>>>> the
>
>>>> perceived need, I disagree.  Think of the people who travel abroad=20
>>>> and cannot call the 800 number. (I routinely use Web interface for=20
>>>> calls when traveling.)
>>>>
>>>>
>>>>
>>>> I am all for  the use case, as described by Jim.
>>>>
>>>> Igor
>>>>
>>>> On 4/30/2012 9:54 AM, Tim Panton wrote:
>>>>
>>>> ...
>>>>
>>>> I can't tell you the actual numbers, but when presented with the=20
>>>> choice of calling a toll free number
>>>>
>>>> or clicking a button marked "free internet call" - almost no-one on

>>>> a
>>
>>>> real, busy site clicked the button.
>>>>
>>>> ( for every button click there were several orders of magnitude=20
>>>> more
>>>> 0800 calls from that page).
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> So from my perspective this is a legacy interop use case with=20
>>>> almost
>
>>>> zero user acceptance.
>>>>
>>>>
>>>>
>>>> (as far as I can see no-one has made this use-case desirable in=20
>>>> practice yet.)
>>>>
>>>> Tim.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>>
>>>> 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=20
>>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
>>> _______________________________________________ rtcweb 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
>


From roman@telurix.com  Thu May  3 08:01:17 2012
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <CA+rAfUMfZBHxAjR0zmwhSxftQbPK0X3sNuxu4UV6bnMvnJ_9xA@mail.gmail.com> <CAOJ7v-3K=NK6zCLr-E-aSHDKVv2hMqnUhZKroX0YoBTo6-nQ+Q@mail.gmail.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se>
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On Thu, May 3, 2012 at 2:02 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> You are absolutely right. I still question whether we need to support
> parallel forking. If you say it occurs often, I can't argue against you,
> even though my own experience is different :)
>
>
You do end up with parallel forking when you allow registrations from
multiple devices. This is a policy option, but once you enable this,
parallel forking or B2BUA are the only options. Furthermore, even if you do
serial forking, you end up with multiple parallel media streams being
received by the client due to loose synchronization between RTP streams and
SIP signaling. The main discrepancy between current WebRTC signaling model
and SIP is that in SIP, when you do forking, you still have separate
dialogs that each maintain state. Even if only render media from one RTP
stream, black hole all the other media, and send media to one RTP
destination, you still have dialog specific SIP and ICE state that you can
update (for instance due to UPDATE transaction) and recover if the
currently selected dialog ended up not being the one that was selected in
200 OK. This is not an option with WebRTC.


> (Another option, if we want to support parallel forking, would of course
> be to create a completely new PeerConnection, with a *new* local IP
> address:port. But, you would of course then have to provide that
> information to the remote peer.)
>

This would probably do absolutely nothing to SIP interop. And if we do not
need SIP interop we can live without forking (at the cost of being slightly
less efficient in some call setup scenarios).


> I have no strong feeling on whether we want to do cloning, but do people
> agree that, for a given PeerConnection, we only need to support a single
> remote peer (which can be modified, though)?
>

I think if we do cloning we will only need to support a single remote peer
per PeerConnection. Strictly speaking even updates due to provisional
responses would not be necessary, since you can always clone the connection
and provide a new answer to it instead.
_____________
Roman Shpount

--0015175d67b4abfb7204bf2312d4
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<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, May 3, 20=
12 at 2:02 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@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"h5">You are absolutely right. =
I still question whether we need to support parallel forking. If you say it=
 occurs often, I can&#39;t argue against you, even though my own experience=
 is different :)<br>
</div>
<br></blockquote><div><br>You do end up with parallel forking when you allo=
w registrations from multiple devices. This is a policy option, but once yo=
u enable this, parallel forking or B2BUA are the only options. Furthermore,=
 even if you do serial forking, you end up with multiple parallel media str=
eams being received by the client due to loose synchronization between RTP =
streams and SIP signaling. The main discrepancy between current WebRTC sign=
aling model and SIP is that in SIP, when you do forking, you still have sep=
arate dialogs that each maintain state. Even if only render media from one =
RTP stream, black hole all the other media, and send media to one RTP desti=
nation, you still have dialog specific SIP and ICE state that you can updat=
e (for instance due to UPDATE transaction) and recover if the currently sel=
ected dialog ended up not being the one that was selected in 200 OK. This i=
s not an option with WebRTC.<br>
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8e=
x;border-left:1px solid rgb(204,204,204);padding-left:1ex">
(Another option, if we want to support parallel forking, would of course be=
 to create a completely new PeerConnection, with a *new* local IP address:p=
ort. But, you would of course then have to provide that information to the =
remote peer.)<br>
</blockquote><div><br>This would probably do absolutely nothing to SIP inte=
rop. And if we do not need SIP interop we can live without forking (at the =
cost of being slightly less efficient in some call setup scenarios).=A0 <br=
>
</div><div>=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0=
pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
I have no strong feeling on whether we want to do cloning, but do people ag=
ree that, for a given PeerConnection, we only need to support a single remo=
te peer (which can be modified, though)?<br></blockquote></div><br>I think =
if we do cloning we will only need to support a single remote peer per Peer=
Connection. Strictly speaking even updates due to provisional responses wou=
ld not be necessary, since you can always clone the connection and provide =
a new answer to it instead.<br>
_____________<br>Roman Shpount<br>
<br><br></div>

--0015175d67b4abfb7204bf2312d4--

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Date: Thu, 3 May 2012 11:48:33 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Final plea about SRTP
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On Wed, May 2, 2012 at 8:41 PM, Bernard Aboba <bernard_aboba@hotmail.com>wr=
ote:

> *A few years ago, the thought of turning on SRTP by default was a bit
> scary (mostly because of potential interop issues, not cost).  However,
> today turning it on by default =93just works=94 with minimal performance =
impact
> or other hassles (other than occasional interop gremlins).  By the time
> RTCWEB is widely deployed any argument against SRTP will probably be
> vestigial.*
>
> *Given this, it seems to me that the =93right thing=94 is for SRTP to be
> mandatory to implement and use, especially if SDES is available, so the
> likelihood of interoperability will be high.   *****
>
First of all, SDES-SRTP in combination with HTTP delivered application
provides no security at all. If you argue that you need SDES-SRTP for
interop, you might add support for RTP just as well.

Second, we just went through a couple of large scale SDES-SRTP deployments.
Virtually every device had some issues. Even though a lot of those issues
were not critical (like using RTP encryption routines to encrypt RTCP),
they were wide spread. What it indicates, is that even though SRTP is
widely implemented, it does not get nearly as much use as plain RTP. Even
though it is supported in the datasheet, it is often a datasheet support
only. Another indication of this is the state of libsrtp. It is widely used
to implement SRTP support in open source (and probably in quite a few
closed source) projects. At the same time it has issues that would had to
be addressed if it had any real life use. One examples is crashing on out
of order SRTCP packets which, even though addressed in libsrtp svn, is
present in release source code distribution that is commonly used. Another
example is random number generator that stops operating after a certain
number of calls which is not addressed anywhere. All of this can be
addressed given enough motivation, but this will take time.

Keep in mind, I am not saying that SRTP should not be a MUST implement
feature, and that plain RTP should be allowed from HTTPS sessions. What I
am saying that the value of SDES-SRTP for interop is questionable
(certainly less then value of plain RTP), and SRTP from HTTP session with
no identity provides no security, so it has little value.
_____________
Roman Shpount

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<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, May 2, 20=
12 at 8:41 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a href=3D"mailto:bernar=
d_aboba@hotmail.com" target=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</s=
pan> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div link=3D"blue" vlink=3D"purple" lang=3D"=
EN-US"><div><div><b><span style=3D"font-size:10.0pt;font-family:&quot;Tahom=
a&quot;,&quot;sans-serif&quot;;color:#1f497d">A few years ago, the thought =
of turning on SRTP by default was a bit scary (mostly because of potential =
interop issues, not cost).=A0 However, today turning it on by default =93ju=
st works=94 with minimal performance impact or other hassles (other than oc=
casional interop gremlins).=A0 By the time RTCWEB is widely deployed any ar=
gument against SRTP will probably be vestigial.</span></b><p class=3D"MsoNo=
rmal" style=3D"margin-bottom:12.0pt">
<b><span style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;san=
s-serif&quot;;color:#1f497d">Given this, it seems to me that the =93right t=
hing=94 is for SRTP to be mandatory to implement and use, especially if SDE=
S is available, so the likelihood of interoperability will be high. =A0=A0<=
/span></b><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></span></p>
</div></div></div></blockquote></div>First of all, SDES-SRTP in combination=
 with HTTP delivered application provides no security at all. If you argue =
that you need SDES-SRTP for interop, you might add support for RTP just as =
well.<br>
<br>Second, we just went through a couple of large scale SDES-SRTP deployme=
nts. Virtually every device had some issues. Even though a lot of those iss=
ues were not critical (like using RTP encryption routines to encrypt RTCP),=
 they were wide spread. What it indicates, is that even though SRTP is wide=
ly implemented, it does not get nearly as much use as plain RTP. Even thoug=
h it is supported in the datasheet, it is often a datasheet support only. A=
nother indication of this is the state of libsrtp. It is widely used to imp=
lement SRTP support in open source (and probably in quite a few closed sour=
ce) projects. At the same time it has issues that would had to be addressed=
 if it had any real life use. One examples is crashing on out of order SRTC=
P packets which, even though addressed in libsrtp svn, is present in releas=
e source code distribution that is commonly used. Another example is random=
 number generator that stops operating after a certain number of calls whic=
h is not addressed anywhere. All of this can be addressed given enough moti=
vation, but this will take time.<br>
<br>Keep in mind, I am not saying that SRTP should not be a MUST implement =
feature, and that plain RTP should be allowed from HTTPS sessions. What I a=
m saying that the value of SDES-SRTP for interop is questionable (certainly=
 less then value of plain RTP), and SRTP from HTTP session with no identity=
 provides no security, so it has little value.<br>
_____________<br>Roman Shpount<br>
<br></div>

--047d7b1605c97e96fb04bf23bcf9--

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From: jesse <chat2jesse@gmail.com>
To: Bernard Aboba <bernard_aboba@hotmail.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Final plea about SRTP
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On May 2, 2012 5:41 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:
>
> At IETF 83,  Dan showed some slides relating to SRTP/RTP gateways that
seemed to cover the =93extreme legacy=94 case quite well.
>
> In those slides, the =93RTCWEB=94 components were still talking SRTP, so =
they
could be said to be compliant with a =93mandatory to implement, mandatory t=
o
use=94 posture.   Yet, the circa 2005 IP phone still got to receive and sen=
d
RTP.
>
> Since the =93extreme legacy=94 cases are solvable via gateways,

First of all, solvable via gateway is a clumsy solution. For most small end
users, support and maintain gateway for srtp takes extra certain effort.
Very few VoIP systems make srtp default unless the whole system is provided
by big vendors.

Second, those are not extreme legacy. There are tons of traditional VoIP
systems there.

rtcweb is proposing a high standard, but "no legacy system should be left
behind" if the objective is to make rtcweb accessible everywhere and
everydevice.

> and the performance impact and cost on virtually any new hardware
(PC/tablet/mobile) is minimal, it seems to me that making SRTP mandatory to
implement and use has little downside.
>
> A few years ago, the thought of turning on SRTP by default was a bit
scary (mostly because of potential interop issues, not cost).  However,
today turning it on by default =93just works=94 with minimal performance im=
pact
or other hassles (other than occasional interop gremlins).  By the time
RTCWEB is widely deployed any argument against SRTP will probably be
vestigial.
>
> Given this, it seems to me that the =93right thing=94 is for SRTP to be
mandatory to implement and use, especially if SDES is available, so the
likelihood of interoperability will be high.
>
> On Wed, May 2, 2012 at 5:32 PM, Cullen Jennings <fluffy@iii.ca> wrote:
>
>
> Roman,
>
> One comment on this - I think people understand there could be services
with no security requirements that could run over RTP, and HTTP, with no
identity. But we need to have a secure solution for some other services.
The questions is once you have a secure solution, what is the incentive to
also support an insecure solution - so far no one has come up with a super
compelling story about dealing with the bid down and I suspect that lots of
people did not view the overhead of running the secure version as all that
high. I suspect that is part of why the decisions went the way it did -
basically people agreed we needed a secure solution, and when they
considered also having an insecure solution, they saw lots of complications
of doing both and not much gain in the insecure solution over the secure
solution.
>
> Cullen
>
>
>
>
> On May 2, 2012, at 10:03 AM, Roman Shpount wrote:
>
> > I know there was a consensus call on this list that SRTP shall be used
for all the calls in WebRTC, but I still do not understand the
justification for this requirement for WebRTC applications delivered over
HTTP with no identity. For such scenarios SRTP (even DTLS-SRTP) serves
almost no purpose. If application is delivered over HTTP attacker can spoof
the entire web site. It is trivial if the attacker is on the communications
path. If attacker is seating in the airport using the same network, it can
put itself on the communications path using arp cache poisoning. Once the
web site is spoofed, any type of man in the middle attack can be
implemented. If DTLS-SRTP is used user can detect the attack by checking
the key signature, but in reality very few people will do this.
> >
> > The main argument to require SRTP everywhere was that it does not break
anything. But neither would naming all the API methods in High Elfish.
Either requirement does not break things, but make working with WebRTC
harder then it should. At the same time both of those requirements are
completely unjustified.
> >
> > Furthermore, assumption on this list that most of the WebRTC use would
be peer-to-peer communications between browsers with all the rest of the
communication modes, such as calling automated services or PSTN being
insignificant. I simply do not agree to this point of view. I expect that
communication with automated services, such as video greeting cards or
voice blogging, would be a significant portion of WebRTC user base. If such
automated service is deployed as a plain HTTP web site, it should be able
to communicate with web browsers using RTP. SRTP in such case would serve
no purpose.
> >
> > Finally, requiring secure communications for everything is going
against the way most of the web works. Most of it is not secured and only
requires secure communications when secure (HTTPS) web site is accessed. I
think it should be the same for WebRTC, with DTLS-SRTP required when
connected to HTTPS web site and plain RTP allowed when connected to plan
HTTP.
> > _____________
> > Roman Shpount
>
> > _______________________________________________
> > 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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<p><br>
On May 2, 2012 5:41 PM, &quot;Bernard Aboba&quot; &lt;<a href=3D"mailto:ber=
nard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<br>
&gt;<br>
&gt; At IETF 83, =A0Dan showed some slides relating to SRTP/RTP gateways th=
at seemed to cover the =93extreme legacy=94 case quite well.<br>
&gt;<br>
&gt; In those slides, the =93RTCWEB=94 components were still talking SRTP, =
so they could be said to be compliant with a =93mandatory to implement, man=
datory to use=94 posture.=A0=A0 Yet, the circa 2005 IP phone still got to r=
eceive and send RTP. =A0<br>

&gt;<br>
&gt; Since the =93extreme legacy=94 cases are solvable via gateways, </p>
<p>First of all, solvable via gateway is a clumsy solution. For most small =
end users, support and maintain gateway for srtp takes extra certain effort=
. Very few VoIP systems make srtp default unless the whole system is provid=
ed by big vendors. </p>

<p>Second, those are not extreme legacy. There are tons of traditional VoIP=
 systems there.</p>
<p> rtcweb is proposing a high standard, but &quot;no legacy system should =
be left behind&quot; if the objective is to make rtcweb accessible everywhe=
re and everydevice.</p>
<p>&gt; and the performance impact and cost on virtually any new hardware (=
PC/tablet/mobile) is minimal, it seems to me that making SRTP mandatory to =
implement and use has little downside.<br>
&gt;<br>
&gt; A few years ago, the thought of turning on SRTP by default was a bit s=
cary (mostly because of potential interop issues, not cost).=A0 However, to=
day turning it on by default =93just works=94 with minimal performance impa=
ct or other hassles (other than occasional interop gremlins).=A0 By the tim=
e RTCWEB is widely deployed any argument against SRTP will probably be vest=
igial.<br>

&gt;<br>
&gt; Given this, it seems to me that the =93right thing=94 is for SRTP to b=
e mandatory to implement and use, especially if SDES is available, so the l=
ikelihood of interoperability will be high. =A0=A0<br>
&gt;<br>
&gt; On Wed, May 2, 2012 at 5:32 PM, Cullen Jennings &lt;<a href=3D"mailto:=
fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt;<br>
&gt;<br>
&gt; Roman,<br>
&gt;<br>
&gt; One comment on this - I think people understand there could be service=
s with no security requirements that could run over RTP, and HTTP, with no =
identity. But we need to have a secure solution for some other services. Th=
e questions is once you have a secure solution, what is the incentive to al=
so support an insecure solution - so far no one has come up with a super co=
mpelling story about dealing with the bid down and I suspect that lots of p=
eople did not view the overhead of running the secure version as all that h=
igh. I suspect that is part of why the decisions went the way it did - basi=
cally people agreed we needed a secure solution, and when they considered a=
lso having an insecure solution, they saw lots of complications of doing bo=
th and not much gain in the insecure solution over the secure solution.<br>

&gt;<br>
&gt; Cullen<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; On May 2, 2012, at 10:03 AM, Roman Shpount wrote:<br>
&gt;<br>
&gt; &gt; I know there was a consensus call on this list that SRTP shall be=
 used for all the calls in WebRTC, but I still do not understand the justif=
ication for this requirement for WebRTC applications delivered over HTTP wi=
th no identity. For such scenarios SRTP (even DTLS-SRTP) serves almost no p=
urpose. If application is delivered over HTTP attacker can spoof the entire=
 web site. It is trivial if the attacker is on the communications path. If =
attacker is seating in the airport using the same network, it can put itsel=
f on the communications path using arp cache poisoning. Once the web site i=
s spoofed, any type of man in the middle attack can be implemented. If DTLS=
-SRTP is used user can detect the attack by checking the key signature, but=
 in reality very few people will do this.<br>

&gt; &gt;<br>
&gt; &gt; The main argument to require SRTP everywhere was that it does not=
 break anything. But neither would naming all the API methods in High Elfis=
h. Either requirement does not break things, but make working with WebRTC h=
arder then it should. At the same time both of those requirements are compl=
etely unjustified.<br>

&gt; &gt;<br>
&gt; &gt; Furthermore, assumption on this list that most of the WebRTC use =
would be peer-to-peer communications between browsers with all the rest of =
the communication modes, such as calling automated services or PSTN being i=
nsignificant. I simply do not agree to this point of view. I expect that co=
mmunication with automated services, such as video greeting cards or voice =
blogging, would be a significant portion of WebRTC user base. If such autom=
ated service is deployed as a plain HTTP web site, it should be able to com=
municate with web browsers using RTP. SRTP in such case would serve no purp=
ose.<br>

&gt; &gt;<br>
&gt; &gt; Finally, requiring secure communications for everything is going =
against the way most of the web works. Most of it is not secured and only r=
equires secure communications when secure (HTTPS) web site is accessed. I t=
hink it should be the same for WebRTC, with DTLS-SRTP required when connect=
ed to HTTPS web site and plain RTP allowed when connected to plan HTTP.<br>

&gt; &gt; _____________<br>
&gt; &gt; Roman Shpount<br>
&gt;<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt;<br>
&gt; =A0<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>
</p>

--e89a8ff1cafa92412404bf2467d5--

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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 5/3/2012 11:01 AM, Roman Shpount wrote:
>
> On Thu, May 3, 2012 at 2:02 AM, Christer Holmberg
> <christer.holmberg@ericsson.com <mailto:christer.holmberg@ericsson.com>>
> wrote:
>
>     You are absolutely right. I still question whether we need to
>     support parallel forking. If you say it occurs often, I can't argue
>     against you, even though my own experience is different :)
>
>
> You do end up with parallel forking when you allow registrations from
> multiple devices. This is a policy option, but once you enable this,
> parallel forking or B2BUA are the only options.

Correct; I brought this up multiple times last year, when we had one of 
the original discussions of cloning.

If I call scotty@gmail.com, and he has a phone, and a tablet, and a 
computer, and a star trek communicator all "attached" to google's JSEP 
signaling proxy, it's going to want to fork the offer to all of them. 
So I really believe parallel forking is needed for very basic use-cases. 
  And both Scotty (in the Away Team) and his assistant (sitting at the 
computer) might accept.  If his assistant accepts first, Scotty is out 
of luck unless some parallel forking is supported and PeerConnection 
cloning is done.  That doesn't mean you must render or encode data for 
all clones simultaneously, though that's a nice option and mirrors 
normal requirements for mesh conferencing.  You can send the same data 
to both (mesh), or possibly act as a conference bridge instead of a full 
mesh conference (ah-hoc bridged conference).

Another usecase I bandied about was "hanging" cloned offers being 
provided to new participants in a partial-mesh conference.  Think Second 
Life, or a virtual conference with positional locality (it doesn't have 
to be visual Virtual Reality ala Second Life; it can be abstract - but 
it's easiest to think of in a Second Life or MMORPG type of setting). 
I.e. the signaling server holds a cloned copy of your invite, and when 
someone new comes into hearing range of you it forwards your (cloned) 
OFFER to them (delayed parallel forking, if you will).  Otherwise, it 
would need a protocol to let the server tell you to generate a new offer 
that it could then forward, adding roughly 1 client-server-RTT to setup 
time, plus requiring extra application protocols and logic.

>     (Another option, if we want to support parallel forking, would of
>     course be to create a completely new PeerConnection, with a *new*
>     local IP address:port. But, you would of course then have to provide
>     that information to the remote peer.)
>
>
> This would probably do absolutely nothing to SIP interop. And if we do
> not need SIP interop we can live without forking (at the cost of being
> slightly less efficient in some call setup scenarios).

Ignoring SIP interop, this might work as an equivalent to cloning, but 
requiring an extra O/A exchange (and adding considerable chance of 
clipping on a second fork being activated).

>
>     I have no strong feeling on whether we want to do cloning, but do
>     people agree that, for a given PeerConnection, we only need to
>     support a single remote peer (which can be modified, though)?
>
>
> I think if we do cloning we will only need to support a single remote
> peer per PeerConnection. Strictly speaking even updates due to
> provisional responses would not be necessary, since you can always clone
> the connection and provide a new answer to it instead.

I believe that's correct.

-- 
Randell Jesup
randell-ietf@jesup.org

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Correct; I brought this up multiple times last year, when we had one of the=
 original discussions of cloning.

[RRR] Did you also think about sub-conferencing as a part of the main confe=
rencing as well?

From iesg-secretary@ietf.org  Thu May  3 11:28:46 2012
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Subject: [rtcweb] RTCWEB WG Interim Meeting: June 12-13, 2012
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The RTCWEB working group will be holding an interim meeting in
Stockholm, Sweden on June 12th and 13th, 2012, hosted by Ericsson.
Remote attendance by Webex will be supported. More information on the
location and remote meeting details will be supplied on the RTCWEB
mailing list. Note that there is a related W3C WEBRTC meeting on June
11th, for which public attendance is possible; please contact the
chairs of that group (Harald Alvestrand and Stefan H=C3=A5kansson) for
details.

From christer.holmberg@ericsson.com  Thu May  3 11:55:51 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 3 May 2012 20:51:23 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>>     (Another option, if we want to support parallel forking, would of
>>     course be to create a completely new PeerConnection, with a *new*
>>     local IP address:port. But, you would of course then have to provide
>>     that information to the remote peer.)
>>
>>
>> This would probably do absolutely nothing to SIP interop. And if we do
>> not need SIP interop we can live without forking (at the cost of being
>> slightly less efficient in some call setup scenarios).
>
> Ignoring SIP interop, this might work as an equivalent to cloning, but
> requiring an extra O/A exchange (and adding considerable chance of
> clipping on a second fork being activated).

Assuming you have to perform the ICE procedures anyway, I don't think it wo=
uld cause clipping. It probably would cause some additinoal call setup dela=
y, though, as you need to perform that extra O/A exchange before the ICE pr=
ocedures can start.


>>     I have no strong feeling on whether we want to do cloning, but do
>>     people agree that, for a given PeerConnection, we only need to
>>     support a single remote peer (which can be modified, though)?
>>
>>
>> I think if we do cloning we will only need to support a single remote
>> peer per PeerConnection. Strictly speaking even updates due to
>> provisional responses would not be necessary, since you can always clone
>> the connection and provide a new answer to it instead.
>
> I believe that's correct.

Sure. But, that would also mean that you would have to clone the PeerConnec=
tion mid-session, if the remote entity changes.

Regards,

Christer=

From roman@telurix.com  Thu May  3 12:09:37 2012
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--047d7b3392e95d344504bf268b0a
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On Thu, May 3, 2012 at 2:51 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Sure. But, that would also mean that you would have to clone the
> PeerConnection mid-session, if the remote entity changes.
>

What I meant was that you do not need ability to update remote destination
multiple times per single offer. You should still be able to initiate
another O/A exchange on the same peer connection and change the remote
party this way.
_____________
Roman Shpount

--047d7b3392e95d344504bf268b0a
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<div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, May 3, 20=
12 at 2:51 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:ch=
rister.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@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">
Sure. But, that would also mean that you would have to clone the PeerConnec=
tion mid-session, if the remote entity changes.<br></blockquote></div><br>W=
hat I meant was that you do not need ability to update remote destination m=
ultiple times per single offer. You should still be able to initiate anothe=
r O/A exchange on the same peer connection and change the remote party this=
 way. <br>
_____________<br>Roman Shpount<br>
<br><br></div>

--047d7b3392e95d344504bf268b0a--

From randell-ietf@jesup.org  Thu May  3 12:55:17 2012
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 5/3/2012 2:51 PM, Christer Holmberg wrote:
>>>      (Another option, if we want to support parallel forking, would of
>>>      course be to create a completely new PeerConnection, with a *new*
>>>      local IP address:port. But, you would of course then have to provide
>>>      that information to the remote peer.)
>>>
>>>
>>> This would probably do absolutely nothing to SIP interop. And if we do
>>> not need SIP interop we can live without forking (at the cost of being
>>> slightly less efficient in some call setup scenarios).
>> Ignoring SIP interop, this might work as an equivalent to cloning, but
>> requiring an extra O/A exchange (and adding considerable chance of
>> clipping on a second fork being activated).
> Assuming you have to perform the ICE procedures anyway, I don't think it would cause clipping. It probably would cause some additinoal call setup delay, though, as you need to perform that extra O/A exchange before the ICE procedures can start.

Call setup delay causes clipping, unless you in some manner use feedback 
to the answerer to hold them off from talking.  The classic problem you 
probably all know is "pickup receiver, say "hello" - if call setup takes 
(or media is clipped by) more than a couple hundred ms, the clipping is 
audible to the caller.  And it gets worse (100ms or perhaps even less) 
for "click-to-answer".  The only thing to help you there is possible 
visual indication that it's not connected yet to encourage the answerer 
to wait before talking, and in some uses that's not possible or isn't 
sufficiently obvious.  (audio-only connections, especially in a context 
where you're paying attention to other things as well (game, VR 
conference, etc)).

-- 
Randell Jesup
randell-ietf@jesup.org


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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericsson.se> <4FA2E264.4040207@jesup.org>
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From: Roman Shpount <roman@telurix.com>
To: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On Thu, May 3, 2012 at 3:54 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

>
> Call setup delay causes clipping, unless you in some manner use feedback
> to the answerer to hold them off from talking.  The classic problem you
> probably all know is "pickup receiver, say "hello" - if call setup takes
> (or media is clipped by) more than a couple hundred ms, the clipping is
> audible to the caller.  And it gets worse (100ms or perhaps even less) for
> "click-to-answer".  The only thing to help you there is possible visual
> indication that it's not connected yet to encourage the answerer to wait
> before talking, and in some uses that's not possible or isn't sufficiently
> obvious.  (audio-only connections, especially in a context where you're
> paying attention to other things as well (game, VR conference, etc)).
>

Unfortunately this is unavoidable with ICE. Once you have ICE setup enabled
you normally end up with at least a RTT delay. If this is a call from east
to west coast in US, this typically means that about 100 ms worth of media
is clipped (pretty much means no initial hello). If you add DTLS setup on
top of this, you end up with more then half a second worth of audio
clipped, so this typically means people saying hello several times to see
if there is a person on the other side. All of this for the calls in
continental US. If you are talking to somebody over a satellite link,
situation is much worse. The way I've seen dealt with this was playing some
sort of audio and media to the answering party until call is actually
connected to prevent them from speaking.
_____________
Roman Shpount

--047d7b3392e937c06604bf279e6a
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Thu, May 3, 2012 a=
t 3:54 PM, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ie=
tf@jesup.org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex">
<div class=3D"im"><br></div>
Call setup delay causes clipping, unless you in some manner use feedback to=
 the answerer to hold them off from talking. =A0The classic problem you pro=
bably all know is &quot;pickup receiver, say &quot;hello&quot; - if call se=
tup takes (or media is clipped by) more than a couple hundred ms, the clipp=
ing is audible to the caller. =A0And it gets worse (100ms or perhaps even l=
ess) for &quot;click-to-answer&quot;. =A0The only thing to help you there i=
s possible visual indication that it&#39;s not connected yet to encourage t=
he answerer to wait before talking, and in some uses that&#39;s not possibl=
e or isn&#39;t sufficiently obvious. =A0(audio-only connections, especially=
 in a context where you&#39;re paying attention to other things as well (ga=
me, VR conference, etc)).<span class=3D"HOEnZb"><font color=3D"#888888"></f=
ont></span><br>
</blockquote></div><br>Unfortunately this is unavoidable with ICE. Once you=
 have ICE setup enabled you normally end up with at least a RTT delay. If t=
his is a call from east to west coast in US, this typically means that abou=
t 100 ms worth of media is clipped (pretty much means no initial hello). If=
 you add DTLS setup on top of this, you end up with more then half a second=
 worth of audio clipped, so this typically means people saying hello severa=
l times to see if there is a person on the other side. All of this for the =
calls in continental US. If you are talking to somebody over a satellite li=
nk, situation is much worse. The way I&#39;ve seen dealt with this was play=
ing some sort of audio and media to the answering party until call is actua=
lly connected to prevent them from speaking.<br>
_____________<br>Roman Shpount<br>
<br><br></div>

--047d7b3392e937c06604bf279e6a--

From christer.holmberg@ericsson.com  Thu May  3 13:31:54 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>, Randell Jesup <randell-ietf@jesup.org>
Date: Thu, 3 May 2012 22:29:04 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericsson.se> <4FA2E264.4040207@jesup.org>, <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

You could also choose not to alert the remote user until the ICE procedures=
 have finished - more or less using ICE with preconditions.

Of course, that is going to trigger actions in STUN/TURN servers, even if t=
he called party won't accept the call, but at least from a user experience =
perspective that is still better than lifting the hook, and having to wait =
for whatever-time-it-takes-for-ICE-to-finish seconds before one can start t=
o talk.

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Roman =
Shpount [roman@telurix.com]
Sent: Thursday, May 03, 2012 11:26 PM
To: Randell Jesup
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP

On Thu, May 3, 2012 at 3:54 PM, Randell Jesup <randell-ietf@jesup.org<mailt=
o:randell-ietf@jesup.org>> wrote:

Call setup delay causes clipping, unless you in some manner use feedback to=
 the answerer to hold them off from talking.  The classic problem you proba=
bly all know is "pickup receiver, say "hello" - if call setup takes (or med=
ia is clipped by) more than a couple hundred ms, the clipping is audible to=
 the caller.  And it gets worse (100ms or perhaps even less) for "click-to-=
answer".  The only thing to help you there is possible visual indication th=
at it's not connected yet to encourage the answerer to wait before talking,=
 and in some uses that's not possible or isn't sufficiently obvious.  (audi=
o-only connections, especially in a context where you're paying attention t=
o other things as well (game, VR conference, etc)).

Unfortunately this is unavoidable with ICE. Once you have ICE setup enabled=
 you normally end up with at least a RTT delay. If this is a call from east=
 to west coast in US, this typically means that about 100 ms worth of media=
 is clipped (pretty much means no initial hello). If you add DTLS setup on =
top of this, you end up with more then half a second worth of audio clipped=
, so this typically means people saying hello several times to see if there=
 is a person on the other side. All of this for the calls in continental US=
. If you are talking to somebody over a satellite link, situation is much w=
orse. The way I've seen dealt with this was playing some sort of audio and =
media to the answering party until call is actually connected to prevent th=
em from speaking.
_____________
Roman Shpount



From roman@telurix.com  Thu May  3 13:34:24 2012
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericsson.se> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se>
Date: Thu, 3 May 2012 16:34:21 -0400
Message-ID: <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--047d7b1605c998e32a04bf27ba7f
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On Thu, May 3, 2012 at 4:29 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> You could also choose not to alert the remote user until the ICE
> procedures have finished - more or less using ICE with preconditions.
>
> Of course, that is going to trigger actions in STUN/TURN servers, even if
> the called party won't accept the call, but at least from a user experience
> perspective that is still better than lifting the hook, and having to wait
> for whatever-time-it-takes-for-ICE-to-finish seconds before one can start
> to talk.
>
>
This also has a downside of disclosing user's IP to the caller without the
user confirmation. For a lot of applications this can be serious security
flaw.
_____________
Roman Shpount

--047d7b1605c998e32a04bf27ba7f
Content-Type: text/html; charset=ISO-8859-1
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<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Thu, May 3, 2012 a=
t 4:29 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christ=
er.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@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">You could also choose not to alert the remot=
e user until the ICE procedures have finished - more or less using ICE with=
 preconditions.<br>

<br>
Of course, that is going to trigger actions in STUN/TURN servers, even if t=
he called party won&#39;t accept the call, but at least from a user experie=
nce perspective that is still better than lifting the hook, and having to w=
ait for whatever-time-it-takes-for-ICE-to-finish seconds before one can sta=
rt to talk.<br>

<br></blockquote><div><br>This also has a downside of disclosing user&#39;s=
 IP to the caller without the user confirmation. For a lot of applications =
this can be serious security flaw. <br></div></div>_____________<br>Roman S=
hpount<br>

<br></div>

--047d7b1605c998e32a04bf27ba7f--

From christer.holmberg@ericsson.com  Thu May  3 13:50:08 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Date: Thu, 3 May 2012 22:50:05 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericsson.se> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se>, <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>>You could also choose not to alert the remote user until the ICE procedur=
es have finished - more or less using ICE with preconditions.
>>
>> Of course, that is going to trigger actions in STUN/TURN servers, even i=
f the called party won't accept the call, but at least from a user=20
>> experience perspective that is still better than lifting the hook, and h=
aving to wait for whatever-time-it-takes-for-ICE-to-finish seconds before o=
ne can start to talk.
>
> This also has a downside of disclosing user's IP to the caller without th=
e user confirmation. For a lot of applications this can be serious security=
 flaw.=20

The client can still log when ICE procedures occur.

Because, even if I wait until your phone starts to ring, most likely I woul=
d still get your IP address without user confirmation (speaking in SIP term=
s, phones normally don't ask for user permission before they send 18x with =
SDP), eventhough you would easier be made aware of that it happens.

Another alternative is of course to do ICE with an SBC, and/or having an SB=
C doing ICE on behalf of you :)

Regards,

Christer=

From roman@telurix.com  Thu May  3 14:01:54 2012
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericsson.se> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001339@ESESSCMS0356.eemea.ericsson.se>
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From: Roman Shpount <roman@telurix.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--e89a8ff24a9bf60e3e04bf281c1b
Content-Type: text/plain; charset=ISO-8859-1

On Thu, May 3, 2012 at 4:50 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>You could also choose not to alert the remote user until the ICE
> procedures have finished - more or less using ICE with preconditions.
> >>
> >> Of course, that is going to trigger actions in STUN/TURN servers, even
> if the called party won't accept the call, but at least from a user
> >> experience perspective that is still better than lifting the hook, and
> having to wait for whatever-time-it-takes-for-ICE-to-finish seconds before
> one can start to talk.
> >
> > This also has a downside of disclosing user's IP to the caller without
> the user confirmation. For a lot of applications this can be serious
> security flaw.
>
> The client can still log when ICE procedures occur.
>
> Because, even if I wait until your phone starts to ring, most likely I
> would still get your IP address without user confirmation (speaking in SIP
> terms, phones normally don't ask for user permission before they send 18x
> with SDP), eventhough you would easier be made aware of that it happens.
>
> Another alternative is of course to do ICE with an SBC, and/or having an
> SBC doing ICE on behalf of you :)
>

This is true for SIP but was as far as I know was specifically designed
around in WebRTC. WebRTC signaling server acts as B2BUA so any type of
ringing notification goes through the web server and does not need to carry
any type of client address information. The client address information is
only provided when SDP answer is sent or ICE negotiation is started.
_____________
Roman Shpount

--e89a8ff24a9bf60e3e04bf281c1b
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<div class=3D"gmail_extra"><div class=3D"gmail_quote">On Thu, May 3, 2012 a=
t 4:50 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christ=
er.holmberg@ericsson.com" target=3D"_blank">christer.holmberg@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">Hi,<br>
<div class=3D"im"><br>
&gt;&gt;You could also choose not to alert the remote user until the ICE pr=
ocedures have finished - more or less using ICE with preconditions.<br>
&gt;&gt;<br>
&gt;&gt; Of course, that is going to trigger actions in STUN/TURN servers, =
even if the called party won&#39;t accept the call, but at least from a use=
r<br>
&gt;&gt; experience perspective that is still better than lifting the hook,=
 and having to wait for whatever-time-it-takes-for-ICE-to-finish seconds be=
fore one can start to talk.<br>
&gt;<br>
&gt; This also has a downside of disclosing user&#39;s IP to the caller wit=
hout the user confirmation. For a lot of applications this can be serious s=
ecurity flaw.<br>
<br>
</div>The client can still log when ICE procedures occur.<br>
<br>
Because, even if I wait until your phone starts to ring, most likely I woul=
d still get your IP address without user confirmation (speaking in SIP term=
s, phones normally don&#39;t ask for user permission before they send 18x w=
ith SDP), eventhough you would easier be made aware of that it happens.<br>

<br>
Another alternative is of course to do ICE with an SBC, and/or having an SB=
C doing ICE on behalf of you :)<br></blockquote></div><br>This is true for =
SIP but was as far as I know was specifically designed around in WebRTC. We=
bRTC signaling server acts as B2BUA so any type of ringing notification goe=
s through the web server and does not need to carry any type of client addr=
ess information. The client address information is only provided when SDP a=
nswer is sent or ICE negotiation is started. <br>
_____________<br>Roman Shpount<br>
<br><br></div>

--e89a8ff24a9bf60e3e04bf281c1b--

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Roman Shpount <roman@telurix.com>
Date: Thu, 3 May 2012 23:16:38 +0200
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Hi,

>>>>You could also choose not to alert the remote user until the ICE proced=
ures have finished - more or less using ICE with preconditions.
>>>>
>>>> Of course, that is going to trigger actions in STUN/TURN servers, even=
 if the called party won't accept the call, but at least from a user
>>>> experience perspective that is still better than lifting the hook, and=
 having to wait for whatever-time-it-takes-for-ICE-to-finish seconds before=
 one can start to talk.
>>>
>>> This also has a downside of disclosing user's IP to the caller without =
the user confirmation. For a lot of applications this can be serious securi=
ty flaw.
>>
>> The client can still log when ICE procedures occur.
>>
>> Because, even if I wait until your phone starts to ring, most likely I w=
ould still get your IP address without user confirmation (speaking in SIP t=
erms, phones normally don't ask for user permission before they send 18x wi=
th SDP), eventhough you would easier be made aware of that it happens.
>
> Another alternative is of course to do ICE with an SBC, and/or having an =
SBC doing ICE on behalf of you :)
>
>
> This is true for SIP but was as far as I know was specifically designed a=
round in WebRTC. WebRTC signaling server acts as B2BUA so any type of ringi=
ng notification goes through the web server and does not need to carry any =
type of client address information. The client address information is only =
provided=20
> when SDP answer is sent or ICE negotiation is started.=20

Assuming you are only going to make user confirmation (read: accept calls) =
in cases where you absolutely sure that the caller isn't someone trying to =
get your IP address.

But, never the less, having a solution where I first have to give user conf=
irmation, and then wait until I can start to talk, is probably something mo=
st people want to avoid. Depending upon, of course, how long the waiting ti=
me is.

Regards,

Christer=

From dean.willis@softarmor.com  Thu May  3 22:11:52 2012
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From: Dean Willis <dean.willis@softarmor.com>
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Subject: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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I asked a colleague who develops serious enterprise video stuff:

> Here's a question you can perhaps help me with. One of the big fights =
in the IETF right now is which codec to require as the "must implement" =
for real-time web communications. The leading candidates are H.264 and =
VP8, although I've proposed H.263 as a cheaper-to-license starting =
point.
> =20
> How much of an issue has codec licensing been for you, and what would =
you prefer as the "standard" codec?
> =20


And the (somewhat surprising to me) response (which I've expurgated a =
bit as I'm passing it on without explicit permission)  was:

> I believe the battle is over and H.264 has won. The question is =
ultimately where has the market gone in terms of playback mechanisms. =
The answer:
> =20
> Flash: H.264 and (maybe)VP8
> Adobe Adaptive HDS: H.264
> QuickTime: H.264
> Apple Adaptive HLS =96 H.264
> Smooth Streaming: VC-1 and H.264
> DASH =96 All implementations are H.264
> Android Cell Phones: H.264
> Cell Phone hardware: H.264
> HTML5 on IE: H.264
> HTML5 on Safari: H.264
> HTML5 on Firefox: VP8 but recently announced support for H.264 see =
http://www.theregister.co.uk/2012/03/19/firefox_sucks_on_h264/
> HTML5 on Chrome: H.264 and VP8
> Video Conferencing World: H.264 supported universally
> Broadcast World =96 Moving from MPEG2 to H.264
> Every Set Top Box in the world =96 MPEG2 and H.264
> =20
> Once the chip manufacturers started building H.264 into the hardware, =
it was game over. Even though Google was the big VP8 =
developer/supporter, it had to support H.264 in Chrome and Android =
phones since that=92s all they had to work with on the phones.
> =20
> Pretty obvious pattern.
> =20
> Codec licensing is largely an issue of the past for us. We do license =
the H.264 encoders, but it is a miniscule part of our material cost. The =
decoder license is on a per-machine basis. Every PC/phone/pad that ships =
today has been licensed for H.264(and AAC) by the manufacturer of the =
hardware and/or the OS provider. Of course the main complaint on =
licensing of H.264 is from the content producers which is where the =
MPEG-LA members want to make money. I don=92t have much comment on that, =
but in our customer base =96 enterprise =96 I have never once heard it =
raised as an issue during a buying decision.
> =20
> Also =96 if VP8 really got popular it is likely that the MPEG-LA =
patent holders would come after them. Everything I have read indicates =
that it infringes lots of patents and is basically the same as H.264 in =
many ways. That is what happened to VC-1 (Windows Media Video). The =
patent holders won=92t bother unless is appears to be a real competitor.
> =20
> The html5 people originally wanted to insist on a royalty free codec =
and gave up based on the reality was that H.264 is the only essential =
codec. If IETF insists on VP8 or 363 as a =93must implement=94 they will =
be ignored. It would be a huge deviation from their general principal of =
consensus and following the market. I cannot conceive of how it could =
happen.
> =20
> We really like a standard codec since it eliminates one area of =
confusion.=20


This seems like a fairly compelling argument for H.264 as a baseline =
must-implement in RTCWeb. I'm thinking of changing my position.

--
Dean


--Apple-Mail-11--828668167
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	charset=windows-1252

<html><head><base href=3D"x-msg://155/"></head><body style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div><br></div><div>I asked a colleague who =
develops serious enterprise video stuff:</div><div><br></div><blockquote =
type=3D"cite"><span class=3D"Apple-style-span" style=3D"border-collapse: =
separate; font-family: Helvetica; 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-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div class=3D"WordSection1" =
style=3D"page: WordSection1; "><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">Here's a question you can =
perhaps help me with. One of the big fights in the IETF right now is =
which codec to require as the "must implement" for real-time web =
communications. The leading candidates are H.264 and VP8, although I've =
proposed H.263 as a cheaper-to-license starting =
point.<o:p></o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div><div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; ">How much of an issue has =
codec licensing been for you, and what would you prefer as the =
"standard" codec?<o:p></o:p></div></div><div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; =
"><o:p>&nbsp;</o:p></div></div></div></div></span></blockquote><div><br><d=
iv><br></div><div>And the (somewhat surprising to me) response (which =
I've expurgated a bit as I'm passing it on without explicit permission) =
&nbsp;was:</div><div><br></div><blockquote type=3D"cite"><span =
class=3D"Apple-style-span" style=3D"color: rgb(31, 73, 125); =
font-family: Calibri, sans-serif; font-size: 15px; ">I believe the =
battle is over and H.264 has won. The question is ultimately where has =
the market gone in terms of playback mechanisms. The answer:</span><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
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-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div lang=3D"EN-US" link=3D"blue" =
vlink=3D"purple"><div class=3D"WordSection1" style=3D"page: =
WordSection1; "><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: =
'Times New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Flash: H.264 and (maybe)VP8<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Adobe Adaptive HDS: =
H.264<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">QuickTime: H.264<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Apple =
Adaptive HLS =96 H.264<o:p></o:p></span></div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">Smooth Streaming: VC-1 and =
H.264<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">DASH =
=96 All implementations are H.264<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Android Cell Phones: =
H.264<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Cell =
Phone hardware: H.264<o:p></o:p></span></div><div style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); ">HTML5 on IE: H.264<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">HTML5 on Safari: =
H.264<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">HTML5 =
on Firefox: VP8 but recently announced support for H.264 see&nbsp;<a =
href=3D"http://www.theregister.co.uk/2012/03/19/firefox_sucks_on_h264/" =
style=3D"color: blue; text-decoration: underline; =
">http://www.theregister.co.uk/2012/03/19/firefox_sucks_on_h264/</a><o:p><=
/o:p></span></div><div style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 12pt; font-family: =
'Times New Roman', serif; "><span style=3D"font-size: 11pt; font-family: =
Calibri, sans-serif; color: rgb(31, 73, 125); ">HTML5 on Chrome: H.264 =
and VP8<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Video =
Conferencing World: H.264 supported =
universally<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Broadcast World =96 Moving from MPEG2 to H.264<br>Every Set Top Box in =
the world =96 MPEG2 and H.264<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Once =
the chip manufacturers started building H.264 into the hardware, it was =
game over. Even though Google was the big VP8 developer/supporter, it =
had to support H.264 in Chrome and Android phones since that=92s all =
they had to work with on the phones.<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
">Pretty obvious pattern.<o:p></o:p></span></div><div style=3D"margin-top:=
 0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 12pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 11pt; font-family: Calibri, sans-serif; color: =
rgb(31, 73, 125); "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 12pt; font-family: 'Times New =
Roman', serif; "><span style=3D"font-size: 11pt; font-family: Calibri, =
sans-serif; color: rgb(31, 73, 125); ">Codec licensing is largely an =
issue of the past for us. We do license the H.264 encoders, but it is a =
miniscule part of our material cost. The decoder license is on a =
per-machine basis. Every PC/phone/pad that ships today has been licensed =
for H.264(and AAC) by the manufacturer of the hardware and/or the OS =
provider. Of course the main complaint on licensing of H.264 is from the =
content producers which is where the MPEG-LA members want to make money. =
I don=92t have much comment on that, but in our customer base =96 =
enterprise =96 I have never once heard it raised as an issue during a =
buying decision.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">Also =
=96 if VP8 really got popular it is likely that the MPEG-LA patent =
holders would come after them. Everything I have read indicates that it =
infringes lots of patents and is basically the same as H.264 in many =
ways. That is what happened to VC-1 (Windows Media Video). The patent =
holders won=92t bother unless is appears to be a real =
competitor.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">The =
html5 people originally wanted to insist on a royalty free codec and =
gave up based on the reality was that H.264 is the only essential codec. =
If IETF insists on VP8 or 363 as a =93must implement=94 they will be =
ignored. It would be a huge deviation from their general principal of =
consensus and following the market. I cannot conceive of how it could =
happen.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); =
"><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
12pt; font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
11pt; font-family: Calibri, sans-serif; color: rgb(31, 73, 125); ">We =
really like a standard codec since it eliminates one area of =
confusion.&nbsp;</span></div></div></div></span></blockquote><br></div><di=
v><br></div><div>This seems like a fairly compelling argument for H.264 =
as a baseline must-implement in RTCWeb. I'm thinking of changing my =
position.</div><div><br></div><div>--</div><div>Dean</div><br></body></htm=
l>=

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Hi Roman,

In my role as a WG chair I have to say that the decision to make SRTP
mandatory to use for WebRTC had a very strong consensus behind it. Yes,
there are some few individuals like yourself that are on the rough side
of this decision.

My personal opinion is that the discussion so far in this thread has
raised most of the issues with supporting both. I think the bid-down
problem is one of the largest for most people. I also see a great
benefit with always using SRTP, in that we will get rid of RTP profile
negotiation. There will be no need to support any other RTP profile than
SAVPF.

Cheers

Magnus

On 2012-05-03 17:48, Roman Shpount wrote:
> 
> On Wed, May 2, 2012 at 8:41 PM, Bernard Aboba <bernard_aboba@hotmail.com
> <mailto:bernard_aboba@hotmail.com>> wrote:
> 
>     *A few years ago, the thought of turning on SRTP by default was a
>     bit scary (mostly because of potential interop issues, not cost). 
>     However, today turning it on by default just works with minimal
>     performance impact or other hassles (other than occasional interop
>     gremlins).  By the time RTCWEB is widely deployed any argument
>     against SRTP will probably be vestigial.*
> 
>     *Given this, it seems to me that the right thing is for SRTP to be
>     mandatory to implement and use, especially if SDES is available, so
>     the likelihood of interoperability will be high.   *____
> 
> First of all, SDES-SRTP in combination with HTTP delivered application
> provides no security at all. If you argue that you need SDES-SRTP for
> interop, you might add support for RTP just as well.
> 
> Second, we just went through a couple of large scale SDES-SRTP
> deployments. Virtually every device had some issues. Even though a lot
> of those issues were not critical (like using RTP encryption routines to
> encrypt RTCP), they were wide spread. What it indicates, is that even
> though SRTP is widely implemented, it does not get nearly as much use as
> plain RTP. Even though it is supported in the datasheet, it is often a
> datasheet support only. Another indication of this is the state of
> libsrtp. It is widely used to implement SRTP support in open source (and
> probably in quite a few closed source) projects. At the same time it has
> issues that would had to be addressed if it had any real life use. One
> examples is crashing on out of order SRTCP packets which, even though
> addressed in libsrtp svn, is present in release source code distribution
> that is commonly used. Another example is random number generator that
> stops operating after a certain number of calls which is not addressed
> anywhere. All of this can be addressed given enough motivation, but this
> will take time.
> 
> Keep in mind, I am not saying that SRTP should not be a MUST implement
> feature, and that plain RTP should be allowed from HTTPS sessions. What
> I am saying that the value of SDES-SRTP for interop is questionable
> (certainly less then value of plain RTP), and SRTP from HTTP session
> with no identity provides no security, so it has little value.
> _____________
> Roman Shpount
> 


-- 

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: Re: [rtcweb] Final plea about SRTP
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On 5/4/12 8:21 AM, Magnus Westerlund wrote:
> Hi Roman,
> 
> In my role as a WG chair I have to say that the decision to make SRTP
> mandatory to use for WebRTC had a very strong consensus behind it. Yes,
> there are some few individuals like yourself that are on the rough side
> of this decision.
> 
> My personal opinion is that the discussion so far in this thread has
> raised most of the issues with supporting both. I think the bid-down
> problem is one of the largest for most people. I also see a great
> benefit with always using SRTP, in that we will get rid of RTP profile
> negotiation. There will be no need to support any other RTP profile than
> SAVPF.

So next main points to be defined, as far as i understand, is by
consensus working on key exchange methods that could be more or less:
- Use only DTLS-SRTP (as it is)
- Use only DTLS-SRTP-EKT
- Use DTLS-SRTP + SDES-SRTP

Other than this i would also suggest to suggest discussing about the
"Authentication" of the call, that currently with DTLS-SRTP can be:
- Based on idP (external identity provide)
- Unauthorized

I would also introduce the ability to verify the DTLS-SRTP call directly
and without intermediary (no trusted third party like idP), with methods
such as SAS.

That's the only way to achieve the "end-to-end security" property that
DTLS-SRTP would like to bring in WebRTC standard.

Otherwise DTLS-SRTP will provide "end-to-end encryption with end-to-site
security" but NO end-to-end security.

Fabio

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Subject: Re: [rtcweb] Use Case draft - legacy interop
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(changing subject line to fork thread)

On 05/03/2012 03:06 PM, Jim Barnett wrote:
> Yes, the use cases in  4.3 are browser-GW, though none of them really
> covers the call center case (routing to an arbitrary agent, recording
> the call, etc.)
>
> However, the requirements in section 5 don't seem to mention legacy
> connectivity, unless that's what F27 is supposed to cover ("The browser
> MUST be able to initiate and accept a media session where the data
> needed for establishment can be carried in SIP.")  It might be good to
> have a specific legacy requirement along the forms of "The Browser must
> be able to communicate with legacy devices that have the following
> characteristics ...." Of course, this gets into the whole
> interoperability discussion. For example, should the Browser be able to
> interoperate with devices that don't support  ICE?  (I have been
> assuming that the call center gateway would  support ICE).
My memory of our last rounds of discussion was that we decided not to 
have a non-gateway legacy interop case because
a) given the variety of legacy equipment, it would basically require a 
specific device as the "interop partner" (a rathole), and
b) it would bind the architecture of RTCWeb in ways that might prevent 
us from achieving our goals in other ways (especially security issues).

YMMV.


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Subject: Re: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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Il giorno Fri, 4 May 2012 00:11:40 -0500
Dean Willis <dean.willis@softarmor.com> ha scritto:

> 
> I asked a colleague who develops serious enterprise video stuff:
> 
> > Here's a question you can perhaps help me with. One of the big
> > fights in the IETF right now is which codec to require as the "must
> > implement" for real-time web communications. The leading candidates
> > are H.264 and VP8, although I've proposed H.263 as a
> > cheaper-to-license starting point. How much of an issue has codec
> > licensing been for you, and what would you prefer as the "standard"
> > codec? 
> 
> 
> And the (somewhat surprising to me) response (which I've expurgated a
> bit as I'm passing it on without explicit permission)  was:
> 
> > I believe the battle is over and H.264 has won. The question is
> > ultimately where has the market gone in terms of playback
> > mechanisms. The answer: Flash: H.264 and (maybe)VP8
> > Adobe Adaptive HDS: H.264
> > QuickTime: H.264
> > Apple Adaptive HLS  H.264
> > Smooth Streaming: VC-1 and H.264
> > DASH  All implementations are H.264
> > Android Cell Phones: H.264
> > Cell Phone hardware: H.264
> > HTML5 on IE: H.264
> > HTML5 on Safari: H.264
> > HTML5 on Firefox: VP8 but recently announced support for H.264 see
> > http://www.theregister.co.uk/2012/03/19/firefox_sucks_on_h264/
> > HTML5 on Chrome: H.264 and VP8 Video Conferencing World: H.264
> > supported universally Broadcast World  Moving from MPEG2 to H.264
> > Every Set Top Box in the world  MPEG2 and H.264
> >  
> > Once the chip manufacturers started building H.264 into the
> > hardware, it was game over. Even though Google was the big VP8
> > developer/supporter, it had to support H.264 in Chrome and Android
> > phones since thats all they had to work with on the phones. Pretty
> > obvious pattern. Codec licensing is largely an issue of the past
> > for us. We do license the H.264 encoders, but it is a miniscule
> > part of our material cost. The decoder license is on a per-machine
> > basis. Every PC/phone/pad that ships today has been licensed for
> > H.264(and AAC) by the manufacturer of the hardware and/or the OS
> > provider. Of course the main complaint on licensing of H.264 is
> > from the content producers which is where the MPEG-LA members want
> > to make money. I dont have much comment on that, but in our
> > customer base  enterprise  I have never once heard it raised as
> > an issue during a buying decision. Also  if VP8 really got popular
> > it is likely that the MPEG-LA patent holders would come after them.
> > Everything I have read indicates that it infringes lots of patents
> > and is basically the same as H.264 in many ways. That is what
> > happened to VC-1 (Windows Media Video). The patent holders wont
> > bother unless is appears to be a real competitor. The html5 people
> > originally wanted to insist on a royalty free codec and gave up
> > based on the reality was that H.264 is the only essential codec. If
> > IETF insists on VP8 or 363 as a must implement they will be
> > ignored. It would be a huge deviation from their general principal
> > of consensus and following the market. I cannot conceive of how it
> > could happen. We really like a standard codec since it eliminates
> > one area of confusion. 
> 
> 
> This seems like a fairly compelling argument for H.264 as a baseline
> must-implement in RTCWeb. I'm thinking of changing my position.
> 
> --
> Dean
> 

Good arguments but this WILL cut out a lot of content producers or
interested folks. As you pointed out, you asked someone "who develops
serious enterprise video stuff", not the student, the geeky guy or the
small startup who wants to try and make some business with a
"funny-hat-chat" or "send-your-friends-a-silly-postcard" application
based on RTCWEB, that is, what is supposed to be the real fuel behind
innovation in RTCWEB in the future. None of them are likely to be
able/willing to afford or be encumbered with the license fees and
limitations such a choice would impose (what is "minuscule" to
enterprises is not that tiny for the poor mob I'm part of).

That said, I currently have no strong opinion towards any particular
choice, but I really am not that comfortable with an H.264 mandatory
codec.

L.

-- 
Lorenzo Miniero, COB

Meetecho s.r.l.
Web Conferencing and Collaboration Tools
http://www.meetecho.com

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Date: Fri, 4 May 2012 10:29:19 +0100
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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <013101cd288c$09328250$1b9786f0$@com> <CALiegf=RsHHf9jCBhE55t7qFUpts8yJ1c8qUX12nc_vd4vgjSQ@mail.gmail.com> <CAJNg7VLCGgJGpV1+YTdBGjMOLBj4x=2-xyu8bpcjRt8Riyo6hA@mail.gmail.com>
To: Marshall Eubanks <marshall.eubanks@gmail.com>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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There is no reason that the RTP to SRTP/ICE gateway needs to be hosted =
server side, it could equally be a software component downloaded to each =
client machine running a WebRTC browser.

In fact, you could even do it in a Java Applet that's sole purpose is to =
relay RTP to SRTP and perform ICE with the local WebRTC stack.

No, actually, forget I said that.

Neil

On 2 May 2012, at 19:19, Marshall Eubanks wrote:
>=20
> My objection is that the proposed system will require middleware to
> interoperate with the
> vast number of videoconferencing sessions out there, most of which use
> RTP. =46rom the
> standpoint of a video service provider, buying hardware to support
> video to laptops is likely to
> lead to requests that participants download some other software which
> interoperates natively.
>=20
> This is an existing business with a fairly large scale and installed
> base. Not operating the way that they do is not likely to go over
> well.
>=20
> Regards
> Marshall


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div>There is no reason that the RTP to SRTP/ICE gateway needs to be =
hosted server side, it could equally be a software component downloaded =
to each client machine running a WebRTC =
browser.</div><div><br></div><div>In fact, you could even do it in a =
Java Applet that's sole purpose is to relay RTP to SRTP and perform ICE =
with the local WebRTC stack.</div><div><br></div><div>No, actually, =
forget I said that.</div><div><br></div><div>Neil</div><br><div><div>On =
2 May 2012, at 19:19, Marshall Eubanks wrote:</div><blockquote =
type=3D"cite"><div><font class=3D"Apple-style-span" =
color=3D"#000000"><br></font>My objection is that the proposed system =
will require middleware to<br>interoperate with the<br>vast number of =
videoconferencing sessions out there, most of which use<br>RTP. =46rom =
the<br>standpoint of a video service provider, buying hardware to =
support<br>video to laptops is likely to<br>lead to requests that =
participants download some other software which<br>interoperates =
natively.<br><br>This is an existing business with a fairly large scale =
and installed<br>base. Not operating the way that they do is not likely =
to go =
over<br>well.<br><br>Regards<br>Marshall<br></div></blockquote></div><br><=
/body></html>=

--Apple-Mail=_868F79E6-7FC1-47E7-8A54-92683166ACE1--

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From: Stephan Wenger <stewe@stewe.org>
To: Lorenzo Miniero <lorenzo@meetecho.com>, Dean Willis <dean.willis@softarmor.com>
Thread-Topic: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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If this is about the MPEG-LA license: it is my understanding that content
for conversational purposes is covered by that license without any charge.
 RTCWEB is about conversational.  We can worry about licensing for the
encoder and decoder (which are part of the web browser), but users should
be fine.
Of course it is fairly well known that MPEG-LA is not the only entity
which has the right to use patent claims against an H.264 implementation
or an H.264 bitstream.
Stephan


On 5.4.2012 10:44 , "Lorenzo Miniero" <lorenzo@meetecho.com> wrote:

>Il giorno Fri, 4 May 2012 00:11:40 -0500
>Dean Willis <dean.willis@softarmor.com> ha scritto:
>
>>=20
>> I asked a colleague who develops serious enterprise video stuff:
>>=20
>> > Here's a question you can perhaps help me with. One of the big
>> > fights in the IETF right now is which codec to require as the "must
>> > implement" for real-time web communications. The leading candidates
>> > are H.264 and VP8, although I've proposed H.263 as a
>> > cheaper-to-license starting point. How much of an issue has codec
>> > licensing been for you, and what would you prefer as the "standard"
>> > codec?=20
>>=20
>>=20
>> And the (somewhat surprising to me) response (which I've expurgated a
>> bit as I'm passing it on without explicit permission)  was:
>>=20
>> > I believe the battle is over and H.264 has won. The question is
>> > ultimately where has the market gone in terms of playback
>> > mechanisms. The answer: Flash: H.264 and (maybe)VP8
>> > Adobe Adaptive HDS: H.264
>> > QuickTime: H.264
>> > Apple Adaptive HLS =AD H.264
>> > Smooth Streaming: VC-1 and H.264
>> > DASH =AD All implementations are H.264
>> > Android Cell Phones: H.264
>> > Cell Phone hardware: H.264
>> > HTML5 on IE: H.264
>> > HTML5 on Safari: H.264
>> > HTML5 on Firefox: VP8 but recently announced support for H.264 see
>> > http://www.theregister.co.uk/2012/03/19/firefox_sucks_on_h264/
>> > HTML5 on Chrome: H.264 and VP8 Video Conferencing World: H.264
>> > supported universally Broadcast World =AD Moving from MPEG2 to H.264
>> > Every Set Top Box in the world =AD MPEG2 and H.264
>> > =20
>> > Once the chip manufacturers started building H.264 into the
>> > hardware, it was game over. Even though Google was the big VP8
>> > developer/supporter, it had to support H.264 in Chrome and Android
>> > phones since that=B9s all they had to work with on the phones. Pretty
>> > obvious pattern. Codec licensing is largely an issue of the past
>> > for us. We do license the H.264 encoders, but it is a miniscule
>> > part of our material cost. The decoder license is on a per-machine
>> > basis. Every PC/phone/pad that ships today has been licensed for
>> > H.264(and AAC) by the manufacturer of the hardware and/or the OS
>> > provider. Of course the main complaint on licensing of H.264 is
>> > from the content producers which is where the MPEG-LA members want
>> > to make money. I don=B9t have much comment on that, but in our
>> > customer base =AD enterprise =AD I have never once heard it raised as
>> > an issue during a buying decision. Also =AD if VP8 really got popular
>> > it is likely that the MPEG-LA patent holders would come after them.
>> > Everything I have read indicates that it infringes lots of patents
>> > and is basically the same as H.264 in many ways. That is what
>> > happened to VC-1 (Windows Media Video). The patent holders won=B9t
>> > bother unless is appears to be a real competitor. The html5 people
>> > originally wanted to insist on a royalty free codec and gave up
>> > based on the reality was that H.264 is the only essential codec. If
>> > IETF insists on VP8 or 363 as a =B3must implement=B2 they will be
>> > ignored. It would be a huge deviation from their general principal
>> > of consensus and following the market. I cannot conceive of how it
>> > could happen. We really like a standard codec since it eliminates
>> > one area of confusion.
>>=20
>>=20
>> This seems like a fairly compelling argument for H.264 as a baseline
>> must-implement in RTCWeb. I'm thinking of changing my position.
>>=20
>> --
>> Dean
>>=20
>
>Good arguments but this WILL cut out a lot of content producers or
>interested folks. As you pointed out, you asked someone "who develops
>serious enterprise video stuff", not the student, the geeky guy or the
>small startup who wants to try and make some business with a
>"funny-hat-chat" or "send-your-friends-a-silly-postcard" application
>based on RTCWEB, that is, what is supposed to be the real fuel behind
>innovation in RTCWEB in the future. None of them are likely to be
>able/willing to afford or be encumbered with the license fees and
>limitations such a choice would impose (what is "minuscule" to
>enterprises is not that tiny for the poor mob I'm part of).
>
>That said, I currently have no strong opinion towards any particular
>choice, but I really am not that comfortable with an H.264 mandatory
>codec.
>
>L.
>
>--=20
>Lorenzo Miniero, COB
>
>Meetecho s.r.l.
>Web Conferencing and Collaboration Tools
>http://www.meetecho.com
>_______________________________________________
>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: Fri, 4 May 2012 13:17:16 +0200
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2012/5/4 Neil Stratford <neils@belltower.co.uk>:
> There is no reason that the RTP to SRTP/ICE gateway needs to be hosted
> server side, it could equally be a software component downloaded to each
> client machine running a WebRTC browser.
>
> In fact, you could even do it in a Java Applet that's sole purpose is to
> relay RTP to SRTP and perform ICE with the local WebRTC stack.
>
> No, actually, forget I said that.

No no no, you said that, you said that !!!
I cannot forget it, it's the most painful concept I've ever read !!! XDDD

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

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To: =?utf-8?B?ScOxYWtpIEJheiBDYXN0aWxsbw==?= <ibc@aliax.net>, Neil Stratford <neils@belltower.co.uk>
Date: Fri, 4 May 2012 13:44:57 +0200
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 4 May 2012 13:22:38 +0200
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints [was RE: Use Case draft]
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2012/5/3 Ravindran, Parthasarathi <pravindran@sonusnet.com>:
> Could you please explain how to differentiate through protocol means that=
 the peer is site (gateway) or endpoint (webbrowser).

The gateway (or the SIP/XMPP peer or whatever) will not announce
DTLS-SRTP support in its SDP.

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

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To: "Fabio Pietrosanti (naif)" <lists@infosecurity.ch>
References: <CAD5OKxtSvdu9gMqfb3ptw5aQJt1NZKLJ1UB_vKRWDXCZurD+1w@mail.gmail.com> <BDA69428-93F2-475B-ABBB-5DE539671DD1@iii.ca> <CAD5OKxs+oZj47DrTSnvaLV7-jNEPOkxjZfJuC5F2fo71kB3-4g@mail.gmail.com> <BLU169-DS251D322307BC173FD221AE932F0@phx.gbl> <CAD5OKxvahkBEs6iVuuyrwuYXzcbKKPvVWL5rx02d6DOhtX_0Cg@mail.gmail.com> <4FA3754D.6020004@ericsson.com> <4FA3776C.5030107@infosecurity.ch>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Final plea about SRTP
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Fabio,

Yes the topics you raise are all suitable for continued discussion and
for future determination of what the consensus is on those.

Cheers

Magnus Westerlund
(As WG chair)



On 2012-05-04 08:30, Fabio Pietrosanti (naif) wrote:
> On 5/4/12 8:21 AM, Magnus Westerlund wrote:
>> Hi Roman,
>>
>> In my role as a WG chair I have to say that the decision to make SRTP
>> mandatory to use for WebRTC had a very strong consensus behind it. Yes,
>> there are some few individuals like yourself that are on the rough side
>> of this decision.
>>
>> My personal opinion is that the discussion so far in this thread has
>> raised most of the issues with supporting both. I think the bid-down
>> problem is one of the largest for most people. I also see a great
>> benefit with always using SRTP, in that we will get rid of RTP profile
>> negotiation. There will be no need to support any other RTP profile than
>> SAVPF.
> 
> So next main points to be defined, as far as i understand, is by
> consensus working on key exchange methods that could be more or less:
> - Use only DTLS-SRTP (as it is)
> - Use only DTLS-SRTP-EKT
> - Use DTLS-SRTP + SDES-SRTP
> 
> Other than this i would also suggest to suggest discussing about the
> "Authentication" of the call, that currently with DTLS-SRTP can be:
> - Based on idP (external identity provide)
> - Unauthorized
> 
> I would also introduce the ability to verify the DTLS-SRTP call directly
> and without intermediary (no trusted third party like idP), with methods
> such as SAS.
> 
> That's the only way to achieve the "end-to-end security" property that
> DTLS-SRTP would like to bring in WebRTC standard.
> 
> Otherwise DTLS-SRTP will provide "end-to-end encryption with end-to-site
> security" but NO end-to-end security.
> 
> Fabio
> _______________________________________________
> 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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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com>, <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl>, <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com>, <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com>, <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com>, <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com>, <4F9EC0B2.10903@alcatel-lucent.com>, <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net>, <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com>, <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com>, <4FA0F43E.4020308@ericsson.com>, <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com>, <013101cd288c$09328250$1b9786f0$@com>, <CALiegf=RsHHf9jCBhE55t7qFUpts8yJ1c8qUX12nc_vd4vgjSQ@mail.gmail.com>, <CAJNg7VLCGgJGpV1+YTdBGjMOLBj4x=2-xyu8bpcjRt8Riyo6hA@mail.gmail.com>, < 4FA21FA3.1090702@alves trand .no> <BLU169-W71F02518D72C33DED96B90932F0@phx.gbl>
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints
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+1 another reason to mandate H.264  

 

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of
Bernard Aboba
Sent: Thursday, May 03, 2012 10:44 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints

 

I concur with Marshall's objections, based on my discussions with customers.


Video conferencing systems are frequently purchased with the expectation
that they will be
used to communicate outside the organization, as well as with other devices,
including PCs,
tablets and mobile devices.   

As an example, an enterprise with a video conferencing system may use it to
communicate
with employees in the field who are using a notebook, tablet or mobile
device. 

Therefore, while an enterprise may have a finite number of video
conferencing units,  it will
often deploy them along with orders of magnitude more devices that
interoperate with those
units.  

The usual requirements for a desirable mobile or tablet implementation apply
here -- power
management (e.g. ability to utilize native encode/decode hardware) is an
important capability.  Also,
the cost of supporting video transcoding for a large number of devices will
frequently be prohibitive,
so that the expectation is that videoconferencing systems and
implementations on PCs, tablets
and mobile devices will be able to negotiate a mutually supported codec. 





On 05/02/2012 08:19 PM, Marshall Eubanks wrote:



My objection is that the proposed system will require middleware to
interoperate with the
vast number of videoconferencing sessions out there, most of which use
RTP. From the
standpoint of a video service provider, buying hardware to support
video to laptops is likely to
lead to requests that participants download some other software which
interoperates natively.
 
This is an existing business with a fairly large scale and installed
base. Not operating the way that they do is not likely to go over
well.
 

Marshall,

I'd like to draw your attention to two numbers:

- Number of installed room videoconferencing units: On the order of 1
million.

http://www.polycom.com/global/documents/company/video_conferencing_by_the_nu
mbers.pdf

- Number of installed Chrome browsers: On the order of 200 million.

http://www.techradar.com/news/internet/google-chrome-browser-hits-200-millio
n-users-1033951

(pulling out Chrome just because I know we've promised to ship this stuff.
And we auto-update, which means most of the users WILL be running a
WebRTC-compatible browser the week after we release it.)

I argue strongly for doing things in ways that we know work, which means not
inventing stuff until we really have to. And I've even argued strongly for
doing things in ways that *permit* interoperation with those older devices -
but not in the cases where doing so risks harming the security, stability
and operational complexity of the installed base that is to come.

                    Harald


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


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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>+1 another reason to mandate H.264 &nbsp;<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
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style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in =
0in 0in'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span>=
</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of =
</b>Bernard Aboba<br><b>Sent:</b> Thursday, May 03, 2012 10:44 =
AM<br><b>To:</b> rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] =
interworking with non-WEBRTC =
endpoints<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>I concur =
with Marshall's objections, based on my discussions with customers. =
<br><br>Video conferencing systems are frequently purchased with the =
expectation that they will be<br>used to communicate outside the =
organization, as well as with other devices, including PCs,<br>tablets =
and mobile devices.&nbsp;&nbsp; <br><br>As an example, an enterprise =
with a video conferencing system may use it to communicate<br>with =
employees in the field who are using a notebook, tablet or mobile =
device. <br><br>Therefore, while an enterprise may have a finite number =
of video conferencing units,&nbsp; it will<br>often deploy them along =
with orders of magnitude more devices that interoperate with =
those<br>units.&nbsp; <br><br>The usual requirements for a desirable =
mobile or tablet implementation apply here -- power<br>management (e.g. =
ability to utilize native encode/decode hardware) is an important =
capability.&nbsp; Also,<br>the cost of supporting video transcoding for =
a large number of devices will frequently be prohibitive,<br>so that the =
expectation is that videoconferencing systems and implementations on =
PCs, tablets<br>and mobile devices will be able to negotiate a mutually =
supported codec. <br><br><br><o:p></o:p></span></p><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><br>On =
05/02/2012 08:19 PM, Marshall Eubanks =
wrote:<br><br><o:p></o:p></span></p><pre>My objection is that the =
proposed system will require middleware =
to<o:p></o:p></pre><pre>interoperate with the<o:p></o:p></pre><pre>vast =
number of videoconferencing sessions out there, most of which =
use<o:p></o:p></pre><pre>RTP. From the<o:p></o:p></pre><pre>standpoint =
of a video service provider, buying hardware to =
support<o:p></o:p></pre><pre>video to laptops is likely =
to<o:p></o:p></pre><pre>lead to requests that participants download some =
other software which<o:p></o:p></pre><pre>interoperates =
natively.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>This is an =
existing business with a fairly large scale and =
installed<o:p></o:p></pre><pre>base. Not operating the way that they do =
is not likely to go =
over<o:p></o:p></pre><pre>well.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></p=
re><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Marshall,<br=
><br>I'd like to draw your attention to two numbers:<br><br>- Number of =
installed room videoconferencing units: On the order of 1 =
million.<br><br><a =
href=3D"http://www.polycom.com/global/documents/company/video_conferencin=
g_by_the_numbers.pdf" =
target=3D"_blank">http://www.polycom.com/global/documents/company/video_c=
onferencing_by_the_numbers.pdf</a><br><br>- Number of installed Chrome =
browsers: On the order of 200 million.<br><br><a =
href=3D"http://www.techradar.com/news/internet/google-chrome-browser-hits=
-200-million-users-1033951" =
target=3D"_blank">http://www.techradar.com/news/internet/google-chrome-br=
owser-hits-200-million-users-1033951</a><br><br>(pulling out Chrome just =
because I know we've promised to ship this stuff. And we auto-update, =
which means most of the users WILL be running a WebRTC-compatible =
browser the week after we release it.)<br><br>I argue strongly for doing =
things in ways that we know work, which means not inventing stuff until =
we really have to. And I've even argued strongly for doing things in =
ways that *permit* interoperation with those older devices - but not in =
the cases where doing so risks harming the security, stability and =
operational complexity of the installed base that is to =
come.<br><br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Harald<br><br><br>_______________________________________________ rtcweb =
mailing list <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a> <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></span></p></div></div></div></bo=
dy></html>
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WG,

We at Ericsson hosting the meeting have now secured meeting rooms for
the Interim. Therefore I can now announce that we will be holding the
meeting in Kista.

We also provide some more information about Kista, Hotels and travel in
Stockholm. I would recommend that people secure a hotel booking as soon
as possible as there is quite some pressure on hotel rooms during the
period of the Interim.

Please see for details:
http://dl.dropbox.com/u/11888030/Practical%20Information%20for%20CLUE%20and%20WebRTC.htm

Questions to the host about the location etc can be sent to me.

Best Regards

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 magnus.westerlund@ericsson.com  Fri May  4 06:28:22 2012
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For people that have problems accessing dropbox please try this location
instead.

http://www.denstore.se/IETF/Practical%20Information%20for%20CLUE%20and%20WebRTC.htm

Cheers

Magnus

On 2012-05-04 15:14, Magnus Westerlund wrote:
> WG,
> 
> We at Ericsson hosting the meeting have now secured meeting rooms for
> the Interim. Therefore I can now announce that we will be holding the
> meeting in Kista.
> 
> We also provide some more information about Kista, Hotels and travel in
> Stockholm. I would recommend that people secure a hotel booking as soon
> as possible as there is quite some pressure on hotel rooms during the
> period of the Interim.
> 
> Please see for details:
> http://dl.dropbox.com/u/11888030/Practical%20Information%20for%20CLUE%20and%20WebRTC.htm
> 
> Questions to the host about the location etc can be sent to me.
> 
> Best Regards
> 
> 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 xavier.marjou@gmail.com  Fri May  4 06:50:41 2012
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Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints
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+1

On Fri, May 4, 2012 at 3:13 PM, Richard Shockey <richard@shockey.us> wrote:
> +1 another reason to mandate H.264
>
>
>
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf =
Of
> Bernard Aboba
> Sent: Thursday, May 03, 2012 10:44 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] interworking with non-WEBRTC endpoints
>
>
>
> I concur with Marshall's objections, based on my discussions with custome=
rs.
>
> Video conferencing systems are frequently purchased with the expectation
> that they will be
> used to communicate outside the organization, as well as with other devic=
es,
> including PCs,
> tablets and mobile devices.
>
> As an example, an enterprise with a video conferencing system may use it =
to
> communicate
> with employees in the field who are using a notebook, tablet or mobile
> device.
>
> Therefore, while an enterprise may have a finite number of video
> conferencing units,=A0 it will
> often deploy them along with orders of magnitude more devices that
> interoperate with those
> units.
>
> The usual requirements for a desirable mobile or tablet implementation ap=
ply
> here -- power
> management (e.g. ability to utilize native encode/decode hardware) is an
> important capability.=A0 Also,
> the cost of supporting video transcoding for a large number of devices wi=
ll
> frequently be prohibitive,
> so that the expectation is that videoconferencing systems and
> implementations on PCs, tablets
> and mobile devices will be able to negotiate a mutually supported codec.
>
>
>
> On 05/02/2012 08:19 PM, Marshall Eubanks wrote:
>
> My objection is that the proposed system will require middleware to
>
> interoperate with the
>
> vast number of videoconferencing sessions out there, most of which use
>
> RTP. From the
>
> standpoint of a video service provider, buying hardware to support
>
> video to laptops is likely to
>
> lead to requests that participants download some other software which
>
> interoperates natively.
>
>
>
> This is an existing business with a fairly large scale and installed
>
> base. Not operating the way that they do is not likely to go over
>
> well.
>
>
>
> Marshall,
>
> I'd like to draw your attention to two numbers:
>
> - Number of installed room videoconferencing units: On the order of 1
> million.
>
> http://www.polycom.com/global/documents/company/video_conferencing_by_the=
_numbers.pdf
>
> - Number of installed Chrome browsers: On the order of 200 million.
>
> http://www.techradar.com/news/internet/google-chrome-browser-hits-200-mil=
lion-users-1033951
>
> (pulling out Chrome just because I know we've promised to ship this stuff=
.
> And we auto-update, which means most of the users WILL be running a
> WebRTC-compatible browser the week after we release it.)
>
> I argue strongly for doing things in ways that we know work, which means =
not
> inventing stuff until we really have to. And I've even argued strongly fo=
r
> doing things in ways that *permit* interoperation with those older device=
s -
> but not in the cases where doing so risks harming the security, stability
> and operational complexity of the installed base that is to come.
>
> =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Harald
>
>
> _______________________________________________ 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 dean.willis@softarmor.com  Fri May  4 07:10:24 2012
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Subject: Re: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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On May 4, 2012 3:45 AM, "Lorenzo Miniero" <lorenzo@meetecho.com> wrote:
>
> Good arguments but this WILL cut out a lot of content producers or
> interested folks. As you pointed out, you asked someone "who develops
> serious enterprise video stuff", not the student, the geeky guy or the
> small startup who wants to try and make some business with a
> "funny-hat-chat" or "send-your-friends-a-silly-postcard" application
> based on RTCWEB, that is, what is supposed to be the real fuel behind
> innovation in RTCWEB in the future.

I am more worried about the guy implementing a WebRTC security camera that
uses an embedded Linux kernel and a software video encoder. Each unit might
"produce" video 24x7. But there is no MPEG-LA licensed browser or OS or
encoder chip to fall back on. The whole product might sell for less than it
might cost him to license the codec.

--e89a8fb1f1f8fe528904bf367a0b
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<p><br>
On May 4, 2012 3:45 AM, &quot;Lorenzo Miniero&quot; &lt;<a href=3D"mailto:l=
orenzo@meetecho.com">lorenzo@meetecho.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Good arguments but this WILL cut out a lot of content producers or<br>
&gt; interested folks. As you pointed out, you asked someone &quot;who deve=
lops<br>
&gt; serious enterprise video stuff&quot;, not the student, the geeky guy o=
r the<br>
&gt; small startup who wants to try and make some business with a<br>
&gt; &quot;funny-hat-chat&quot; or &quot;send-your-friends-a-silly-postcard=
&quot; application<br>
&gt; based on RTCWEB, that is, what is supposed to be the real fuel behind<=
br>
&gt; innovation in RTCWEB in the future. </p>
<p>I am more worried about the guy implementing a WebRTC security camera th=
at uses an embedded Linux kernel and a software video encoder. Each unit mi=
ght &quot;produce&quot; video 24x7. But there is no MPEG-LA licensed browse=
r or OS or encoder chip to fall back on. The whole product might sell for l=
ess than it might cost him to license the codec.</p>


--e89a8fb1f1f8fe528904bf367a0b--

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On 05/04/2012 10:10 AM, Dean Willis wrote:
>
>
> On May 4, 2012 3:45 AM, "Lorenzo Miniero" <lorenzo@meetecho.com
> <mailto:lorenzo@meetecho.com>> wrote:
> >
> > Good arguments but this WILL cut out a lot of content producers or
> > interested folks. As you pointed out, you asked someone "who develops
> > serious enterprise video stuff", not the student, the geeky guy or the
> > small startup who wants to try and make some business with a
> > "funny-hat-chat" or "send-your-friends-a-silly-postcard" application
> > based on RTCWEB, that is, what is supposed to be the real fuel behind
> > innovation in RTCWEB in the future.
>
> I am more worried about the guy implementing a WebRTC security camera
> that uses an embedded Linux kernel and a software video encoder. Each
> unit might "produce" video 24x7. But there is no MPEG-LA licensed
> browser or OS or encoder chip to fall back on. The whole product might
> sell for less than it might cost him to license the codec.
>
+1

The number of embedded devices, smartphones, cameras, etc, are rapidly
outnumbering the number of conventional desktop/laptops, and are FAR
more cost sensitive than those uses, and not necessarily going to have
bought out the patent for conventional video playback.

So I don't find arguments that come from "Enterprise" environments by
themselves at all compelling, but merely one more factoid to join the
other factoids in the discussion.
                    - Jim
                               

From bernard_aboba@hotmail.com  Fri May  4 07:46:40 2012
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Subject: Re: [rtcweb] Use Case draft - legacy interop
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> My memory of our last rounds of discussion was that we decided not to=20
> have a non-gateway legacy interop case because
> a) given the variety of legacy equipment=2C it would basically require a=
=20
> specific device as the "interop partner" (a rathole)=2C and
> b) it would bind the architecture of RTCWeb in ways that might prevent=20
> us from achieving our goals in other ways (especially security issues).

[BA] That was my recollection as well. =20

The discussion we're getting into now involves how much the "gateway" will
need to do.  This will of necessity vary depending on how ancient the "lega=
cy"
equipment is.  For example=2C if the "legacy" is only a few years old=2C it=
 may=20
support SRTP=2C and with appropriate RTCWEB accommodation (e.g. EKT or
SDES)=2C the "gateway" would not need to decrypt/encrypt every media frame.=
=20
In this scenario=2C the "gateway" functionality needed by WebRTC would be i=
n
the range of capabilities already supported by SBCs (which are proliferatin=
g
like weeds even in "legacy" deployments).  So one could claim that the=20
additional burden imposed by WebRTC interop gateways would be tolerable.

If the "legacy" is older (e.g. circa mid-2000s)=2C then it will probably no=
t
support SRTP=2C and the burden on the gateway becomes greater.  Personally=
=2C
I'm ok with this because many of those "extreme legacy" deployments are
nearing (or have already passed) "end of life" and by the time RTCWEB
is widely deployed=2C they will have become difficult to support/maintain.=
=20
Since those "extreme legacy" deployments don't support most of the=20
capabilities of WebRTC anyway without an upgrade (e.g. typically no video=20
or IM&P)=2C it doesn't seem sensible to me to warp the entire WebRTC
architecture to accommodate them.
 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
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<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B My memory of our last rounds of discussion was that we deci=
ded not to <br>&gt=3B have a non-gateway legacy interop case because<br>&gt=
=3B a) given the variety of legacy equipment=2C it would basically require =
a <br>&gt=3B specific device as the "interop partner" (a rathole)=2C and<br=
>&gt=3B b) it would bind the architecture of RTCWeb in ways that might prev=
ent <br>&gt=3B us from achieving our goals in other ways (especially securi=
ty issues).<br><br>[BA] That was my recollection as well.&nbsp=3B <br><br>T=
he discussion we're getting into now involves how much the "gateway" will<b=
r>need to do.&nbsp=3B This will of necessity vary depending on how ancient =
the "legacy"<br>equipment is.&nbsp=3B For example=2C if the "legacy" is onl=
y a few years old=2C it may <br>support SRTP=2C and with appropriate RTCWEB=
 accommodation (e.g. EKT or<br>SDES)=2C the "gateway" would not need to dec=
rypt/encrypt every media frame. <br>In this scenario=2C the "gateway" funct=
ionality needed by WebRTC would be in<br>the range of capabilities already =
supported by SBCs (which are proliferating<br>like weeds even in "legacy" d=
eployments).&nbsp=3B So one could claim that the <br>additional burden impo=
sed by WebRTC interop gateways would be tolerable.<br><br>If the "legacy" i=
s older (e.g. circa mid-2000s)=2C then it will probably not<br>support SRTP=
=2C and the burden on the gateway becomes greater.&nbsp=3B Personally=2C<br=
>I'm ok with this because many of those "extreme legacy" deployments are<br=
>nearing (or have already passed) "end of life" and by the time RTCWEB<br>i=
s widely deployed=2C they will have become difficult to support/maintain. <=
br>Since those "extreme legacy" deployments don't support most of the <br>c=
apabilities of WebRTC anyway without an upgrade (e.g. typically no video <b=
r>or IM&amp=3BP)=2C it doesn't seem sensible to me to warp the entire WebRT=
C<br>architecture to accommodate them.<br></div> 		 	   		  </div></body>
</html>=

--_fe0a380e-3a14-4ffa-8bd3-4fcdd05acc00_--

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Subject: Re: [rtcweb] Final plea about SRTP
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On Fri, May 4, 2012 at 2:21 AM, Magnus Westerlund <
magnus.westerlund@ericsson.com> wrote:

> In my role as a WG chair I have to say that the decision to make SRTP
> mandatory to use for WebRTC had a very strong consensus behind it. Yes,
> there are some few individuals like yourself that are on the rough side
> of this decision.
>

Let's hope this will work in real life use cases. If it does not (which is
what I believe, but I can certainly be wrong), then RTP can be put back at
1.1 version of the standard.

 I think the bid-down problem is one of the largest for most people.


I do not think we need to support auto-negotiation of RTP vs SRTP. Also,
RTP should not be allowed from HTTPS sessions, so I do not think bid down
is a problem at all.

I also see a great benefit with always using SRTP, in that we will get rid
> of RTP profile
> negotiation. There will be no need to support any other RTP profile than
> SAVPF.
>

I do see a benefit of using one RTP profile only, but this will require
WebRTC to use yet another feature that had almost no real life use. This
will also ensure that RTCP will need to be re-encoded (as probably RTP)
when processing calls to anything outside of WebRTC world.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Fri, May 4, 2012 at 2:21 AM=
, Magnus Westerlund <span dir=3D"ltr">&lt;<a href=3D"mailto:magnus.westerlu=
nd@ericsson.com" target=3D"_blank">magnus.westerlund@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">In my role as a WG chair I have to say that =
the decision to make SRTP<br>
mandatory to use for WebRTC had a very strong consensus behind it. Yes,<br>
there are some few individuals like yourself that are on the rough side<br>
of this decision.<br>
</blockquote></div><br>
Let&#39;s hope this will work in real life use cases. If it does not (which=
 is what I believe, but I can certainly be wrong), then RTP can be put back=
 at 1.1 version of the standard.<br clear=3D"all"><br>
<div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

I think the bid-down
problem is one of the largest for most people. </blockquote><div><br>I do n=
ot think we need to support auto-negotiation of RTP vs SRTP. Also, RTP shou=
ld not be allowed from HTTPS sessions, so I do not think bid down is a prob=
lem at all. <br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">I also see a gr=
eat
benefit with always using SRTP, in that we will get rid of RTP profile<br>
negotiation. There will be no need to support any other RTP profile than
SAVPF.<br></blockquote></div><br>I do see a benefit of using one RTP profil=
e only, but this will require WebRTC to use yet another feature that had al=
most no real life use. This will also ensure that RTCP will need to be re-e=
ncoded (as probably RTP) when processing calls to anything outside of WebRT=
C world.<br>
_____________<br>Roman Shpount<br>
<br>

--047d7b15b17182ce3e04bf38966d--

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Subject: Re: [rtcweb] Final plea about SRTP
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On 5/4/2012 12:41 PM, Roman Shpount wrote:
>
> On Fri, May 4, 2012 at 2:21 AM, Magnus Westerlund
> <magnus.westerlund@ericsson.com <mailto:magnus.westerlund@ericsson.com>>
> wrote:

>     I think the bid-down problem is one of the largest for most people.
>
> I do not think we need to support auto-negotiation of RTP vs SRTP. Also,
> RTP should not be allowed from HTTPS sessions, so I do not think bid
> down is a problem at all.

You forget that bid-down includes bid-downs by the JS or server (which 
are not trusted in our model), not just by on-path attackers.

>     I also see a great benefit with always using SRTP, in that we will
>     get rid of RTP profile
>     negotiation. There will be no need to support any other RTP profile
>     than SAVPF.
>
> I do see a benefit of using one RTP profile only, but this will require
> WebRTC to use yet another feature that had almost no real life use. This
> will also ensure that RTCP will need to be re-encoded (as probably RTP)
> when processing calls to anything outside of WebRTC world.

I used to work on hardware endpoints that have been using SAVPF since 
2004, with hundreds of thousands of units in the field.


-- 
Randell Jesup
randell-ietf@jesup.org

From ted.ietf@gmail.com  Fri May  4 10:10:13 2012
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Subject: Re: [rtcweb] Host and Local Information for Stockholm Interim
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Thanks for the details Magnus.  I personally decided to go for the
Scandic Victoria Tower in Kista; should we start a wiki of where folks
will be so aggregation can happen for those who want it?

Ted




On Fri, May 4, 2012 at 6:28 AM, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> For people that have problems accessing dropbox please try this location
> instead.
>
> http://www.denstore.se/IETF/Practical%20Information%20for%20CLUE%20and%20=
WebRTC.htm
>
> Cheers
>
> Magnus
>
> On 2012-05-04 15:14, Magnus Westerlund wrote:
>> WG,
>>
>> We at Ericsson hosting the meeting have now secured meeting rooms for
>> the Interim. Therefore I can now announce that we will be holding the
>> meeting in Kista.
>>
>> We also provide some more information about Kista, Hotels and travel in
>> Stockholm. I would recommend that people secure a hotel booking as soon
>> as possible as there is quite some pressure on hotel rooms during the
>> period of the Interim.
>>
>> Please see for details:
>> http://dl.dropbox.com/u/11888030/Practical%20Information%20for%20CLUE%20=
and%20WebRTC.htm
>>
>> Questions to the host about the location etc can be sent to me.
>>
>> Best Regards
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> 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
>>
>>
>
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> 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

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On Fri, May 4, 2012 at 1:04 PM, Randell Jesup <randell-ietf@jesup.org>wrote:

> You forget that bid-down includes bid-downs by the JS or server (which are
> not trusted in our model), not just by on-path attackers.
>

If your session is initiated by HTTPS, using RTP should not be an option
(the same way as using HTTP from HTTPS is not normally an option). If your
session is HTTP, whole application can be spoofed, so there is no security
to begin with.

I used to work on hardware endpoints that have been using SAVPF since 2004,
> with hundreds of thousands of units in the field.
>

I thought SAVPF was only standardized in 2008 and AVPF was standardized in
2006. AVPF was discussed for a while though, so I would assumed you worked
with something that implemented one of the drafts...
_____________
Roman Shpount

--047d7b15b1717dd00f04bf397d76
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<br clear=3D"all"><div class=3D"gmail_quote">On Fri, May 4, 2012 at 1:04 PM=
, Randell Jesup <span dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.=
org" target=3D"_blank">randell-ietf@jesup.org</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">
You forget that bid-down includes bid-downs by the JS or server (which are =
not trusted in our model), not just by on-path attackers.<div class=3D"im">=
</div></blockquote><div><br>If your session is initiated by HTTPS, using RT=
P should not be an option (the same way as using HTTP from HTTPS is not nor=
mally an option). If your session is HTTP, whole application can be spoofed=
, so there is no security to begin with. <br>
<br></div><blockquote class=3D"gmail_quote" style=3D"margin:0pt 0pt 0pt 0.8=
ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div class=3D"i=
m">I used to work on hardware endpoints that have been using SAVPF since 20=
04, with hundreds of thousands of units in the field.<br>
</div></blockquote><div><br>I thought SAVPF was only standardized in 2008 a=
nd AVPF was standardized in 2006. AVPF was discussed for a while though, so=
 I would assumed you worked with something that implemented one of the draf=
ts...=A0 <br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b15b1717dd00f04bf397d76--

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From: Stephan Wenger <stewe@stewe.org>
To: Dean Willis <dean.willis@softarmor.com>, Lorenzo Miniero <lorenzo@meetecho.com>
Thread-Topic: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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--_000_CBC9F45386B26stewesteweorg_
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From: Dean Willis <dean.willis@softarmor.com<mailto:dean.willis@softarmor.c=
om>>
Date: Friday, 4 May, 2012 16:10
[=85]
[=85]

I am more worried about the guy implementing a WebRTC security camera that =
uses an embedded Linux kernel and a software video encoder. Each unit might=
 "produce" video 24x7. But there is no MPEG-LA licensed browser or OS or en=
coder chip to fall back on. The whole product might sell for less than it m=
ight cost him to license the codec.

Really?  The MPEG-LA license terms for one encoder/decoder: $0 for the firs=
t 100,000 units per year.  $0.20 for the next several millions, after that =
$0.10 until the cap is reached.  And, the MPEG-LA license is "take it or le=
ave it" (at least for small fish like your hypothetical camera guy), so eve=
n the lawyer cost for review should be minimal.

No, I continue to believe that the MPEG-LA license is not a hurdle for anyo=
ne but those giving stuff away for free, or having a business model that di=
sallows them to pay licensing fees.  It's the patents not covered by that l=
icense that you may worry about.

Stephan

--_000_CBC9F45386B26stewesteweorg_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; font-size: 14px; font-family: Calibri, sans-ser=
if; ">
<div style=3D"color: rgb(0, 0, 0); "><br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"color: rgb(0, 0, 0); ">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Dean Willis &lt;<a href=3D"ma=
ilto:dean.willis@softarmor.com">dean.willis@softarmor.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday, 4 May, 2012 16:10 <br=
>
</div>
</span>
<div style=3D"color: rgb(0, 0, 0); ">[=85]</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"color: rgb(0, 0, 0); ">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<b>[=85]</b></div>
<div>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<p>I am more worried about the guy implementing a WebRTC security camera th=
at uses an embedded Linux kernel and a software video encoder. Each unit mi=
ght &quot;produce&quot; video 24x7. But there is no MPEG-LA licensed browse=
r or OS or encoder chip to fall back on. The
 whole product might sell for less than it might cost him to license the co=
dec.</p>
</blockquote>
</div>
</div>
</span>
<div><font class=3D"Apple-style-span" color=3D"#0000ff">Really? &nbsp;The M=
PEG-LA license terms for one encoder/decoder: $0 for the first 100,000 unit=
s per year. &nbsp;$0.20 for the next several millions, after that $0.10 unt=
il the cap is reached. &nbsp;And, the MPEG-LA license
 is &quot;take it or leave it&quot; (at least for small fish like your hypo=
thetical camera guy), so even the lawyer cost for review should be minimal.=
</font></div>
<div><font class=3D"Apple-style-span" color=3D"#0000ff"><br>
</font></div>
<div><font class=3D"Apple-style-span" color=3D"#0000ff">No, I continue to b=
elieve that the MPEG-LA license is not a hurdle for anyone but those giving=
 stuff away for free, or having a business model that disallows them to pay=
 licensing fees. &nbsp;It's the patents not
 covered by that license that you may worry about.</font></div>
<div><font class=3D"Apple-style-span" color=3D"#0000ff"><br>
</font></div>
<div><font class=3D"Apple-style-span" color=3D"#0000ff">Stephan &nbsp;</fon=
t></div>
</body>
</html>

--_000_CBC9F45386B26stewesteweorg_--

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References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <4FA1575C.4050508@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F4BF@NAHALD.us.int.genesyslab.com> <4FA37A1E.4080806@alvestrand.no>
From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Fri, 4 May 2012 18:17:07 +0200
Message-ID: <CALiegf=H5QH_YY-cJ4z29wChWZ-VoQpHvsZCeaJPjTgVp+km3Q@mail.gmail.com>
To: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - legacy interop
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2012/5/4 Harald Alvestrand <harald@alvestrand.no>:
> My memory of our last rounds of discussion was that we decided not to hav=
e a
> non-gateway legacy interop case

Hi Harald, is that really *decided*? I will ask the same question in a
different way:

If I have a SIP phone implementing ICE and DTLS-SRTP (which is
standarized for SIP regardless it has null impementation), will my SIP
phone be able to *directly* talk in the media plane with a WebRTC
browser? or not?

And also, what does it mean a "non-gateway"? Gateways are not
specified, they are custom boxes implementing some kind of protocol
translation. The only way in which it can be said that "there is not
non-gateway legacy interop" is by making WebRTC *explicitly*
non-compliant with any other existing device (for example by requiring
some annoying a=3DensureWebRTC line in the SDP, in this way you can be
sure that WebRTC entities will interop with nobody but with specific
"gateways" that know the trick).


Thanks a lot.


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

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> I am more worried about the guy implementing a WebRTC security camera
> that uses an embedded Linux kernel and a software video encoder. Each
> unit might "produce" video 24x7. But there is no MPEG-LA licensed
> browser or OS or encoder chip to fall back on. The whole product might
> sell for less than it might cost him to license the codec.

[BA]  We are rapidly moving toward inclusion of hardware video encode/decod=
e by default on battery-powered devices (mobile handsets=2C tablets=2C etc.=
).=20

Since software encode/decode is processor (and battery) intensive=2C the la=
test generation of webcams have video encode/decode built-in.  For example=
=2C see:
http://www.ehomeupgrade.com/2011/06/02/skype-certified-webcams-tote-built-i=
n-720p-h-264-video-encoders/

When handled purely in software=2C it is quite difficult to enable video se=
ssions of substantial duration without draining the battery.  =20

While my assumption is that hardware support for encode/decode will become =
less codec-specific over time=2C  that is typically not the case today.

 		 	   		  =

--_0b5b1707-703a-4d7c-9fde-6b8eefcb3b86_
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<html>
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font-family:Tahoma
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--></style></head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B I am more worried about the guy implementing a WebRTC secur=
ity camera<br>&gt=3B that uses an embedded Linux kernel and a software vide=
o encoder. Each<br>&gt=3B unit might "produce" video 24x7. But there is no =
MPEG-LA licensed<br>&gt=3B browser or OS or encoder chip to fall back on. T=
he whole product might<br>&gt=3B sell for less than it might cost him to li=
cense the codec.<br><br>[BA]&nbsp=3B We are rapidly moving toward inclusion=
 of hardware video encode/decode by default on battery-powered devices (mob=
ile handsets=2C tablets=2C etc.). <br><br>Since software encode/decode is p=
rocessor (and battery) intensive=2C the latest generation of webcams have v=
ideo encode/decode built-in.&nbsp=3B For example=2C see:<br>http://www.ehom=
eupgrade.com/2011/06/02/skype-certified-webcams-tote-built-in-720p-h-264-vi=
deo-encoders/<br><br>When handled purely in software=2C it is quite difficu=
lt to enable video sessions of substantial duration without draining the ba=
ttery.&nbsp=3B&nbsp=3B <br><br>While my assumption is that hardware support=
 for encode/decode will become less codec-specific over time=2C&nbsp=3B tha=
t is typically not the case today.<br><br></div> 		 	   		  </div></body>
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Subject: [rtcweb] Something new for the RTCWEB requirements document ..
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http://news.cnet.com/8301-1009_3-57428067-83/fbi-we-need-wiretap-ready-web-s
ites-now/

 

<sigh> 

 

Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
< <mailto:richard(at)shockey.us> mailto:richard(at)shockey.us>
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Date: Sat, 05 May 2012 15:39:52 +0200
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Subject: [rtcweb] SIP over WebSocket update
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Hi all,

the SIPCore WG has accepted the SIP over WebSocket transport:
http://www.ietf.org/mail-archive/web/sipcore/current/msg04970.html

As WebRTC WG we should consider this normal trend/evolution in bringing
existing standard over Web Transport without reinventing too much new
non compatible technologies.

One step towards easy SDP transport (usable for SDES-SRTP, ZRTP
fingerprint transport, etc).

Fabio

From bernard_aboba@hotmail.com  Sat May  5 09:21:25 2012
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Subject: Re: [rtcweb] Something new for the RTCWEB requirements document ..
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>From RFC 2804:

   The Internet Engineering Task Force (IETF) has been asked to take a
   position on the inclusion into IETF standards-track documents of
   functionality designed to facilitate wiretapping.

   This memo explains what the IETF thinks the question means=2C why its
   answer is "no"=2C and what that answer means.From: richard@shockey.us
To: rtcweb@ietf.org
Date: Fri=2C 4 May 2012 21:46:38 -0400
Subject: [rtcweb] Something new for the RTCWEB requirements document ..

 http://news.cnet.com/8301-1009_3-57428067-83/fbi-we-need-wiretap-ready-web=
-sites-now/ <sigh>  Richard Shockey
Shockey Consulting
Chairman of the Board of Directors SIP Forum
PSTN Mobile: +1 703.593.2683
<mailto:richard(at)shockey.us>
skype-linkedin-facebook: rshockey101
http//www.sipforum.org=20
_______________________________________________
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'>
>From RFC 2804:<br><br><pre>   The Internet Engineering Task Force (IETF) ha=
s been asked to take a
   position on the inclusion into IETF standards-track documents of
   functionality designed to facilitate wiretapping.

   This memo explains what the IETF thinks the question means=2C why its
   answer is "no"=2C and what that answer means.</pre><div><div id=3D"SkyDr=
ivePlaceholder"></div><hr id=3D"stopSpelling">From: richard@shockey.us<br>T=
o: rtcweb@ietf.org<br>Date: Fri=2C 4 May 2012 21:46:38 -0400<br>Subject: [r=
tcweb] Something new for the RTCWEB requirements document ..<br><br><style>=
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3-57428067-83/fbi-we-need-wiretap-ready-web-sites-now/" target=3D"_blank">h=
ttp://news.cnet.com/8301-1009_3-57428067-83/fbi-we-need-wiretap-ready-web-s=
ites-now/</a></p><p class=3D"ecxMsoNormal">&nbsp=3B</p><p class=3D"ecxMsoNo=
rmal">&lt=3Bsigh&gt=3B </p><p class=3D"ecxMsoNormal">&nbsp=3B</p><p class=
=3D"ecxMsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-size:1=
0.0pt=3Bfont-family:&quot=3BTimes New Roman&quot=3B=2C&quot=3Bserif&quot=3B=
">Richard Shockey<br>Shockey Consulting<br>Chairman of the Board of Directo=
rs SIP Forum<br>PSTN Mobile: +1 703.593.2683<br>&lt=3B<a href=3D"mailto:ric=
hard%28at%29shockey.us"><span style=3D"color:blue">mailto:richard(at)shocke=
y.us</span></a>&gt=3B<br>skype-linkedin-facebook: rshockey101<br>http//www.=
sipforum.org</span><span style=3D"font-size:12.0pt=3Bfont-family:&quot=3BTi=
mes New Roman&quot=3B=2C&quot=3Bserif&quot=3B"></span></p><p class=3D"ecxMs=
oNormal">&nbsp=3B</p></div><br>____________________________________________=
___
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
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Date: Sat, 5 May 2012 09:24:48 -0700
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - legacy interop
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of I=C3=B1aki Baz Castillo
> Sent: Friday, May 04, 2012 9:17 AM
> To: Harald Alvestrand
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft - legacy interop
>=20
> 2012/5/4 Harald Alvestrand <harald@alvestrand.no>:
> > My memory of our last rounds of discussion was that we decided not =
to
> have a
> > non-gateway legacy interop case
>=20
> Hi Harald, is that really *decided*? I will ask the same question in a
> different way:
>=20
> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
> standarized for SIP regardless it has null impementation), will my SIP
> phone be able to *directly* talk in the media plane with a WebRTC
> browser? or not?

That would work.

-d


> And also, what does it mean a "non-gateway"? Gateways are not
> specified, they are custom boxes implementing some kind of protocol
> translation. The only way in which it can be said that "there is not
> non-gateway legacy interop" is by making WebRTC *explicitly*
> non-compliant with any other existing device (for example by requiring
> some annoying a=3DensureWebRTC line in the SDP, in this way you can be
> sure that WebRTC entities will interop with nobody but with specific
> "gateways" that know the trick).
>=20
>=20
> Thanks a lot.
>=20
>=20
> --
> I=C3=B1aki 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: Sat, 5 May 2012 18:32:02 +0200
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - legacy interop
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2012/5/5 Dan Wing <dwing@cisco.com>:
>> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
>> standarized for SIP regardless it has null impementation), will my SIP
>> phone be able to *directly* talk in the media plane with a WebRTC
>> browser? or not?
>
> That would work.

Ok, so then, taking into account that SIP defines the usage of
DTLS-SRTP in RFC 5763 (regardless no one device implements it), why
does this WG assumes that "interop with non WebRTC endpoints will be
made via *gateways*"?

Let me please repeat my question: if my SIP phone implements RFC 5763,
can my SIP phone directly interop at media plane with a WebRTC
browser? (I know you already replied this, but I want to be very very
sure) :)

Thanks a lot.


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

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Inaki said:=20


> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
> standarized for SIP regardless it has null impementation)=2C will my SIP
> phone be able to *directly* talk in the media plane with a WebRTC
> browser? or not?

[BA] While I believe that the answer to this question should be required to=
 be "yes"=2C I do not think we can yet say whether this will be true or not=
.=20

For example=2C for interoperation to be possible=2C the "continuing consent=
" mechanism used in ICE would need to be backward compatible with existing =
implementations.=20

This is not the case for the mechanism proposed in draft-muthu-behave-conse=
nt-freshness.=20

Similarly=2C we would need to ensure that the DTLS-SRTP functionality speci=
fied in WebRTC could be used to interoperate with existing implementations=
=2C which as I understand it=2C support EKT.=20



 		 	   		  =

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<head>
<style><!--
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<body class=3D'hmmessage'><div dir=3D'ltr'>
Inaki said: <br><br><div><div id=3D"SkyDrivePlaceholder"></div><br>&gt=3B I=
f I have a SIP phone implementing ICE and DTLS-SRTP (which is<br>&gt=3B sta=
ndarized for SIP regardless it has null impementation)=2C will my SIP<br>&g=
t=3B phone be able to *directly* talk in the media plane with a WebRTC<br>&=
gt=3B browser? or not?<br><br>[BA] While I believe that the answer to this =
question should be required to be "yes"=2C I do not think we can yet say wh=
ether this will be true or not. <br><br>For example=2C for interoperation t=
o be possible=2C the "continuing consent" mechanism used in ICE would need =
to be backward compatible with existing implementations. <br><br>This is no=
t the case for the mechanism proposed in draft-muthu-behave-consent-freshne=
ss. <br><br>Similarly=2C we would need to ensure that the DTLS-SRTP functio=
nality specified in WebRTC could be used to interoperate with existing impl=
ementations=2C which as I understand it=2C support EKT. <br><br><br><br></d=
iv> 		 	   		  </div></body>
</html>=

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From: "Dan Wing" <dwing@cisco.com>
To: "=?utf-8?Q?'I=C3=B1aki_Baz_Castillo'?=" <ibc@aliax.net>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F066@NAHALD.us.int.genesyslab.com> <BLU169-W7C59E1EDB4CB06B648577932B0@phx.gbl> <387F9047F55E8C42850AD6B3A7A03C6C0E23AFFF@inba-mail01.sonusnet.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <4FA1575C.4050508@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F4BF@NAHALD.us.int.genesyslab.com> <4FA37A1E.4080806@alvestrand.no> <CALiegf=H5QH_YY-cJ4z29wChWZ-VoQpHvsZCeaJPjTgVp+km3Q@mai l.gmail.com> <0db701cd 2adb$98082f10$c8188d30$@com> <CALiegf=C7a6zeUDHn-Wuku9eFWmADG5N+D8oXSQbwJKSYYjcQg@mail.gmail.com>
In-Reply-To: <CALiegf=C7a6zeUDHn-Wuku9eFWmADG5N+D8oXSQbwJKSYYjcQg@mail.gmail.com>
Date: Sat, 5 May 2012 10:24:48 -0700
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use Case draft - legacy interop
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> -----Original Message-----
> From: I=C3=B1aki Baz Castillo [mailto:ibc@aliax.net]
> Sent: Saturday, May 05, 2012 9:32 AM
> To: Dan Wing
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Use Case draft - legacy interop
>=20
> 2012/5/5 Dan Wing <dwing@cisco.com>:
> >> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
> >> standarized for SIP regardless it has null impementation), will my
> SIP
> >> phone be able to *directly* talk in the media plane with a WebRTC
> >> browser? or not?
> >
> > That would work.
>=20
> Ok, so then, taking into account that SIP defines the usage of
> DTLS-SRTP in RFC 5763 (regardless no one device implements it), why
> does this WG assumes that "interop with non WebRTC endpoints will be
> made via *gateways*"?

Perhaps a more accurate statement would be:

  directly, if the remote endpoint supports ICE and supports
  DTLS-SRTP; otherwise, via a gateway.

> Let me please repeat my question: if my SIP phone implements RFC 5763,
> can my SIP phone directly interop at media plane with a WebRTC
> browser? (I know you already replied this, but I want to be very very
> sure) :)

Sure seems it should work.  Only nuance that I am aware of at the moment
is WEBRTC seems to want to require Consent Freshness (periodic ICE
connectivity checks) for the duration of the call.  This is okay, and
would work with an existing SIP phone -- so long as the SIP phone
continues to respond to ICE messages.  It should.  But I doubt that
is exercised in SIPIT testing.  The other nuance is that, because
doing the SHA1 for MESSAGE-INTEGRITY isn't needed for consent freshness,
there is desire to allow those periodic ICE connectivity checks to
omit the MESSAGE-INTEGRITY, which is a change to ICE.  See=20
draft-muthu-behave-consent-freshness.  Of course, that doesn't require
a gateway, but would require softening the implementation on the
phone.  draft-muthu-behave-consent-freshness might also add a new
SDP attribute, I dunno yet.

-d

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


From richard.ejzak@alcatel-lucent.com  Sun May  6 11:40:38 2012
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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sun, 6 May 2012 13:40:31 -0500
Thread-Topic: [rtcweb] Question about JSEP createOffer
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Subject: [rtcweb]  Question about JSEP createOffer
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I have a question about something I came across in the JSEP draft.

6.1.2 of JSEP draft 0 has the following text regarding "createOffer":


   As an offer, the generated SDP will contain the full set of

   capabilities supported by the session (as opposed to an answer, which

   will include only a specific negotiated subset to use); for each SDP

   line, the generation of the SDP must follow the appropriate process

   for generating an offer. In the event createOffer is called after the

   session is established, createOffer will generate an offer that is

   compatible with the current session, incorporating any changes that

   have been made to the session since the last complete offer-answer

   exchange, such as addition or removal of streams. If no changes have

   been made, the offer will be identical to the current local

   description.

The first and last sentences might conflict if the SDP Answerer later wants=
 to create a new Offer.  The local configuration based on the first offer/a=
nswer will contain only the selected media configuration, whereas it is des=
irable for the new Offer to contain the full set of capabilities of the (ne=
w) Offerer.  I think this text needs to change to reflect this case.  Am I =
missing something?

Richard

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<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>I have a question about something I came across in the J=
SEP
draft.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>6.1.2 of JSEP draft 0 has the following text regarding &=
#8220;createOffer&#8221;:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; As an offer, the generated SDP will contain the full s=
et
of<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; capabilities supported by the session (as opposed to a=
n
answer, which<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; will include only a specific negotiated subset to use)=
;
for each SDP<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; line, the generation of the SDP must follow the
appropriate process<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; for generating an offer. In the event createOffer is
called after the<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; session is established, createOffer will generate an o=
ffer
that is<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; compatible with the current session, incorporating any
changes that<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; have been made to the session since the last complete
offer-answer<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; exchange, such as addition or removal of streams. If n=
o
changes have<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; been made, the offer will be identical to the current
local<o:p></o:p></span></font></p>

<p class=3DMsoPlainText><font size=3D2 face=3D"Courier New"><span style=3D'=
font-size:
10.0pt'>&nbsp;&nbsp; description.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>The first and last sentences might conflict if the SDP
Answerer later wants to create a new Offer. &nbsp;The local configuration b=
ased
on the first offer/answer will contain only the selected media configuratio=
n,
whereas it is desirable for the new Offer to contain the full set of
capabilities of the (new) Offerer. &nbsp;I think this text needs to change =
to
reflect this case. &nbsp;Am I missing something?<o:p></o:p></span></font></=
p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Richard<o:p></o:p></span></font></p>

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sun, 6 May 2012 13:54:16 -0500
Thread-Topic: [rtcweb] Proposal for PeerConnection cloning
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Harald asked for a proposal for PeerConnection cloning, so here it goes:

I propose to create a new constructor "ClonePeerConnection" (CPC below) wit=
h the following semantics.


 1.  CPC will create a new PeerConnection object when successful.  The resu=
lting PC behaves just like any other PC except as described below.
 2.  CPC will take as input a reference to an existing PeerConnection (PC) =
object (no configuration or IceCallback parameters).
 3.  CPC can clone any other PC (including a cloned PC).  The sequence of c=
loning is not important and parents have no particular status different fro=
m any generation of clone.
 4.  CPC will fail if either the local description of the parent PC is not =
set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The pare=
nt PC can be in the OPENING or ACTIVE state.
 5.  The CPC object will inherit the local streams, local ICE candidates, a=
nd local description of the PC.
 6.  The remote streams and remote description for the CPC object will be s=
et to empty.
 7.  The CPC object will be set to the OPENING state to reflect that only t=
he local description is set.
 8.  The ICE states other than CLOSED and GATHERING will be handled indepen=
dently for each PC and its clones (as is true in standard forking scenarios=
).
 9.  The ICE state for this clone will be set to ICE_WAITING to reflect tha=
t all candidates are available but the remote configuration is not yet set.
 10. The PC and its clone(s) use a common pool of media resources.
 11. If the parent PC object has already released unused resources (final A=
NSWER), resources are reallocated as available to reflect the capabilities =
for each stream (as they would be reflected in a createOffer).
 12. The local streams might multicast toward the remote targets depending =
on the directionality attributes independently set for each PC and clone.
 13. The application should manage the directionality attributes for remote=
 streams from different targets to avoid resource conflicts.

CPC will be used primarily if forking is anticipated or actually occurs.  I=
t can also be used to clone a stable PC if desired for other reasons.  When=
 used for SIP forking, creation of the clone can be delayed until an ANSWER=
 actually arrives from a 2nd target as long as final ANSWER hasn't been app=
lied to the parent (PRANSWER is OK).  The parent always handles the first t=
arget; the first clone handles the 2nd target; etc.  The application can ev=
en try to clone for forking after the first ANSWER is applied to the parent=
 and resources are released, as long as the local description has not chang=
ed, at the risk that some resources needed for the 2nd target are no longer=
 available and must be renegotiated.

Comments on this proposal are welcome.

Richard


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Harald asked for a proposal for PeerConnection cloning, =
so
here it goes:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>I propose to create a new constructor &#8220;ClonePeerCo=
nnection&#8221;
(CPC below) with the following semantics.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<ol style=3D'margin-top:0in' start=3D1 type=3D1>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>CPC will create a new
     PeerConnection object when successful.&nbsp; The resulting PC behaves =
just
     like any other PC except as described below.<o:p></o:p></span></font><=
/li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>CPC will take as input a
     reference to an existing PeerConnection (PC) object (no configuration =
or IceCallback
     parameters).<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>CPC can clone any other P=
C
     (including a cloned PC).&nbsp; The sequence of cloning is not importan=
t
     and parents have no particular status different from any generation of
     clone.<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>CPC will fail if either t=
he
     local description of the parent PC is not set or the parent PC ICE is =
in
     ICE_CLOSED or ICE_GATHERING state.&nbsp; The parent PC can be in the
     OPENING or ACTIVE state.<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The CPC object will inher=
it the
     local streams, local ICE candidates, and local description of the PC.<=
o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The remote streams and re=
mote description
     for the CPC object will be set to empty.<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The CPC object will be se=
t to the
     OPENING state to reflect that only the local description is set.<o:p><=
/o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The ICE states other than
     CLOSED and GATHERING will be handled independently for each PC and its
     clones (as is true in standard forking scenarios).<o:p></o:p></span></=
font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The ICE state for this cl=
one
     will be set to ICE_WAITING to reflect that all candidates are availabl=
e
     but the remote configuration is not yet set.<o:p></o:p></span></font><=
/li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The PC and its clone(s) u=
se a
     common pool of media resources.<o:p></o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>If the parent PC object h=
as
     already released unused resources (final ANSWER), resources are
     reallocated as available to reflect the capabilities for each stream (=
as
     they would be reflected in a createOffer).<o:p></o:p></span></font></l=
i>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The local streams might
     multicast toward the remote targets depending on the directionality
     attributes independently set for each PC and clone.<o:p></o:p></span><=
/font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo1'><font size=3D2 fac=
e=3DArial><span
     style=3D'font-size:10.0pt;font-family:Arial'>The application should ma=
nage
     the directionality attributes for remote streams from different target=
s to
     avoid resource conflicts.<o:p></o:p></span></font></li>
</ol>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>CPC will be used primarily if forking is anticipated or
actually occurs. &nbsp;It can also be used to clone a stable PC if desired =
for
other reasons. &nbsp;When used for SIP forking, creation of the clone can b=
e
delayed until an ANSWER actually arrives from a 2<sup>nd</sup> target as lo=
ng
as final ANSWER hasn&#8217;t been applied to the parent (PRANSWER is OK). &=
nbsp;The
parent always handles the first target; the first clone handles the 2<sup>n=
d</sup>
target; etc. &nbsp;The application can even try to clone for forking after =
the
first ANSWER is applied to the parent and resources are released, as long a=
s
the local description has not changed, at the risk that some resources need=
ed
for the 2<sup>nd</sup> target are no longer available and must be renegotia=
ted.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Comments on this proposal are welcome.<o:p></o:p></span>=
</font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span style=3D'font-size:1=
0.0pt;
font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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From christer.holmberg@ericsson.com  Sun May  6 12:05:56 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: Sun, 6 May 2012 21:01:12 +0200
Thread-Topic: [rtcweb]  Proposal for PeerConnection cloning
Thread-Index: Ac0ruaKw/sIXfcxHRLuCOzit4TRF6gAAPhbN
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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Hi Richard,

Interesting proposal.

In bullet 11, you say that, if a final answer has been received (and unused=
 resources released), resources would have to be reallocated once a clone i=
s created.

A couple of questions:

1. Is there a need to allow cloning once a final answer has been received?

2. Once a final answer is received on a given PC, would it automatically re=
move all associated CPCs?

The basic question is really whether there is a need for clones once a fina=
l answer has been received. After all, there is a reason we call it "final"=
 :)

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Ejzak,=
 Richard P (Richard) [richard.ejzak@alcatel-lucent.com]
Sent: Sunday, May 06, 2012 9:54 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Proposal for PeerConnection cloning

Harald asked for a proposal for PeerConnection cloning, so here it goes:

I propose to create a new constructor =93ClonePeerConnection=94 (CPC below)=
 with the following semantics.


 1.  CPC will create a new PeerConnection object when successful.  The resu=
lting PC behaves just like any other PC except as described below.
 2.  CPC will take as input a reference to an existing PeerConnection (PC) =
object (no configuration or IceCallback parameters).
 3.  CPC can clone any other PC (including a cloned PC).  The sequence of c=
loning is not important and parents have no particular status different fro=
m any generation of clone.
 4.  CPC will fail if either the local description of the parent PC is not =
set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The pare=
nt PC can be in the OPENING or ACTIVE state.
 5.  The CPC object will inherit the local streams, local ICE candidates, a=
nd local description of the PC.
 6.  The remote streams and remote description for the CPC object will be s=
et to empty.
 7.  The CPC object will be set to the OPENING state to reflect that only t=
he local description is set.
 8.  The ICE states other than CLOSED and GATHERING will be handled indepen=
dently for each PC and its clones (as is true in standard forking scenarios=
).
 9.  The ICE state for this clone will be set to ICE_WAITING to reflect tha=
t all candidates are available but the remote configuration is not yet set.
 10. The PC and its clone(s) use a common pool of media resources.
 11. If the parent PC object has already released unused resources (final A=
NSWER), resources are reallocated as available to reflect the capabilities =
for each stream (as they would be reflected in a createOffer).
 12. The local streams might multicast toward the remote targets depending =
on the directionality attributes independently set for each PC and clone.
 13. The application should manage the directionality attributes for remote=
 streams from different targets to avoid resource conflicts.

CPC will be used primarily if forking is anticipated or actually occurs.  I=
t can also be used to clone a stable PC if desired for other reasons.  When=
 used for SIP forking, creation of the clone can be delayed until an ANSWER=
 actually arrives from a 2nd target as long as final ANSWER hasn=92t been a=
pplied to the parent (PRANSWER is OK).  The parent always handles the first=
 target; the first clone handles the 2nd target; etc.  The application can =
even try to clone for forking after the first ANSWER is applied to the pare=
nt and resources are released, as long as the local description has not cha=
nged, at the risk that some resources needed for the 2nd target are no long=
er available and must be renegotiated.

Comments on this proposal are welcome.

Richard


From richard.ejzak@alcatel-lucent.com  Sun May  6 13:40:43 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>
Date: Sun, 6 May 2012 15:40:35 -0500
Thread-Topic: [rtcweb]  Proposal for PeerConnection cloning
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Hi Christer,
Question 1: I agree that for parallel forking, as long as a PC is cloned be=
fore the unused parent resources are released, it is not necessary to allow=
 cloning later.  I would be ok to remove this feature.  I threw it in becau=
se it seemed not too expensive to implement (others have to comment on this=
) and it might be useful to allow delaying of cloning to the last possible =
instant (and for potentially other applications). =20

Question 2: I do not think we want to automatically remove all related PCs =
when one gets a final answer, else we could not support Partha's nested O/A=
 scenario.  I propose that this be done manually (so it is not mentioned in=
 the proposal).  This allows ICE and DTLS to progress independently with mu=
ltiple targets until a winner is selected and the others are closed.

BR, Richard

-----Original Message-----
From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
Sent: Sunday, May 06, 2012 2:01 PM
To: Ejzak, Richard P (Richard); rtcweb@ietf.org
Subject: RE: [rtcweb] Proposal for PeerConnection cloning

Hi Richard,

Interesting proposal.

In bullet 11, you say that, if a final answer has been received (and unused=
 resources released), resources would have to be reallocated once a clone i=
s created.

A couple of questions:

1. Is there a need to allow cloning once a final answer has been received?

2. Once a final answer is received on a given PC, would it automatically re=
move all associated CPCs?

The basic question is really whether there is a need for clones once a fina=
l answer has been received. After all, there is a reason we call it "final"=
 :)

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Ejzak,=
 Richard P (Richard) [richard.ejzak@alcatel-lucent.com]
Sent: Sunday, May 06, 2012 9:54 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Proposal for PeerConnection cloning

Harald asked for a proposal for PeerConnection cloning, so here it goes:

I propose to create a new constructor "ClonePeerConnection" (CPC below) wit=
h the following semantics.


 1.  CPC will create a new PeerConnection object when successful.  The resu=
lting PC behaves just like any other PC except as described below.
 2.  CPC will take as input a reference to an existing PeerConnection (PC) =
object (no configuration or IceCallback parameters).
 3.  CPC can clone any other PC (including a cloned PC).  The sequence of c=
loning is not important and parents have no particular status different fro=
m any generation of clone.
 4.  CPC will fail if either the local description of the parent PC is not =
set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The pare=
nt PC can be in the OPENING or ACTIVE state.
 5.  The CPC object will inherit the local streams, local ICE candidates, a=
nd local description of the PC.
 6.  The remote streams and remote description for the CPC object will be s=
et to empty.
 7.  The CPC object will be set to the OPENING state to reflect that only t=
he local description is set.
 8.  The ICE states other than CLOSED and GATHERING will be handled indepen=
dently for each PC and its clones (as is true in standard forking scenarios=
).
 9.  The ICE state for this clone will be set to ICE_WAITING to reflect tha=
t all candidates are available but the remote configuration is not yet set.
 10. The PC and its clone(s) use a common pool of media resources.
 11. If the parent PC object has already released unused resources (final A=
NSWER), resources are reallocated as available to reflect the capabilities =
for each stream (as they would be reflected in a createOffer).
 12. The local streams might multicast toward the remote targets depending =
on the directionality attributes independently set for each PC and clone.
 13. The application should manage the directionality attributes for remote=
 streams from different targets to avoid resource conflicts.

CPC will be used primarily if forking is anticipated or actually occurs.  I=
t can also be used to clone a stable PC if desired for other reasons.  When=
 used for SIP forking, creation of the clone can be delayed until an ANSWER=
 actually arrives from a 2nd target as long as final ANSWER hasn't been app=
lied to the parent (PRANSWER is OK).  The parent always handles the first t=
arget; the first clone handles the 2nd target; etc.  The application can ev=
en try to clone for forking after the first ANSWER is applied to the parent=
 and resources are released, as long as the local description has not chang=
ed, at the risk that some resources needed for the 2nd target are no longer=
 available and must be renegotiated.

Comments on this proposal are welcome.

Richard


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Date: Sun, 6 May 2012 15:19:57 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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--0015174c43aabb2dbe04bf658d2f
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Such a complicated process would not be necessary if the PeerConnection
object were not home to two largely independent state machines.

Nor would it be necessary if this scenario were considered out of scope.

Certainly, if you consider the scenario interesting enough to do something
about (I don't), then you news some support from the browser in setting up
two ICE associations on the same local port, but that is easier without
having to worry about the state of the media negotiation.

Let's assume you could have transport and media separate. Then what you
need is a clone operation for the transport object. You can simply start
the media negotiation, as if from scratch on the cloned transport.

The state of the media has no real bearing on this then, and Christer's
concern becomes irrelevant.  ...Though it is certainly valid given the
current proposal.

Other considerations:
-does the clone get a copy of local streams if they are added to the
original? I'm guessing not.
-the transition from PR_ANSWER to OFFER will be difficult given that this
is the obvious way to implement this
-how does the application learn that cloning fails due to lack of
resources? Normally, an offer can't include capabilities that cannot be
provided, but this implies it.

An alternative, within the constraints offered by JSEP, is to create a new
PeerConfection and bind it to an existing one. What reasons can you offer
for avoiding this choice?

--Martin
 On May 6, 2012 11:54 AM, "Ejzak, Richard P (Richard)" <
richard.ejzak@alcatel-lucent.com> wrote:

>  Harald asked for a proposal for PeerConnection cloning, so here it goes:=
*
> ***
>
> ** **
>
> I propose to create a new constructor =E2=80=9CClonePeerConnection=E2=80=
=9D (CPC below)
> with the following semantics.****
>
> ** **
>
>    1. CPC will create a new PeerConnection object when successful.  The
>    resulting PC behaves just like any other PC except as described below.=
*
>    ***
>    2. CPC will take as input a reference to an existing PeerConnection
>    (PC) object (no configuration or IceCallback parameters).****
>    3. CPC can clone any other PC (including a cloned PC).  The sequence
>    of cloning is not important and parents have no particular status diff=
erent
>    from any generation of clone.****
>    4. CPC will fail if either the local description of the parent PC is
>    not set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state. =
 The
>    parent PC can be in the OPENING or ACTIVE state.****
>    5. The CPC object will inherit the local streams, local ICE
>    candidates, and local description of the PC.****
>    6. The remote streams and remote description for the CPC object will
>    be set to empty.****
>    7. The CPC object will be set to the OPENING state to reflect that
>    only the local description is set.****
>    8. The ICE states other than CLOSED and GATHERING will be handled
>    independently for each PC and its clones (as is true in standard forki=
ng
>    scenarios).****
>    9. The ICE state for this clone will be set to ICE_WAITING to reflect
>    that all candidates are available but the remote configuration is not =
yet
>    set.****
>    10. The PC and its clone(s) use a common pool of media resources.****
>    11. If the parent PC object has already released unused resources
>    (final ANSWER), resources are reallocated as available to reflect the
>    capabilities for each stream (as they would be reflected in a createOf=
fer).
>    ****
>    12. The local streams might multicast toward the remote targets
>    depending on the directionality attributes independently set for each =
PC
>    and clone.****
>    13. The application should manage the directionality attributes for
>    remote streams from different targets to avoid resource conflicts.****
>
> ** **
>
> CPC will be used primarily if forking is anticipated or actually occurs.
>  It can also be used to clone a stable PC if desired for other reasons.
>  When used for SIP forking, creation of the clone can be delayed until an
> ANSWER actually arrives from a 2nd target as long as final ANSWER hasn=E2=
=80=99t
> been applied to the parent (PRANSWER is OK).  The parent always handles t=
he
> first target; the first clone handles the 2nd target; etc.  The
> application can even try to clone for forking after the first ANSWER is
> applied to the parent and resources are released, as long as the local
> description has not changed, at the risk that some resources needed for t=
he
> 2nd target are no longer available and must be renegotiated.****
>
> ** **
>
> Comments on this proposal are welcome.****
>
> ** **
>
> Richard****
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--0015174c43aabb2dbe04bf658d2f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p>Such a complicated process would not be necessary if the PeerConnection =
object were not home to two largely independent state machines.</p>
<p>Nor would it be necessary if this scenario were considered out of scope.=
</p>
<p>Certainly, if you consider the scenario interesting enough to do somethi=
ng about (I don&#39;t), then you news some support from the browser in sett=
ing up two ICE associations on the same local port, but that is easier with=
out having to worry about the state of the media negotiation.</p>

<p>Let&#39;s assume you could have transport and media separate. Then what =
you need is a clone operation for the transport object. You can simply star=
t the media negotiation, as if from scratch on the cloned transport.</p>

<p>The state of the media has no real bearing on this then, and Christer&#3=
9;s concern becomes irrelevant.=C2=A0 ...Though it is certainly valid given=
 the current proposal.</p>
<p>Other considerations:<br>
-does the clone get a copy of local streams if they are added to the origin=
al? I&#39;m guessing not.<br>
-the transition from PR_ANSWER to OFFER will be difficult given that this i=
s the obvious way to implement this<br>
-how does the application learn that cloning fails due to lack of resources=
? Normally, an offer can&#39;t include capabilities that cannot be provided=
, but this implies it.</p>
<p>An alternative, within the constraints offered by JSEP, is to create a n=
ew PeerConfection and bind it to an existing one. What reasons can you offe=
r for avoiding this choice?</p>
<p>--Martin<br>
</p>
<div class=3D"gmail_quote">On May 6, 2012 11:54 AM, &quot;Ejzak, Richard P =
(Richard)&quot; &lt;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com">ric=
hard.ejzak@alcatel-lucent.com</a>&gt; wrote:<br type=3D"attribution"><block=
quote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc=
 solid;padding-left:1ex">









<div lang=3D"EN-US" link=3D"blue" vlink=3D"#606420">

<div>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">Harald asked for a proposal for PeerConnection cloning,=
 so
here it goes:<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">I propose to create a new constructor =E2=80=9CClonePee=
rConnection=E2=80=9D
(CPC below) with the following semantics.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<ol style=3D"margin-top:0in" start=3D"1" type=3D"1">
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">CPC will create a new
     PeerConnection object when successful.=C2=A0 The resulting PC behaves =
just
     like any other PC except as described below.<u></u><u></u></span></fon=
t></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">CPC will take as input a
     reference to an existing PeerConnection (PC) object (no configuration =
or IceCallback
     parameters).<u></u><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">CPC can clone any other PC
     (including a cloned PC).=C2=A0 The sequence of cloning is not importan=
t
     and parents have no particular status different from any generation of
     clone.<u></u><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">CPC will fail if either the
     local description of the parent PC is not set or the parent PC ICE is =
in
     ICE_CLOSED or ICE_GATHERING state.=C2=A0 The parent PC can be in the
     OPENING or ACTIVE state.<u></u><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The CPC object will inherit the
     local streams, local ICE candidates, and local description of the PC.<=
u></u><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The remote streams and remote description
     for the CPC object will be set to empty.<u></u><u></u></span></font></=
li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The CPC object will be set to the
     OPENING state to reflect that only the local description is set.<u></u=
><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The ICE states other than
     CLOSED and GATHERING will be handled independently for each PC and its
     clones (as is true in standard forking scenarios).<u></u><u></u></span=
></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The ICE state for this clone
     will be set to ICE_WAITING to reflect that all candidates are availabl=
e
     but the remote configuration is not yet set.<u></u><u></u></span></fon=
t></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The PC and its clone(s) use a
     common pool of media resources.<u></u><u></u></span></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">If the parent PC object has
     already released unused resources (final ANSWER), resources are
     reallocated as available to reflect the capabilities for each stream (=
as
     they would be reflected in a createOffer).<u></u><u></u></span></font>=
</li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The local streams might
     multicast toward the remote targets depending on the directionality
     attributes independently set for each PC and clone.<u></u><u></u></spa=
n></font></li>
 <li class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0=
pt;font-family:Arial">The application should manage
     the directionality attributes for remote streams from different target=
s to
     avoid resource conflicts.<u></u><u></u></span></font></li>
</ol>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">CPC will be used primarily if forking is anticipated or
actually occurs. =C2=A0It can also be used to clone a stable PC if desired =
for
other reasons. =C2=A0When used for SIP forking, creation of the clone can b=
e
delayed until an ANSWER actually arrives from a 2<sup>nd</sup> target as lo=
ng
as final ANSWER hasn=E2=80=99t been applied to the parent (PRANSWER is OK).=
 =C2=A0The
parent always handles the first target; the first clone handles the 2<sup>n=
d</sup>
target; etc. =C2=A0The application can even try to clone for forking after =
the
first ANSWER is applied to the parent and resources are released, as long a=
s
the local description has not changed, at the risk that some resources need=
ed
for the 2<sup>nd</sup> target are no longer available and must be renegotia=
ted.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">Comments on this proposal are welcome.<u></u><u></u></s=
pan></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">Richard<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

</div>

</div>


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

--0015174c43aabb2dbe04bf658d2f--

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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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Hi Martin,
I'm not sure that separating the ICE and media state machines would help ve=
ry much.  ICE has to handle all the streams and components of a PeerConnect=
ion at the same time anyway.

Once cloning occurs, operations on them (in my proposal) are independent ex=
cept in the sharing of resources allocated to the parent at the time of clo=
ning.

If we accept Christer's simplification then we should never need to fail th=
e cloning operation due to lack of resources.  I didn't think that reportin=
g failure would be a problem (if necessary).

You suggest that creating a separate PeerConnection and "binding" it to the=
 parent is somehow a better alternative for cloning.  I look forward to rev=
iewing an alternative proposal with sufficient detail to make a comparison =
(as Harald challenged me to do).  I took my best shot and certainly didn't =
"avoid" any alternatives that I was aware of and thought might be superior.

Richard

________________________________
From: Martin Thomson [mailto:martin.thomson@gmail.com]
Sent: Sunday, May 06, 2012 5:20 PM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Proposal for PeerConnection cloning


Such a complicated process would not be necessary if the PeerConnection obj=
ect were not home to two largely independent state machines.

Nor would it be necessary if this scenario were considered out of scope.

Certainly, if you consider the scenario interesting enough to do something =
about (I don't), then you news some support from the browser in setting up =
two ICE associations on the same local port, but that is easier without hav=
ing to worry about the state of the media negotiation.

Let's assume you could have transport and media separate. Then what you nee=
d is a clone operation for the transport object. You can simply start the m=
edia negotiation, as if from scratch on the cloned transport.

The state of the media has no real bearing on this then, and Christer's con=
cern becomes irrelevant.  ...Though it is certainly valid given the current=
 proposal.

Other considerations:
-does the clone get a copy of local streams if they are added to the origin=
al? I'm guessing not.
-the transition from PR_ANSWER to OFFER will be difficult given that this i=
s the obvious way to implement this
-how does the application learn that cloning fails due to lack of resources=
? Normally, an offer can't include capabilities that cannot be provided, bu=
t this implies it.

An alternative, within the constraints offered by JSEP, is to create a new =
PeerConfection and bind it to an existing one. What reasons can you offer f=
or avoiding this choice?

--Martin
On May 6, 2012 11:54 AM, "Ejzak, Richard P (Richard)" <richard.ejzak@alcate=
l-lucent.com<mailto:richard.ejzak@alcatel-lucent.com>> wrote:
Harald asked for a proposal for PeerConnection cloning, so here it goes:

I propose to create a new constructor "ClonePeerConnection" (CPC below) wit=
h the following semantics.


 1.  CPC will create a new PeerConnection object when successful.  The resu=
lting PC behaves just like any other PC except as described below.
 2.  CPC will take as input a reference to an existing PeerConnection (PC) =
object (no configuration or IceCallback parameters).
 3.  CPC can clone any other PC (including a cloned PC).  The sequence of c=
loning is not important and parents have no particular status different fro=
m any generation of clone.
 4.  CPC will fail if either the local description of the parent PC is not =
set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The pare=
nt PC can be in the OPENING or ACTIVE state.
 5.  The CPC object will inherit the local streams, local ICE candidates, a=
nd local description of the PC.
 6.  The remote streams and remote description for the CPC object will be s=
et to empty.
 7.  The CPC object will be set to the OPENING state to reflect that only t=
he local description is set.
 8.  The ICE states other than CLOSED and GATHERING will be handled indepen=
dently for each PC and its clones (as is true in standard forking scenarios=
).
 9.  The ICE state for this clone will be set to ICE_WAITING to reflect tha=
t all candidates are available but the remote configuration is not yet set.
 10. The PC and its clone(s) use a common pool of media resources.
 11. If the parent PC object has already released unused resources (final A=
NSWER), resources are reallocated as available to reflect the capabilities =
for each stream (as they would be reflected in a createOffer).
 12. The local streams might multicast toward the remote targets depending =
on the directionality attributes independently set for each PC and clone.
 13. The application should manage the directionality attributes for remote=
 streams from different targets to avoid resource conflicts.

CPC will be used primarily if forking is anticipated or actually occurs.  I=
t can also be used to clone a stable PC if desired for other reasons.  When=
 used for SIP forking, creation of the clone can be delayed until an ANSWER=
 actually arrives from a 2nd target as long as final ANSWER hasn't been app=
lied to the parent (PRANSWER is OK).  The parent always handles the first t=
arget; the first clone handles the 2nd target; etc.  The application can ev=
en try to clone for forking after the first ANSWER is applied to the parent=
 and resources are released, as long as the local description has not chang=
ed, at the risk that some resources needed for the 2nd target are no longer=
 available and must be renegotiated.

Comments on this proposal are welcome.

Richard


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

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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Hi Martin,<o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>I&#8217;m not sure that separating the=
 ICE
and media state machines would help very much. &nbsp;ICE has to handle all =
the streams
and components of a PeerConnection at the same time anyway.<o:p></o:p></spa=
n></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Once cloning occurs, operations on the=
m (in
my proposal) are independent except in the sharing of resources allocated t=
o
the parent at the time of cloning.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>If we accept Christer&#8217;s
simplification then we should never need to fail the cloning operation due =
to
lack of resources.&nbsp; I didn&#8217;t think that reporting failure would =
be a
problem (if necessary).<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>You suggest that creating a separate
PeerConnection and &#8220;binding&#8221; it to the parent is somehow a bett=
er alternative
for cloning. &nbsp;I look forward to reviewing an alternative proposal with
sufficient detail to make a comparison (as Harald challenged me to do).&nbs=
p; I
took my best shot and certainly didn&#8217;t &#8220;avoid&#8221; any
alternatives that I was aware of and thought might be superior.<o:p></o:p><=
/span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Martin T=
homson
[mailto:martin.thomson@gmail.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Sunday, May 06, 2012 5=
:20 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] Propos=
al for
PeerConnection cloning</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Such a
complicated process would not be necessary if the PeerConnection object wer=
e
not home to two largely independent state machines.<o:p></o:p></span></font=
></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Nor would
it be necessary if this scenario were considered out of scope.<o:p></o:p></=
span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Certainly,
if you consider the scenario interesting enough to do something about (I
don't), then you news some support from the browser in setting up two ICE
associations on the same local port, but that is easier without having to w=
orry
about the state of the media negotiation.<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Let's
assume you could have transport and media separate. Then what you need is a
clone operation for the transport object. You can simply start the media
negotiation, as if from scratch on the cloned transport.<o:p></o:p></span><=
/font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>The state
of the media has no real bearing on this then, and Christer's concern becom=
es
irrelevant.&nbsp; ...Though it is certainly valid given the current proposa=
l.<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>Other
considerations:<br>
-does the clone get a copy of local streams if they are added to the origin=
al?
I'm guessing not.<br>
-the transition from PR_ANSWER to OFFER will be difficult given that this i=
s
the obvious way to implement this<br>
-how does the application learn that cloning fails due to lack of resources=
?
Normally, an offer can't include capabilities that cannot be provided, but =
this
implies it.<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>An
alternative, within the constraints offered by JSEP, is to create a new
PeerConfection and bind it to an existing one. What reasons can you offer f=
or
avoiding this choice?<o:p></o:p></span></font></p>

<p><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'=
>--Martin<o:p></o:p></span></font></p>

<div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'>On May 6, 2012 11:54 AM, &quot;Ejzak, Richard P (Richard)&quot; &lt=
;<a
href=3D"mailto:richard.ejzak@alcatel-lucent.com">richard.ejzak@alcatel-luce=
nt.com</a>&gt;
wrote:<o:p></o:p></span></font></p>

<div link=3Dblue vlink=3D"#606420">

<div>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>Ha=
rald asked
for a proposal for PeerConnection cloning, so here it goes:</span></font><o=
:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>I =
propose to
create a new constructor &#8220;ClonePeerConnection&#8221; (CPC below) with=
 the
following semantics.</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

<ol start=3D1 type=3D1>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>CPC will create a new PeerConnection object =
when
     successful.&nbsp; The resulting PC behaves just like any other PC exce=
pt
     as described below.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>CPC will take as input a reference to an
     existing PeerConnection (PC) object (no configuration or IceCallback p=
arameters).</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>CPC can clone any other PC (including a clon=
ed
     PC).&nbsp; The sequence of cloning is not important and parents have n=
o
     particular status different from any generation of clone.</span></font=
><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>CPC will fail if either the local descriptio=
n of
     the parent PC is not set or the parent PC ICE is in ICE_CLOSED or
     ICE_GATHERING state.&nbsp; The parent PC can be in the OPENING or ACTI=
VE
     state.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The CPC object will inherit the local stream=
s,
     local ICE candidates, and local description of the PC.</span></font><o=
:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The remote streams and remote description fo=
r
     the CPC object will be set to empty.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The CPC object will be set to the OPENING st=
ate
     to reflect that only the local description is set.</span></font><o:p><=
/o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The ICE states other than CLOSED and GATHERI=
NG
     will be handled independently for each PC and its clones (as is true i=
n
     standard forking scenarios).</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The ICE state for this clone will be set to
     ICE_WAITING to reflect that all candidates are available but the remot=
e
     configuration is not yet set.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The PC and its clone(s) use a common pool of
     media resources.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>If the parent PC object has already released
     unused resources (final ANSWER), resources are reallocated as availabl=
e to
     reflect the capabilities for each stream (as they would be reflected i=
n a
     createOffer).</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The local streams might multicast toward the
     remote targets depending on the directionality attributes independentl=
y
     set for each PC and clone.</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-a=
lt:auto;
     mso-list:l0 level1 lfo1'><font size=3D2 face=3DArial><span style=3D'fo=
nt-size:
     10.0pt;font-family:Arial'>The application should manage the directiona=
lity
     attributes for remote streams from different targets to avoid resource
     conflicts.</span></font><o:p></o:p></li>
</ol>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>CP=
C will be
used primarily if forking is anticipated or actually occurs. &nbsp;It can a=
lso
be used to clone a stable PC if desired for other reasons. &nbsp;When used =
for
SIP forking, creation of the clone can be delayed until an ANSWER actually
arrives from a 2<sup>nd</sup> target as long as final ANSWER hasn&#8217;t b=
een
applied to the parent (PRANSWER is OK). &nbsp;The parent always handles the
first target; the first clone handles the 2<sup>nd</sup> target; etc. &nbsp=
;The
application can even try to clone for forking after the first ANSWER is app=
lied
to the parent and resources are released, as long as the local description =
has
not changed, at the risk that some resources needed for the 2<sup>nd</sup>
target are no longer available and must be renegotiated.</span></font><o:p>=
</o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>Co=
mments on
this proposal are welcome.</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>Ri=
chard</span></font><o:p></o:p></p>

<p class=3DMsoNormal style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt=
:auto'><font
size=3D2 face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>&n=
bsp;</span></font><o:p></o:p></p>

</div>

</div>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><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><o:p></o:p></span></font></=
p>

</div>

</div>

</body>

</html>

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From randell-ietf@jesup.org  Sun May  6 22:34:20 2012
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On 5/6/2012 4:40 PM, Ejzak, Richard P (Richard) wrote:
> Hi Christer,
> Question 1: I agree that for parallel forking, as long as a PC is cloned before the unused parent resources are released, it is not necessary to allow cloning later.  I would be ok to remove this feature.  I threw it in because it seemed not too expensive to implement (others have to comment on this) and it might be useful to allow delaying of cloning to the last possible instant (and for potentially other applications).

> Question 2: I do not think we want to automatically remove all related PCs when one gets a final answer, else we could not support Partha's nested O/A scenario.  I propose that this be done manually (so it is not mentioned in the proposal).  This allows ICE and DTLS to progress independently with multiple targets until a winner is selected and the others are closed.

If you want to support the usecase I suggested, where a single offer can 
result in parallel forking, but where you don't want to lock down to a 
single answerer: then you obviously need to support forking, and likely 
you need/want to support it irrespective of the 'final' state of the 
original PeerConnection.  You *can* require cloning before the final 
answer is applied, though I would argue that you don't gain much by 
doing so.

So this is the usecase: you send a single offer (say to a conference 
server controller, or to a game server) and you get a separate answer 
from each other endpoint, such as in a mesh conference, or who are in 
the same 'area' of the game, or in hearing (at that moment) in a VR 
environment.  You might get these answers well after the initial answer 
or burst of answers.

Using CPC, the sequence would be to:  ClonePeerConnection(), apply the 
new answer to the new CPC-created PeerConnection.  Nothing requires you 
to do so; you can reject the new answerer because you have a final 
already (or if the PeerConnection was made speculatively, you could 
close them if any other one of the clones received an answer).  I 
*think* that CPC will reduce the amount of ICE traffic (in some cases) 
and the number of external ports used (depending on NATs), and therefore 
might reduce call setup-time overhead.

Without CPC, the server will need to be able to ask endpoints to 
generate offers in anticipation of using later, or even immediately 
("send me an offer *now* to forward to the new user, or "send me 8 more 
offers just like that one to send of existing users").

p.s. Thanks for the concrete proposal about how to handle this!  We've 
been talking about it (on occasion) for a year-ish, but no one has 
proposed an actual set of semantics for it.

>
> BR, Richard
>
> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, May 06, 2012 2:01 PM
> To: Ejzak, Richard P (Richard); rtcweb@ietf.org
> Subject: RE: [rtcweb] Proposal for PeerConnection cloning
>
> Hi Richard,
>
> Interesting proposal.
>
> In bullet 11, you say that, if a final answer has been received (and unused resources released), resources would have to be reallocated once a clone is created.
>
> A couple of questions:
>
> 1. Is there a need to allow cloning once a final answer has been received?
>
> 2. Once a final answer is received on a given PC, would it automatically remove all associated CPCs?
>
> The basic question is really whether there is a need for clones once a final answer has been received. After all, there is a reason we call it "final" :)
>
> Regards,
>
> Christer
>
> ________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Ejzak, Richard P (Richard) [richard.ejzak@alcatel-lucent.com]
> Sent: Sunday, May 06, 2012 9:54 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Proposal for PeerConnection cloning
>
> Harald asked for a proposal for PeerConnection cloning, so here it goes:
>
> I propose to create a new constructor "ClonePeerConnection" (CPC below) with the following semantics.
>
>
>   1.  CPC will create a new PeerConnection object when successful.  The resulting PC behaves just like any other PC except as described below.
>   2.  CPC will take as input a reference to an existing PeerConnection (PC) object (no configuration or IceCallback parameters).
>   3.  CPC can clone any other PC (including a cloned PC).  The sequence of cloning is not important and parents have no particular status different from any generation of clone.
>   4.  CPC will fail if either the local description of the parent PC is not set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The parent PC can be in the OPENING or ACTIVE state.
>   5.  The CPC object will inherit the local streams, local ICE candidates, and local description of the PC.
>   6.  The remote streams and remote description for the CPC object will be set to empty.
>   7.  The CPC object will be set to the OPENING state to reflect that only the local description is set.
>   8.  The ICE states other than CLOSED and GATHERING will be handled independently for each PC and its clones (as is true in standard forking scenarios).
>   9.  The ICE state for this clone will be set to ICE_WAITING to reflect that all candidates are available but the remote configuration is not yet set.
>   10. The PC and its clone(s) use a common pool of media resources.
>   11. If the parent PC object has already released unused resources (final ANSWER), resources are reallocated as available to reflect the capabilities for each stream (as they would be reflected in a createOffer).
>   12. The local streams might multicast toward the remote targets depending on the directionality attributes independently set for each PC and clone.
>   13. The application should manage the directionality attributes for remote streams from different targets to avoid resource conflicts.
>
> CPC will be used primarily if forking is anticipated or actually occurs.  It can also be used to clone a stable PC if desired for other reasons.  When used for SIP forking, creation of the clone can be delayed until an ANSWER actually arrives from a 2nd target as long as final ANSWER hasn't been applied to the parent (PRANSWER is OK).  The parent always handles the first target; the first clone handles the 2nd target; etc.  The application can even try to clone for forking after the first ANSWER is applied to the parent and resources are released, as long as the local description has not changed, at the risk that some resources needed for the 2nd target are no longer available and must be renegotiated.
>
> Comments on this proposal are welcome.
>
> Richard

-- 
Randell Jesup
randell-ietf@jesup.org


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 7 May 2012 08:22:03 +0200
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Hi,

>> Hi Christer,
>> Question 1: I agree that for parallel forking, as long as a PC is cloned=
 before the unused parent resources are
>> released, it is not necessary to allow cloning later.  I would be ok to =
remove this feature.  I threw it in because=20
>> it seemed not too expensive to implement (others have to comment on this=
) and it might be useful to allow
>> delaying of cloning to the last possible instant (and for potentially ot=
her applications).
>>
>> Question 2: I do not think we want to automatically remove all related P=
Cs when one gets a final answer, else we
>> could not support Partha's nested O/A scenario.  I propose that this be =
done manually (so it is not mentioned in the
>> proposal).  This allows  ICE and DTLS to progress independently with mul=
tiple targets until a winner is selected=20
>> and the others are closed.
>
> If you want to support the usecase I suggested, where a single offer can =
result in parallel forking, but where you don't
> want to lock down to a single answerer: then you obviously need to suppor=
t forking, and likely you need/want to support
> it irrespective of the 'final' state of the original PeerConnection.  You=
 *can* require cloning before the final answer is=20
> applied, though I would argue that you don't gain much by doing so.
>
> So this is the usecase: you send a single offer (say to a conference serv=
er controller, or to a game server) and you get=20
> a separate answer from each other endpoint, such as in a mesh conference,=
 or who are in the same 'area' of the game,=20
> or in hearing (at that moment) in a VR environment.  You might get these =
answers well after the initial answer or burst=20
> of answers.

Yes. And, my idea would be that each of those answers are provided to the b=
rowser as PRE_ANSWERS.

Then, at some point, you decide that you are not going to accept any more a=
nswers, and you provide an ANSWER to one of the (C)PCs, which will release =
all other associated (C)PCs.

Regards,

Christer



> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]
> Sent: Sunday, May 06, 2012 2:01 PM
> To: Ejzak, Richard P (Richard); rtcweb@ietf.org
> Subject: RE: [rtcweb] Proposal for PeerConnection cloning
>
> Hi Richard,
>
> Interesting proposal.
>
> In bullet 11, you say that, if a final answer has been received (and unus=
ed resources released), resources would have to be reallocated once a clone=
 is created.
>
> A couple of questions:
>
> 1. Is there a need to allow cloning once a final answer has been received=
?
>
> 2. Once a final answer is received on a given PC, would it automatically =
remove all associated CPCs?
>
> The basic question is really whether there is a need for clones once a fi=
nal answer has been received. After all, there is a reason we call it "fina=
l" :)
>
> Regards,
>
> Christer
>
> ________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Ejza=
k, Richard P (Richard) [richard.ejzak@alcatel-lucent.com]
> Sent: Sunday, May 06, 2012 9:54 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Proposal for PeerConnection cloning
>
> Harald asked for a proposal for PeerConnection cloning, so here it goes:
>
> I propose to create a new constructor "ClonePeerConnection" (CPC below) w=
ith the following semantics.
>
>
>   1.  CPC will create a new PeerConnection object when successful.  The r=
esulting PC behaves just like any other PC except as described below.
>   2.  CPC will take as input a reference to an existing PeerConnection (P=
C) object (no configuration or IceCallback parameters).
>   3.  CPC can clone any other PC (including a cloned PC).  The sequence o=
f cloning is not important and parents have no particular status different =
from any generation of clone.
>   4.  CPC will fail if either the local description of the parent PC is n=
ot set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The p=
arent PC can be in the OPENING or ACTIVE state.
>   5.  The CPC object will inherit the local streams, local ICE candidates=
, and local description of the PC.
>   6.  The remote streams and remote description for the CPC object will b=
e set to empty.
>   7.  The CPC object will be set to the OPENING state to reflect that onl=
y the local description is set.
>   8.  The ICE states other than CLOSED and GATHERING will be handled inde=
pendently for each PC and its clones (as is true in standard forking scenar=
ios).
>   9.  The ICE state for this clone will be set to ICE_WAITING to reflect =
that all candidates are available but the remote configuration is not yet s=
et.
>   10. The PC and its clone(s) use a common pool of media resources.
>   11. If the parent PC object has already released unused resources (fina=
l ANSWER), resources are reallocated as available to reflect the capabiliti=
es for each stream (as they would be reflected in a createOffer).
>   12. The local streams might multicast toward the remote targets dependi=
ng on the directionality attributes independently set for each PC and clone=
.
>   13. The application should manage the directionality attributes for rem=
ote streams from different targets to avoid resource conflicts.
>
> CPC will be used primarily if forking is anticipated or actually occurs. =
 It can also be used to clone a stable PC if desired for other reasons.  Wh=
en used for SIP forking, creation of the clone can be delayed until an ANSW=
ER actually arrives from a 2nd target as long as final ANSWER hasn't been a=
pplied to the parent (PRANSWER is OK).  The parent always handles the first=
 target; the first clone handles the 2nd target; etc.  The application can =
even try to clone for forking after the first ANSWER is applied to the pare=
nt and resources are released, as long as the local description has not cha=
nged, at the risk that some resources needed for the 2nd target are no long=
er available and must be renegotiated.
>
> Comments on this proposal are welcome.
>
> Richard

--=20
Randell Jesup
randell-ietf@jesup.org

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

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On 5/7/2012 2:22 AM, Christer Holmberg wrote:
>>
>> If you want to support the usecase I suggested, where a single offer can result in parallel forking, but where you don't
>> want to lock down to a single answerer: then you obviously need to support forking, and likely you need/want to support
>> it irrespective of the 'final' state of the original PeerConnection.  You *can* require cloning before the final answer is
>> applied, though I would argue that you don't gain much by doing so.
>>
>> So this is the usecase: you send a single offer (say to a conference server controller, or to a game server) and you get
>> a separate answer from each other endpoint, such as in a mesh conference, or who are in the same 'area' of the game,
>> or in hearing (at that moment) in a VR environment.  You might get these answers well after the initial answer or burst
>> of answers.
> Yes. And, my idea would be that each of those answers are provided to the browser as PRE_ANSWERS.
>
> Then, at some point, you decide that you are not going to accept any more answers, and you provide an ANSWER to one of the (C)PCs, which will release all other associated (C)PCs.

Sorry, I think you missed the point: the application wants to end up 
with N connected, active PeerConnections, not one.  This is a mesh 
conference (for example), not a central-mixer conference.  And for added 
fun, we don't necessarily know there won't be more coming later, perhaps 
much later (but that's a possibly-separable feature).

-- 
Randell Jesup
randell-ietf@jesup.org


From christer.holmberg@ericsson.com  Mon May  7 00:08:32 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 7 May 2012 09:08:27 +0200
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Hi,

>>> If you want to support the usecase I suggested, where a single offer=20
>>> can result in parallel forking, but where you don't want to lock down=20
>>> to a single answerer: then you obviously need to support forking, and=20
>>> likely you need/want to support it irrespective of the 'final' state of=
 the original PeerConnection.  You *can* require cloning before the final a=
nswer is applied, though I would argue that you don't gain much by doing so=
.
>>>
>>> So this is the usecase: you send a single offer (say to a conference=20
>>> server controller, or to a game server) and you get a separate answer=20
>>> from each other endpoint, such as in a mesh conference, or who are in=20
>>> the same 'area' of the game, or in hearing (at that moment) in a VR env=
ironment.  You might get these answers well after the initial answer or bur=
st of answers.
>> Yes. And, my idea would be that each of those answers are provided to th=
e browser as PRE_ANSWERS.
>>
>> Then, at some point, you decide that you are not going to accept any mor=
e answers, and you provide an ANSWER to one of the (C)PCs, which will relea=
se all other associated (C)PCs.
>
> Sorry, I think you missed the point: the application wants to end up with=
 N connected, active PeerConnections, not one.  This is a=20
> mesh conference (for example), not a central-mixer conference.  And for a=
dded fun, we don't necessarily know there won't be more=20
> coming later, perhaps much later (but that's a possibly-separable feature=
).

Speaking in SIP terms, I don't think it is possible to implement mesh confe=
rence that way, because clients will normally, once they've received one 20=
0 OK, ACK+BYE any additional ones. Or, a forking proxy will CANCEL all othe=
r forked legs once a 200 OK has been forwarded on one of them.

Regards,

Christer

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On 5/7/2012 3:08 AM, Christer Holmberg wrote:
> Hi,
>
>>>> If you want to support the usecase I suggested, where a single offer
>>>> can result in parallel forking, but where you don't want to lock down
>>>> to a single answerer: then you obviously need to support forking, and
>>>> likely you need/want to support it irrespective of the 'final' state of the original PeerConnection.  You *can* require cloning before the final answer is applied, though I would argue that you don't gain much by doing so.
>>>>
>>>> So this is the usecase: you send a single offer (say to a conference
>>>> server controller, or to a game server) and you get a separate answer
>>>> from each other endpoint, such as in a mesh conference, or who are in
>>>> the same 'area' of the game, or in hearing (at that moment) in a VR environment.  You might get these answers well after the initial answer or burst of answers.
>>> Yes. And, my idea would be that each of those answers are provided to the browser as PRE_ANSWERS.
>>>
>>> Then, at some point, you decide that you are not going to accept any more answers, and you provide an ANSWER to one of the (C)PCs, which will release all other associated (C)PCs.
>> Sorry, I think you missed the point: the application wants to end up with N connected, active PeerConnections, not one.  This is a
>> mesh conference (for example), not a central-mixer conference.  And for added fun, we don't necessarily know there won't be more
>> coming later, perhaps much later (but that's a possibly-separable feature).
> Speaking in SIP terms, I don't think it is possible to implement mesh conference that way, because clients will normally, once they've received one 200 OK, ACK+BYE any additional ones. Or, a forking proxy will CANCEL all other forked legs once a 200 OK has been forwarded on one of them.

Why would SIP come into this?  Yes, classic PSTN telecom is focused on 
endpoint to one other connection, and any form of more complex 
arrangement is implemented with a mixing server or equivalent, and thus 
SIP flows are designed with that assumption in mind.  However, we are 
NOT constrained to mirror SIP, and we should be able to build 
applications that go beyond what SIP makes easy to do.

I gave some direct use-cases we've talked about since the beginning or 
near: Mesh conferencing, multiplayer games - where parallel forking 
(without collapse to a single session/connection) is desirable/useful.  
If there's an equally-usable way to implement those cases avoiding 
cloning, please detail that.  I think cloning makes initiation of these 
sorts of mesh connections much simpler.

-- 
Randell Jesup
randell-ietf@jesup.org


From harald@alvestrand.no  Mon May  7 01:20:09 2012
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Subject: Re: [rtcweb] Use Case draft - legacy interop
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On 05/05/2012 06:32 PM, IÃ±aki Baz Castillo wrote:
> 2012/5/5 Dan Wing<dwing@cisco.com>:
>>> If I have a SIP phone implementing ICE and DTLS-SRTP (which is
>>> standarized for SIP regardless it has null impementation), will my SIP
>>> phone be able to *directly* talk in the media plane with a WebRTC
>>> browser? or not?
>> That would work.
> Ok, so then, taking into account that SIP defines the usage of
> DTLS-SRTP in RFC 5763 (regardless no one device implements it), why
> does this WG assumes that "interop with non WebRTC endpoints will be
> made via *gateways*"?
>
> Let me please repeat my question: if my SIP phone implements RFC 5763,
> can my SIP phone directly interop at media plane with a WebRTC
> browser? (I know you already replied this, but I want to be very very
> sure) :)
Inaki, your statement that "the WG assumes" is, as far as I can see, a 
straw man made out of red herrings.

The WG's opinion is what's captured in the WG's documents, and (if 
needed) announced by the chairs. Statements by participants are 
statements by participants. If you have issue with the WG's documents, 
quote the text you want changed, don't just make unfounded statements 
like you do above.

I've quoted the overview document before:

    As for all protocol and API specifications, there is no restriction
    that the protocols can only be used to talk to another browser; since
    they are fully specified, any device that implements the protocols
    faithfully should be able to interoperate with the application
    running in the browser.

Since you did not specify the product number of your hypothetical phone, 
and since the IETF has not finished specifying the set of protocols 
needed, there is no guarantee that interworking will happen in practice. 
That is, there's no guarantee that ICE + RFC 5763 is sufficient for 
interoperation.



From christer.holmberg@ericsson.com  Mon May  7 03:58:25 2012
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 7 May 2012 12:58:19 +0200
Thread-Topic: [rtcweb] Proposal for PeerConnection cloning
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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Hi,

>>>>> If you want to support the usecase I suggested, where a single=20
>>>>> offer can result in parallel forking, but where you don't want to=20
>>>>> lock down to a single answerer: then you obviously need to support=20
>>>>> forking, and likely you need/want to support it irrespective of the '=
final' state of the original PeerConnection.  You *can* require cloning bef=
ore the final answer is applied, though I would argue that you don't gain m=
uch by doing so.
>>>>>
>>>>> So this is the usecase: you send a single offer (say to a=20
>>>>> conference server controller, or to a game server) and you get a=20
>>>>> separate answer from each other endpoint, such as in a mesh=20
>>>>> conference, or who are in the same 'area' of the game, or in hearing =
(at that moment) in a VR environment.  You might get these answers well aft=
er the initial answer or burst of answers.
>>>> Yes. And, my idea would be that each of those answers are provided to =
the browser as PRE_ANSWERS.
>>>>
>>>> Then, at some point, you decide that you are not going to accept any m=
ore answers, and you provide an ANSWER to one of the (C)PCs, which will rel=
ease all other associated (C)PCs.
>>> Sorry, I think you missed the point: the application wants to end up=20
>>> with N connected, active PeerConnections, not one.  This is a mesh=20
>>> conference (for example), not a central-mixer conference.  And for adde=
d fun, we don't necessarily know there won't be more coming later, perhaps =
much later (but that's a possibly-separable feature).
>> Speaking in SIP terms, I don't think it is possible to implement mesh co=
nference that way, because clients will normally, once they've received one=
 200 OK, ACK+BYE any
>> additional ones. Or, a forking proxy will CANCEL all other forked legs o=
nce a 200 OK has been forwarded on one of them.
>
> Why would SIP come into this?  Yes, classic PSTN telecom is focused on en=
dpoint to one other connection, and any form of more complex arrangement is=
=20
> implemented with a mixing server or equivalent, and thus SIP flows are de=
signed with that assumption in mind.  However, we are NOT constrained to mi=
rror=20
> SIP, and we should be able to build applications that go beyond what SIP =
makes easy to do.
>
> I gave some direct use-cases we've talked about since the beginning or ne=
ar: Mesh conferencing, multiplayer games - where parallel forking=20
> (without collapse to a single session/connection) is desirable/useful.  I=
f there's an equally-usable way to implement those cases avoiding cloning, =
please
> detail that.  I think cloning makes initiation of these sorts of mesh con=
nections much simpler.

I didn't say you can't do it with CPC, I simply said that you would have is=
sues if doing it with SIP. But, of course you can use whatever protocol on =
the wire.

Maybe there could be an API parameter to indicate, when a final answer is p=
rovided to the browser, whether it should do a CPC "clean-up". It would mak=
e life easier for the JS App developers in non-mesh-conf cases.

Regards,

Christer


From richard.ejzak@alcatel-lucent.com  Mon May  7 05:31:57 2012
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To: Christer Holmberg <christer.holmberg@ericsson.com>, Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 7 May 2012 07:31:46 -0500
Thread-Topic: [rtcweb] Proposal for PeerConnection cloning
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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Hi Christer,
As I mentioned in a previous response to you, you cannot mandate that clone=
s be cleared out once a final ANSWER is applied since that prevents you fro=
m applying independent offer/answer sequences to each clone (as required in=
 some SIP scenarios).  The optional "cleanup" indicator you propose to asso=
ciate with final ANSWER will not work in the case where a final ANSWER has =
already been applied to a clone to allow another offer/answer sequence on t=
hat clone before a final "winner" is selected.  I would prefer to just clea=
n up the "losers" manually or create a new method for this.

There is at least one ERROR that I found in my proposal.  I forgot that the=
 clone will continue to need to be able to report PEER-REFLEXIVE candidates=
 after the cloning operation, since this occurs after ICE exits the ICE_GAT=
HERING state (this should be made clearer in the JSEP text).  So the CPC st=
ill needs the IceCallback.

The SIP example in JSEP is also missing the 2nd offer/answer needed by ICE =
to update the default candidates.  How is the application triggered to gene=
rate this offer?

BR, Richard

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: Monday, May 07, 2012 5:58 AM
To: Randell Jesup; rtcweb@ietf.org
Subject: Re: [rtcweb] Proposal for PeerConnection cloning

Hi,

>>>>> If you want to support the usecase I suggested, where a single=20
>>>>> offer can result in parallel forking, but where you don't want to=20
>>>>> lock down to a single answerer: then you obviously need to support=20
>>>>> forking, and likely you need/want to support it irrespective of the '=
final' state of the original PeerConnection.  You *can* require cloning bef=
ore the final answer is applied, though I would argue that you don't gain m=
uch by doing so.
>>>>>
>>>>> So this is the usecase: you send a single offer (say to a=20
>>>>> conference server controller, or to a game server) and you get a=20
>>>>> separate answer from each other endpoint, such as in a mesh=20
>>>>> conference, or who are in the same 'area' of the game, or in hearing =
(at that moment) in a VR environment.  You might get these answers well aft=
er the initial answer or burst of answers.
>>>> Yes. And, my idea would be that each of those answers are provided to =
the browser as PRE_ANSWERS.
>>>>
>>>> Then, at some point, you decide that you are not going to accept any m=
ore answers, and you provide an ANSWER to one of the (C)PCs, which will rel=
ease all other associated (C)PCs.
>>> Sorry, I think you missed the point: the application wants to end up=20
>>> with N connected, active PeerConnections, not one.  This is a mesh=20
>>> conference (for example), not a central-mixer conference.  And for adde=
d fun, we don't necessarily know there won't be more coming later, perhaps =
much later (but that's a possibly-separable feature).
>> Speaking in SIP terms, I don't think it is possible to implement mesh co=
nference that way, because clients will normally, once they've received one=
 200 OK, ACK+BYE any
>> additional ones. Or, a forking proxy will CANCEL all other forked legs o=
nce a 200 OK has been forwarded on one of them.
>
> Why would SIP come into this?  Yes, classic PSTN telecom is focused on en=
dpoint to one other connection, and any form of more complex arrangement is=
=20
> implemented with a mixing server or equivalent, and thus SIP flows are de=
signed with that assumption in mind.  However, we are NOT constrained to mi=
rror=20
> SIP, and we should be able to build applications that go beyond what SIP =
makes easy to do.
>
> I gave some direct use-cases we've talked about since the beginning or ne=
ar: Mesh conferencing, multiplayer games - where parallel forking=20
> (without collapse to a single session/connection) is desirable/useful.  I=
f there's an equally-usable way to implement those cases avoiding cloning, =
please
> detail that.  I think cloning makes initiation of these sorts of mesh con=
nections much simpler.

I didn't say you can't do it with CPC, I simply said that you would have is=
sues if doing it with SIP. But, of course you can use whatever protocol on =
the wire.

Maybe there could be an API parameter to indicate, when a final answer is p=
rovided to the browser, whether it should do a CPC "clean-up". It would mak=
e life easier for the JS App developers in non-mesh-conf cases.

Regards,

Christer

_______________________________________________
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: Mon, 7 May 2012 13:33:16 +0200
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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2012/5/7 Christer Holmberg <christer.holmberg@ericsson.com>:
> Maybe there could be an API parameter to indicate, when a final answer is=
 provided to the browser, whether it should do a CPC "clean-up". It would m=
ake life easier for the JS App developers in non-mesh-conf cases.

I agree. IMHO it's much better if the API offers a simple mechanism
for building/imitating existing VoIP technologies (i.e. SIP and
XMPP-Jingle). Of course the API could be so powerful that it allows
receiving late responses and use them (something that makes no sense
in SIP world) but giving a simple parameter "just_one_final_response:
true" in the constructor of the first PC, or offering an API function
for discarding all the previous (C)PC's would be nice.

Regards.

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

From randell-ietf@jesup.org  Mon May  7 06:45:58 2012
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On 5/7/2012 8:31 AM, Ejzak, Richard P (Richard) wrote:
> Hi Christer,
> As I mentioned in a previous response to you, you cannot mandate that clones be cleared out once a final ANSWER is applied since that prevents you from applying independent offer/answer sequences to each clone (as required in some SIP scenarios).  The optional "cleanup" indicator you propose to associate with final ANSWER will not work in the case where a final ANSWER has already been applied to a clone to allow another offer/answer sequence on that clone before a final "winner" is selected.  I would prefer to just clean up the "losers" manually or create a new method for this.

It doesn't have to be via this mechanism, but we could have an onfinal 
notification, or a manually polled jsepState checked (on all 
PeerConnections the app has) after installing an answer on one.  (Unless 
the app knows that all it has are clones, in which case it could just 
kill off the others).  jsepState is probably all that is needed (and 
probably needed even with onfinal, unless it gives a list of all clones 
of a finalized PC).

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On Mon, May 7, 2012 at 6:58 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> > Why would SIP come into this?  Yes, classic PSTN telecom is focused on
> endpoint to one other connection, and any form of more complex arrangement
> is
> > implemented with a mixing server or equivalent, and thus SIP flows are
> designed with that assumption in mind.  However, we are NOT constrained to
> mirror
> > SIP, and we should be able to build applications that go beyond what SIP
> makes easy to do.
> >
> > I gave some direct use-cases we've talked about since the beginning or
> near: Mesh conferencing, multiplayer games - where parallel forking
> > (without collapse to a single session/connection) is desirable/useful.
>  If there's an equally-usable way to implement those cases avoiding
> cloning, please
> > detail that.  I think cloning makes initiation of these sorts of mesh
> connections much simpler.
>
> I didn't say you can't do it with CPC, I simply said that you would have
> issues if doing it with SIP. But, of course you can use whatever protocol
> on the wire.
>
> Maybe there could be an API parameter to indicate, when a final answer is
> provided to the browser, whether it should do a CPC "clean-up". It would
> make life easier for the JS App developers in non-mesh-conf cases.
>
>
First of all, even in classic SIP use case, proxy will not hang up or
cancel other call legs when 200 OK is received. You can setup a mesh
conference this way if your end point supports this. Most of the end points
do not, so they hang up the calls, but I have dealt with a few that
actually start multiple calls this way and allow user to switch between
them or conference them together.

Second, you do need to be able to provide a final answer on the primary or
cloned peer connection to implement Partha's scenario, since for this
scenario you need to process another O/A while the original O/A is still in
progress. This means you need to provide a final response on the call leg
where you received the offer, but still be able to receive answers on other
call legs.

_____________
Roman Shpount

--0015175cfd18ee37c304bf749dad
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Mon, May 7, 2012 at 6:58 AM=
, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmbe=
rg@ericsson.com" target=3D"_blank">christer.holmberg@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">&gt;=
 Why would SIP come into this? =A0Yes, classic PSTN telecom is focused on e=
ndpoint to one other connection, and any form of more complex arrangement i=
s<br>

&gt; implemented with a mixing server or equivalent, and thus SIP flows are=
 designed with that assumption in mind. =A0However, we are NOT constrained =
to mirror<br>
&gt; SIP, and we should be able to build applications that go beyond what S=
IP makes easy to do.<br>
&gt;<br>
&gt; I gave some direct use-cases we&#39;ve talked about since the beginnin=
g or near: Mesh conferencing, multiplayer games - where parallel forking<br=
>
&gt; (without collapse to a single session/connection) is desirable/useful.=
 =A0If there&#39;s an equally-usable way to implement those cases avoiding =
cloning, please<br>
&gt; detail that. =A0I think cloning makes initiation of these sorts of mes=
h connections much simpler.<br>
<br>
</div></div>I didn&#39;t say you can&#39;t do it with CPC, I simply said th=
at you would have issues if doing it with SIP. But, of course you can use w=
hatever protocol on the wire.<br>
<br>
Maybe there could be an API parameter to indicate, when a final answer is p=
rovided to the browser, whether it should do a CPC &quot;clean-up&quot;. It=
 would make life easier for the JS App developers in non-mesh-conf cases.<b=
r>

<br></blockquote><div>=A0</div></div>First of all, even in classic SIP use =
case, proxy will not hang up or cancel other call legs when 200 OK is recei=
ved. You can setup a mesh conference this way if your end point supports th=
is. Most of the end points do not, so they hang up the calls, but I have de=
alt with a few that actually start multiple calls this way and allow user t=
o switch between them or conference them together.<br>
<br>Second, you do need to be able to provide a final answer on the primary=
 or cloned peer connection to implement Partha&#39;s scenario, since for th=
is scenario you need to process another O/A while the original O/A is still=
 in progress. This means you need to provide a final response on the call l=
eg where you received the offer, but still be able to receive answers on ot=
her call legs.<br>
<br>_____________<br>Roman Shpount<br>
<br>

--0015175cfd18ee37c304bf749dad--

From harald@alvestrand.no  Mon May  7 14:14:37 2012
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Subject: [rtcweb] Consent freshness and message-integrity (Re: Use Case draft - legacy interop)
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Forking the thread, since this is a different detail.....

On 05/05/2012 07:24 PM, Dan Wing wrote:
> The other nuance is that, because
> doing the SHA1 for MESSAGE-INTEGRITY isn't needed for consent freshness,
> there is desire to allow those periodic ICE connectivity checks to
> omit the MESSAGE-INTEGRITY, which is a change to ICE.  See
> draft-muthu-behave-consent-freshness.
Omitting MESSAGE-INTEGRITY would allow off-path attackers to inject fake 
connectivity checks, and thus to simulate continued consent.

If connectivity checks for content freshness are worth doing, they're 
worth protecting.
My $0.02.

                    Harald



From dwing@cisco.com  Mon May  7 14:23:42 2012
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From: "Dan Wing" <dwing@cisco.com>
To: "'Harald Alvestrand'" <harald@alvestrand.no>
References: <CA+9kkMCYArLPRP3c00UdOja64WRT6ghN0PSy7XvM_wbxBBB+vA@mail.gmail.com> <2E496AC9-63A0-464A-A628-7407ED8DD9C4@phonefromhere.com> <387F9047F55E8C42850AD6B3A7A03C6C0E23B16B@inba-mail01.sonusnet.com> <E2714FBC-D06B-4A12-9E07-C49EBF55084C@phonefromhere.com> <4F9EC0B2.10903@alcatel-lucent.com> <101C6067BEC68246B0C3F6843BCCC1E31299282765@MCHP058A.global-ad.net> <CAJNg7VKENERKAFA-n5KeoeBNmGgHrnzDOU0BzC9+fSdsuGwdEw@mail.gmail.com> <E17CAD772E76C742B645BD4DC602CD810616F24F@NAHALD.us.int.genesyslab.com> <4FA0F43E.4020308@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F336@NAHALD.us.int.genesyslab.com> <4FA1575C.4050508@ericsson.com> <E17CAD772E76C742B645BD4DC602CD810616F4BF@NAHALD.us.int.genesyslab.com> <4FA37A1E.4080806@alvestrand.no> <CALiegf=H5QH_YY-cJ4z29wChWZ-VoQpHvsZCeaJPjTgVp+km3Q@mai l.gmail.com> <0db701cd	2adb$98082f10$c8188d30$@com> <CALiegf=C7a6zeUDHn-Wuku9eFWmADG5N+D8oXSQbwJKSYYjcQg@mail.gmail.com> <0dc901cd2ae3$f9ffb500$edff1f00$@com> <4FA83B3A.9070600@alvestr and.no>
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Date: Mon, 7 May 2012 14:23:48 -0700
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Consent freshness and message-integrity (Re: Use Case draft - legacy interop)
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> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]
> Sent: Monday, May 07, 2012 2:15 PM
> To: Dan Wing
> Cc: rtcweb@ietf.org
> Subject: Consent freshness and message-integrity (Re: [rtcweb] Use Case
> draft - legacy interop)
> 
> Forking the thread, since this is a different detail.....
> 
> On 05/05/2012 07:24 PM, Dan Wing wrote:
> > The other nuance is that, because
> > doing the SHA1 for MESSAGE-INTEGRITY isn't needed for consent
> freshness,
> > there is desire to allow those periodic ICE connectivity checks to
> > omit the MESSAGE-INTEGRITY, which is a change to ICE.  See
> > draft-muthu-behave-consent-freshness.
> Omitting MESSAGE-INTEGRITY would allow off-path attackers to inject
> fake
> connectivity checks, and thus to simulate continued consent.

To successfully do that, those off-path attackers would have to guess 
the random 96-bit STUN transaction-id.

-d

> If connectivity checks for content freshness are worth doing, they're
> worth protecting.
> My $0.02.
> 
>                     Harald



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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On 05/07/2012 03:44 PM, Randell Jesup wrote:
> On 5/7/2012 8:31 AM, Ejzak, Richard P (Richard) wrote:
>> Hi Christer,
>> As I mentioned in a previous response to you, you cannot mandate that 
>> clones be cleared out once a final ANSWER is applied since that 
>> prevents you from applying independent offer/answer sequences to each 
>> clone (as required in some SIP scenarios).  The optional "cleanup" 
>> indicator you propose to associate with final ANSWER will not work in 
>> the case where a final ANSWER has already been applied to a clone to 
>> allow another offer/answer sequence on that clone before a final 
>> "winner" is selected.  I would prefer to just clean up the "losers" 
>> manually or create a new method for this.
>
> It doesn't have to be via this mechanism, but we could have an onfinal 
> notification, or a manually polled jsepState checked (on all 
> PeerConnections the app has) after installing an answer on one.  
> (Unless the app knows that all it has are clones, in which case it 
> could just kill off the others).  jsepState is probably all that is 
> needed (and probably needed even with onfinal, unless it gives a list 
> of all clones of a finalized PC).
>
It seems to me that

for each Pc in ClonedConnections {
   if (Pc !== winner)
      Pc.close()
}

is simple enough to code that we should simply say that "the JS has to 
do cleanup when he decides he only wants one of the PeerConnection".

Creating mechanisms for things to happen behind the scenes seems to me 
to make life for the programmer more complex, not less; "why did the 
voice of person 2 go silent? Oh - you applied an ANSWER instead of a 
PR-ANSWER to person 1"....



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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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Richard, thank you for doing this! It's definitely the most 
thought-through proposal I've seen on this question!

Two questions that are different from what others have raised:

On 05/06/2012 08:54 PM, Ejzak, Richard P (Richard) wrote:
>
> Harald asked for a proposal for PeerConnection cloning, so here it goes:
>
> I propose to create a new constructor ?ClonePeerConnection? (CPC 
> below) with the following semantics.
>
>    1. CPC will create a new PeerConnection object when successful. 
>       The resulting PC behaves just like any other PC except as
>       described below.
>    2. CPC will take as input a reference to an existing PeerConnection
>       (PC) object (no configuration or IceCallback parameters).
>    3. CPC can clone any other PC (including a cloned PC).  The
>       sequence of cloning is not important and parents have no
>       particular status different from any generation of clone.
>    4. CPC will fail if either the local description of the parent PC
>       is not set or the parent PC ICE is in ICE_CLOSED or
>       ICE_GATHERING state.  The parent PC can be in the OPENING or
>       ACTIVE state.
>    5. The CPC object will inherit the local streams, local ICE
>       candidates, and local description of the PC.
>
when it inherits the local streams, do you assume that unless the JS 
applies some setting, they will also inherit the "enabled/disabled" 
state, but that the "enabled/disabled" state can be changed 
independently for each PeerConnection's local streams after the cloning?
>
>   1.
>
>
>    2. The remote streams and remote description for the CPC object
>       will be set to empty.
>    3. The CPC object will be set to the OPENING state to reflect that
>       only the local description is set.
>    4. The ICE states other than CLOSED and GATHERING will be handled
>       independently for each PC and its clones (as is true in standard
>       forking scenarios).
>    5. The ICE state for this clone will be set to ICE_WAITING to
>       reflect that all candidates are available but the remote
>       configuration is not yet set.
>    6. The PC and its clone(s) use a common pool of media resources.
>    7. If the parent PC object has already released unused resources
>       (final ANSWER), resources are reallocated as available to
>       reflect the capabilities for each stream (as they would be
>       reflected in a createOffer).
>    8. The local streams might multicast toward the remote targets
>       depending on the directionality attributes independently set for
>       each PC and clone.
>
So far, we've stayed away from anything that involves IP multicast. Are 
you sure it's worth the potential complexity to try to use this here?
Remember that using multicast changes a *lot* of things on both ends of 
the connections - the sender has to have IP multicast address 
allocations, he has to decide between SSM and ASM, the recipients have 
to use a multicast joining protocol, the infrastructure has to be 
present to carry the multicast groups, we have to have a fallback in 
case it isn't, the appropriate congestion control protocols are probably 
different ..... I would prefer to not bring this in scope for this round 
of WebRTC.
Is there a compelling reason you see for embarking on this particular 
adventure?
>
>   1.
>
>
>    2. The application should manage the directionality attributes for
>       remote streams from different targets to avoid resource conflicts.
>
> CPC will be used primarily if forking is anticipated or actually 
> occurs.  It can also be used to clone a stable PC if desired for other 
> reasons.  When used for SIP forking, creation of the clone can be 
> delayed until an ANSWER actually arrives from a 2^nd target as long as 
> final ANSWER hasn?t been applied to the parent (PRANSWER is OK).  The 
> parent always handles the first target; the first clone handles the 
> 2^nd target; etc.  The application can even try to clone for forking 
> after the first ANSWER is applied to the parent and resources are 
> released, as long as the local description has not changed, at the 
> risk that some resources needed for the 2^nd target are no longer 
> available and must be renegotiated.
>
> Comments on this proposal are welcome.
>
> Richard
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    Richard, thank you for doing this! It's definitely the most
    thought-through proposal I've seen on this question!<br>
    <br>
    Two questions that are different from what others have raised:<br>
    <br>
    On 05/06/2012 08:54 PM, 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: 10pt; font-family: Arial;">Harald asked
              for a proposal for PeerConnection cloning, so
              here it goes:<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;">I propose to
              create a new constructor &#8220;ClonePeerConnection&#8221;
              (CPC below) with the following semantics.<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
        <ol style="margin-top: 0in;" start="1" type="1">
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">CPC will
                create a new PeerConnection object when successful.  The
                resulting PC behaves just like any other PC except as
                described below.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">CPC will
                take as input a reference to an existing PeerConnection
                (PC) object (no configuration or IceCallback
                parameters).<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">CPC can
                clone any other PC (including a cloned PC).  The
                sequence of cloning is not important and parents have no
                particular status different from any generation of
                clone.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">CPC will
                fail if either the local description of the parent PC is
                not set or the parent PC ICE is in ICE_CLOSED or
                ICE_GATHERING state.  The parent PC can be in the
                OPENING or ACTIVE state.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The CPC
                object will inherit the local streams, local ICE
                candidates, and local description of the PC.</span></font></li>
        </ol>
      </div>
    </blockquote>
    when it inherits the local streams, do you assume that unless the JS
    applies some setting, they will also inherit the "enabled/disabled"
    state, but that the "enabled/disabled" state can be changed
    independently for each PeerConnection's local streams after the
    cloning?<br>
    <blockquote
cite="mid:6F428EFD2B8C2F49A2FB1317291A76C11364EC028C@USNAVSXCHMBSA1.ndc.alcatel-lucent.com"
      type="cite">
      <div class="Section1">
        <ol style="margin-top: 0in;" start="1" type="1">
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;"><o:p></o:p></span></font><br>
          </li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The remote
                streams and remote description for the CPC object will
                be set to empty.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The CPC
                object will be set to the OPENING state to reflect that
                only the local description is set.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The ICE
                states other than CLOSED and GATHERING will be handled
                independently for each PC and its clones (as is true in
                standard forking scenarios).<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The ICE
                state for this clone will be set to ICE_WAITING to
                reflect that all candidates are available but the remote
                configuration is not yet set.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The PC and
                its clone(s) use a common pool of media resources.<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">If the
                parent PC object has already released unused resources
                (final ANSWER), resources are reallocated as available
                to reflect the capabilities for each stream (as they
                would be reflected in a createOffer).<o:p></o:p></span></font></li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The local
                streams might multicast toward the remote targets
                depending on the directionality attributes independently
                set for each PC and clone.</span></font></li>
        </ol>
      </div>
    </blockquote>
    So far, we've stayed away from anything that involves IP multicast.
    Are you sure it's worth the potential complexity to try to use this
    here?<br>
    Remember that using multicast changes a *lot* of things on both ends
    of the connections - the sender has to have IP multicast address
    allocations, he has to decide between SSM and ASM, the recipients
    have to use a multicast joining protocol, the infrastructure has to
    be present to carry the multicast groups, we have to have a fallback
    in case it isn't, the appropriate congestion control protocols are
    probably different ..... I would prefer to not bring this in scope
    for this round of WebRTC.<br>
    Is there a compelling reason you see for embarking on this
    particular adventure?<br>
    <blockquote
cite="mid:6F428EFD2B8C2F49A2FB1317291A76C11364EC028C@USNAVSXCHMBSA1.ndc.alcatel-lucent.com"
      type="cite">
      <div class="Section1">
        <ol style="margin-top: 0in;" start="1" type="1">
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;"><o:p></o:p></span></font><br>
          </li>
          <li class="MsoNormal" style=""><font face="Arial" size="2"><span
                style="font-size: 10pt; font-family: Arial;">The
                application should manage the directionality attributes
                for remote streams from different targets to avoid
                resource conflicts.<o:p></o:p></span></font></li>
        </ol>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;">CPC will be
              used primarily if forking is anticipated or
              actually occurs.  It can also be used to clone a stable PC
              if desired for
              other reasons.  When used for SIP forking, creation of the
              clone can be
              delayed until an ANSWER actually arrives from a 2<sup>nd</sup>
              target as long
              as final ANSWER hasn&#8217;t been applied to the parent
              (PRANSWER is OK).  The
              parent always handles the first target; the first clone
              handles the 2<sup>nd</sup>
              target; etc.  The application can even try to clone for
              forking after the
              first ANSWER is applied to the parent and resources are
              released, as long as
              the local description has not changed, at the risk that
              some resources needed
              for the 2<sup>nd</sup> target are no longer available and
              must be renegotiated.<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;">Comments on
              this proposal are welcome.<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;">Richard<o:p></o:p></span></font></p>
        <p class="MsoNormal"><font face="Arial" size="2"><span
              style="font-size: 10pt; font-family: Arial;"><o:p> </o:p></span></font></p>
      </div>
      <pre wrap="">
<fieldset class="mimeAttachmentHeader"></fieldset>
_______________________________________________
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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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Consent freshness and message-integrity (Re: Use Case draft - legacy interop)
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On 05/07/2012 11:23 PM, Dan Wing wrote:
>> -----Original Message-----
>> From: Harald Alvestrand [mailto:harald@alvestrand.no]
>> Sent: Monday, May 07, 2012 2:15 PM
>> To: Dan Wing
>> Cc: rtcweb@ietf.org
>> Subject: Consent freshness and message-integrity (Re: [rtcweb] Use Case
>> draft - legacy interop)
>>
>> Forking the thread, since this is a different detail.....
>>
>> On 05/05/2012 07:24 PM, Dan Wing wrote:
>>> The other nuance is that, because
>>> doing the SHA1 for MESSAGE-INTEGRITY isn't needed for consent
>> freshness,
>>> there is desire to allow those periodic ICE connectivity checks to
>>> omit the MESSAGE-INTEGRITY, which is a change to ICE.  See
>>> draft-muthu-behave-consent-freshness.
>> Omitting MESSAGE-INTEGRITY would allow off-path attackers to inject
>> fake
>> connectivity checks, and thus to simulate continued consent.
> To successfully do that, those off-path attackers would have to guess
> the random 96-bit STUN transaction-id.
You're right, successfully replying to consent freshness packets 
requires read access to the consent freshness packets being sent (I was 
thinking that consent freshness was being generated from the recipient, 
not the sender, which is opposite to what draft-muthu does .... or 
rather, it's not clear to me what direction a successful 
request/response exchange verifies the freshness of). So it's an on-path 
attacker. I'll go take that argument to behave; I don't think that 
tradeoff is in the right place.



From randell-ietf@jesup.org  Mon May  7 14:54:22 2012
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On 5/7/2012 5:32 PM, Harald Alvestrand wrote:
> Richard, thank you for doing this! It's definitely the most
> thought-through proposal I've seen on this question!

Ditto!

>>  5. The CPC object will inherit the local streams, local ICE
>>     candidates, and local description of the PC.
>>
> when it inherits the local streams, do you assume that unless the JS
> applies some setting, they will also inherit the "enabled/disabled"
> state, but that the "enabled/disabled" state can be changed
> independently for each PeerConnection's local streams after the cloning?

I would assume that personally, and that typically after cloning you'd 
do "appropriate" tweaking to new (and old).

>>  8. The local streams might multicast toward the remote targets
>>     depending on the directionality attributes independently set for
>>     each PC and clone.
>>
> So far, we've stayed away from anything that involves IP multicast. Are
> you sure it's worth the potential complexity to try to use this here?

Richard - do you mean "IP Multicast" (tm) or 'multicast' by sending the 
same encoded data to more than one destination in separate RTP streams? 
  (i.e. one encoding being used to simulcast to multiple receivers?)

Simulcast is possible and useful (almost required for Mesh 
Conferencing); IP Multicast has been pretty much ruled out-of-scope IIRC.

-- 
Randell Jesup
randell-ietf@jesup.org

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Thanks, Harald, Randell.

Question 1: I have no strong opinion on whether each local stream should in=
herit its parent's state or get reset to a default value, but inheriting th=
e state seems cleaner (today).  It must be possible to control the state of=
 each cloned local stream separately.

Question 2: I should have been more careful in my choice of words.  Randell=
 interpreted my meaning correctly.  My meaning is better indicated with the=
 word "simulcast", as Randell suggests.  I NEVER intended the use of "IP mu=
lticast addressing".

Richard


________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Monday, May 07, 2012 4:32 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Proposal for PeerConnection cloning

Richard, thank you for doing this! It's definitely the most thought-through=
 proposal I've seen on this question!

Two questions that are different from what others have raised:

On 05/06/2012 08:54 PM, Ejzak, Richard P (Richard) wrote:
Harald asked for a proposal for PeerConnection cloning, so here it goes:

I propose to create a new constructor "ClonePeerConnection" (CPC below) wit=
h the following semantics.


 1.  CPC will create a new PeerConnection object when successful.  The resu=
lting PC behaves just like any other PC except as described below.
 2.  CPC will take as input a reference to an existing PeerConnection (PC) =
object (no configuration or IceCallback parameters).
 3.  CPC can clone any other PC (including a cloned PC).  The sequence of c=
loning is not important and parents have no particular status different fro=
m any generation of clone.
 4.  CPC will fail if either the local description of the parent PC is not =
set or the parent PC ICE is in ICE_CLOSED or ICE_GATHERING state.  The pare=
nt PC can be in the OPENING or ACTIVE state.
 5.  The CPC object will inherit the local streams, local ICE candidates, a=
nd local description of the PC.
when it inherits the local streams, do you assume that unless the JS applie=
s some setting, they will also inherit the "enabled/disabled" state, but th=
at the "enabled/disabled" state can be changed independently for each PeerC=
onnection's local streams after the cloning?


 1.
 2.  The remote streams and remote description for the CPC object will be s=
et to empty.
 3.  The CPC object will be set to the OPENING state to reflect that only t=
he local description is set.
 4.  The ICE states other than CLOSED and GATHERING will be handled indepen=
dently for each PC and its clones (as is true in standard forking scenarios=
).
 5.  The ICE state for this clone will be set to ICE_WAITING to reflect tha=
t all candidates are available but the remote configuration is not yet set.
 6.  The PC and its clone(s) use a common pool of media resources.
 7.  If the parent PC object has already released unused resources (final A=
NSWER), resources are reallocated as available to reflect the capabilities =
for each stream (as they would be reflected in a createOffer).
 8.  The local streams might multicast toward the remote targets depending =
on the directionality attributes independently set for each PC and clone.
So far, we've stayed away from anything that involves IP multicast. Are you=
 sure it's worth the potential complexity to try to use this here?
Remember that using multicast changes a *lot* of things on both ends of the=
 connections - the sender has to have IP multicast address allocations, he =
has to decide between SSM and ASM, the recipients have to use a multicast j=
oining protocol, the infrastructure has to be present to carry the multicas=
t groups, we have to have a fallback in case it isn't, the appropriate cong=
estion control protocols are probably different ..... I would prefer to not=
 bring this in scope for this round of WebRTC.
Is there a compelling reason you see for embarking on this particular adven=
ture?


 1.
 2.  The application should manage the directionality attributes for remote=
 streams from different targets to avoid resource conflicts.

CPC will be used primarily if forking is anticipated or actually occurs.  I=
t can also be used to clone a stable PC if desired for other reasons.  When=
 used for SIP forking, creation of the clone can be delayed until an ANSWER=
 actually arrives from a 2nd target as long as final ANSWER hasn't been app=
lied to the parent (PRANSWER is OK).  The parent always handles the first t=
arget; the first clone handles the 2nd target; etc.  The application can ev=
en try to clone for forking after the first ANSWER is applied to the parent=
 and resources are released, as long as the local description has not chang=
ed, at the risk that some resources needed for the 2nd target are no longer=
 available and must be renegotiated.

Comments on this proposal are welcome.

Richard






_______________________________________________

rtcweb mailing list

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

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


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Thanks, Harald, Randell.<o:p></o:p></s=
pan></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Question 1: I have no strong opinion o=
n
whether each local stream should inherit its parent&#8217;s state or get re=
set
to a default value, but inheriting the state seems cleaner (today). &nbsp;I=
t
must be possible to control the state of each cloned local stream separatel=
y.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Question 2: I should have been more
careful in my choice of words. &nbsp;Randell interpreted my meaning
correctly.&nbsp; My meaning is better indicated with the word &#8220;simulc=
ast&#8221;,
as Randell suggests.&nbsp; I NEVER intended the use of &#8220;IP multicast
addressing&#8221;.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
color=3Dblack face=3D"Times New Roman"><span style=3D'font-size:12.0pt;font=
-family:
"Times New Roman";color:windowtext'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 color=3Dblack face=3DTahoma><span
style=3D'font-size:10.0pt;font-family:Tahoma;color:windowtext;font-weight:b=
old'>From:</span></font></b><font
size=3D2 color=3Dblack face=3DTahoma><span style=3D'font-size:10.0pt;font-f=
amily:Tahoma;
color:windowtext'> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]=
 <b><span
style=3D'font-weight:bold'>On Behalf Of </span></b>Harald Alvestrand<br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, May 07, 2012 4=
:32 PM<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> Re: [rtcweb] Propos=
al for
PeerConnection cloning</span></font><font size=3D3 color=3Dblack
face=3D"Times New Roman"><span style=3D'font-size:12.0pt;font-family:"Times=
 New Roman";
color:windowtext'><o:p></o:p></span></font></p>

</div>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"FuturaA Bk BT"><s=
pan
style=3D'font-size:11.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3D"FuturaA Bk BT"><s=
pan
style=3D'font-size:11.0pt'>Richard, thank you for doing this! It's definite=
ly the
most thought-through proposal I've seen on this question!<br>
<br>
Two questions that are different from what others have raised:<br>
<br>
On 05/06/2012 08:54 PM, Ejzak, Richard P (Richard) wrote: </span></font><fo=
nt
face=3D"Times New Roman"><span style=3D'font-family:"Times New Roman"'><o:p=
></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><!--[if gte mso 9]><xml>
 <u1:shapedefaults u2:ext=3D"edit" spidmax=3D"1026"/>
</xml><![endif]--><!--[if gte mso 9]><xml>
 <u3:shapelayout u4:ext=3D"edit">
  <u3:idmap u4:ext=3D"edit" data=3D"1"/>
 </u3:shapelayout>
</xml><![endif]-->Harald asked for a proposal for PeerConnection cloning, s=
o
here it goes:<u5:p></u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'>I propose to create a new constructor
&#8220;ClonePeerConnection&#8221; (CPC below) with the following semantics.=
<u5:p></u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<ol style=3D'margin-top:0in' start=3D1 type=3D1>
 <li class=3DMsoNormal style=3D'mso-list:l2 level1 lfo1'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>CPC wi=
ll
     create a new PeerConnection object when successful.&nbsp; The resultin=
g PC
     behaves just like any other PC except as described below.<u5:p></u5:p>=
</span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l2 level1 lfo1'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>CPC wi=
ll take
     as input a reference to an existing PeerConnection (PC) object (no
     configuration or IceCallback parameters).<u5:p></u5:p></span></font><o=
:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l2 level1 lfo1'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>CPC ca=
n clone
     any other PC (including a cloned PC).&nbsp; The sequence of cloning is=
 not
     important and parents have no particular status different from any
     generation of clone.<u5:p></u5:p></span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l2 level1 lfo1'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>CPC wi=
ll fail
     if either the local description of the parent PC is not set or the par=
ent
     PC ICE is in ICE_CLOSED or ICE_GATHERING state.&nbsp; The parent PC ca=
n be
     in the OPENING or ACTIVE state.<u5:p></u5:p></span></font><o:p></o:p><=
/li>
 <li class=3DMsoNormal style=3D'mso-list:l2 level1 lfo1'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The CP=
C object
     will inherit the local streams, local ICE candidates, and local
     description of the PC.</span></font><o:p></o:p></li>
</ol>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt;font-family:"Times New Roman"'>when it inherits t=
he
local streams, do you assume that unless the JS applies some setting, they =
will
also inherit the &quot;enabled/disabled&quot; state, but that the
&quot;enabled/disabled&quot; state can be changed independently for each
PeerConnection's local streams after the cloning?<br>
<br>
<o:p></o:p></span></font></p>

<ol style=3D'margin-top:0in' start=3D1 type=3D1>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3D"FuturaA Bk BT"><span style=3D'font-size:11.0pt'><u5:p></u5:p><=
o:p>&nbsp;</o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The re=
mote
     streams and remote description for the CPC object will be set to empty=
.<u5:p></u5:p></span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The CP=
C object
     will be set to the OPENING state to reflect that only the local
     description is set.<u5:p></u5:p></span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The IC=
E states
     other than CLOSED and GATHERING will be handled independently for each=
 PC
     and its clones (as is true in standard forking scenarios).<u5:p></u5:p=
></span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The IC=
E state
     for this clone will be set to ICE_WAITING to reflect that all candidat=
es
     are available but the remote configuration is not yet set.<u5:p></u5:p=
></span></font><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The PC=
 and its
     clone(s) use a common pool of media resources.<u5:p></u5:p></span></fo=
nt><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>If the=
 parent
     PC object has already released unused resources (final ANSWER), resour=
ces
     are reallocated as available to reflect the capabilities for each stre=
am
     (as they would be reflected in a createOffer).<u5:p></u5:p></span></fo=
nt><o:p></o:p></li>
 <li class=3DMsoNormal style=3D'mso-list:l1 level1 lfo2'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The lo=
cal
     streams might multicast toward the remote targets depending on the
     directionality attributes independently set for each PC and clone.</sp=
an></font><o:p></o:p></li>
</ol>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt;font-family:"Times New Roman"'>So far, we've stay=
ed
away from anything that involves IP multicast. Are you sure it's worth the
potential complexity to try to use this here?<br>
Remember that using multicast changes a *lot* of things on both ends of the
connections - the sender has to have IP multicast address allocations, he h=
as
to decide between SSM and ASM, the recipients have to use a multicast joini=
ng
protocol, the infrastructure has to be present to carry the multicast group=
s,
we have to have a fallback in case it isn't, the appropriate congestion con=
trol
protocols are probably different ..... I would prefer to not bring this in
scope for this round of WebRTC.<br>
Is there a compelling reason you see for embarking on this particular
adventure?<br>
<br>
<o:p></o:p></span></font></p>

<ol style=3D'margin-top:0in' start=3D1 type=3D1>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo3'><font size=3D2 col=
or=3Dblack
     face=3D"FuturaA Bk BT"><span style=3D'font-size:11.0pt'><u5:p></u5:p><=
o:p>&nbsp;</o:p></span></font></li>
 <li class=3DMsoNormal style=3D'mso-list:l0 level1 lfo3'><font size=3D2 col=
or=3Dblack
     face=3DArial><span style=3D'font-size:10.0pt;font-family:Arial'>The ap=
plication
     should manage the directionality attributes for remote streams from
     different targets to avoid resource conflicts.<u5:p></u5:p></span></fo=
nt><o:p></o:p></li>
</ol>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'>CPC will be used primarily if forking is anticipa=
ted
or actually occurs. &nbsp;It can also be used to clone a stable PC if desir=
ed for
other reasons. &nbsp;When used for SIP forking, creation of the clone can b=
e
delayed until an ANSWER actually arrives from a 2<sup>nd</sup> target as lo=
ng
as final ANSWER hasn&#8217;t been applied to the parent (PRANSWER is OK).
&nbsp;The parent always handles the first target; the first clone handles t=
he 2<sup>nd</sup>
target; etc. &nbsp;The application can even try to clone for forking after =
the
first ANSWER is applied to the parent and resources are released, as long a=
s
the local description has not changed, at the risk that some resources need=
ed
for the 2<sup>nd</sup> target are no longer available and must be renegotia=
ted.<u5:p></u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'>Comments on this proposal are welcome.<u5:p></u5:=
p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'>Richard<u5:p></u5:p></span></font><o:p></o:p></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial'><u5:p>&nbsp;</u5:p></span></font><o:p></o:p></p>

<pre wrap=3D""><font size=3D2 color=3Dblack face=3D"Courier New"><span
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre><pre><field=
set class=3D"mimeAttachmentHeader"></fieldset><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'><o:p>&nbsp;</o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>_______________________________________________<o:p></o:p></span></font><=
/pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'>rtcweb mailing list<o:p></o:p></span></font></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'><a
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><o:p></o:p></span></font=
></pre><pre><font
size=3D2 color=3Dblack face=3D"Courier New"><span style=3D'font-size:10.0pt=
'><a
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/=
mailman/listinfo/rtcweb</a><o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D3 color=3Dblack face=3D"Times New Roman">=
<span
style=3D'font-size:12.0pt;font-family:"Times New Roman"'><o:p>&nbsp;</o:p><=
/span></font></p>

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From: Roman Shpount <roman@telurix.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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One question I have is when should TURN allocations be released? TURN would
not be used for 99% of PeerConnections but it cannot be released until it
is guaranteed that the offer that used this TURN candidate would not be
used to create any more PeerConnections. We would need a separate method to
release the TURN candidates, which can be complicated by the fact that you
can have several offers for a given peerconnection in progress, each with
its own TURN candidates that can have overlapping lifetimes.

I would actually think that the model where an offer object is created and
each PeerConnection is created  using the offer object and the answer would
be cleaner.

One way to implement this, is to define a factory object is used to create
offers. SDP is extracted from an offer object. Once answer is received, a
method in the offer object is used to create the PeerConnection. Once no
more answers are expected offer object should be released.

The other option, that would be less disruptive to the curretn API,  is
that instead of cloning peerconnection method, a method that returns
reference to the current offer is used. A cloned peerconnection can be
created using this offer. Once final answer is processed by the
peerconnection, offer object is released by it. If multiple answers are
expected, offer object can be retrieved using the get offer method and
stored separately until the moment when answers for this particular offer a
no longer expected.
_____________
Roman Shpount

--000e0ce0d862ed747e04bf7c6ae0
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One question I have is when should TURN allocations be released? TURN would=
 not be used for 99% of PeerConnections but it cannot be released until it =
is guaranteed that the offer that used this TURN candidate would not be use=
d to create any more PeerConnections. We would need a separate method to re=
lease the TURN candidates, which can be complicated by the fact that you ca=
n have several offers for a given peerconnection in progress, each with its=
 own TURN candidates that can have overlapping lifetimes.<br>
<br>I would actually think that the model where an offer object is created =
and each PeerConnection is created=A0 using the offer object and the answer=
 would be cleaner. <br><br>One way to implement this, is to define a factor=
y object is used to create offers. SDP is extracted from an offer object. O=
nce answer is received, a method in the offer object is used to create the =
PeerConnection. Once no more answers are expected offer object should be re=
leased.<br>
<br>The other option, that would be less disruptive to the curretn API,=A0 =
is that instead of cloning peerconnection method, a method that returns ref=
erence to the current offer is used. A cloned peerconnection can be created=
 using this offer. Once final answer is processed by the peerconnection, of=
fer object is released by it. If multiple answers are expected, offer objec=
t can be retrieved using the get offer method and stored separately until t=
he moment when answers for this particular offer a no longer expected.<br c=
lear=3D"all">
_____________<br>Roman Shpount<br>
<br>

--000e0ce0d862ed747e04bf7c6ae0--

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 7 May 2012 21:10:56 -0500
Thread-Topic: [rtcweb] Proposal for PeerConnection cloning
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You raise a good point since some candidates may be released before cloning=
 occurs.  So this is another bug in the procedure.  I would suggest that th=
e configuration also be passed to the CPC and that if any of the required c=
andidates are released before cloning occurs that they be reallocated.

Alternately, we could recommend that candidates be kept allocated longer th=
an usual and that if cloning is attempted after any of the candidates are r=
eleased then the cloning procedure fails.

The latter approach should be sufficient for SIP forking scenarios, but oth=
ers may prefer the generality of the former.  Or maybe both?  Any votes?

Richard

________________________________
From: Roman Shpount [mailto:roman@telurix.com]
Sent: Monday, May 07, 2012 8:37 PM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Proposal for PeerConnection cloning

One question I have is when should TURN allocations be released? TURN would=
 not be used for 99% of PeerConnections but it cannot be released until it =
is guaranteed that the offer that used this TURN candidate would not be use=
d to create any more PeerConnections. We would need a separate method to re=
lease the TURN candidates, which can be complicated by the fact that you ca=
n have several offers for a given peerconnection in progress, each with its=
 own TURN candidates that can have overlapping lifetimes.

I would actually think that the model where an offer object is created and =
each PeerConnection is created  using the offer object and the answer would=
 be cleaner.

One way to implement this, is to define a factory object is used to create =
offers. SDP is extracted from an offer object. Once answer is received, a m=
ethod in the offer object is used to create the PeerConnection. Once no mor=
e answers are expected offer object should be released.

The other option, that would be less disruptive to the curretn API,  is tha=
t instead of cloning peerconnection method, a method that returns reference=
 to the current offer is used. A cloned peerconnection can be created using=
 this offer. Once final answer is processed by the peerconnection, offer ob=
ject is released by it. If multiple answers are expected, offer object can =
be retrieved using the get offer method and stored separately until the mom=
ent when answers for this particular offer a no longer expected.
_____________
Roman Shpount

--_000_6F428EFD2B8C2F49A2FB1317291A76C11364EC0881USNAVSXCHMBSA_
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<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>You raise a good point since some
candidates may be released before cloning occurs. &nbsp;So this is another =
bug
in the procedure.&nbsp; I would suggest that the configuration also be pass=
ed
to the CPC and that if any of the required candidates are released before
cloning occurs that they be reallocated.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Alternately, we could recommend that
candidates be kept allocated longer than usual and that if cloning is attem=
pted
after any of the candidates are released then the cloning procedure fails.<=
o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>The latter approach should be sufficie=
nt
for SIP forking scenarios, but others may prefer the generality of the form=
er.&nbsp;
Or maybe both?&nbsp; Any votes?<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'>Richard<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dnavy face=3DArial><span style=
=3D'font-size:
10.0pt;font-family:Arial;color:navy'><o:p>&nbsp;</o:p></span></font></p>

<div>

<div class=3DMsoNormal align=3Dcenter style=3D'text-align:center'><font siz=
e=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>

<hr size=3D2 width=3D"100%" align=3Dcenter tabindex=3D-1>

</span></font></div>

<p class=3DMsoNormal><b><font size=3D2 face=3DTahoma><span style=3D'font-si=
ze:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=3D2
face=3DTahoma><span style=3D'font-size:10.0pt;font-family:Tahoma'> Roman Sh=
pount
[mailto:roman@telurix.com] <br>
<b><span style=3D'font-weight:bold'>Sent:</span></b> Monday, May 07, 2012 8=
:37 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] Propos=
al for
PeerConnection cloning</span></font><o:p></o:p></p>

</div>

<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span style=3D=
'font-size:
12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3
face=3D"Times New Roman"><span style=3D'font-size:12.0pt'>One question I ha=
ve is
when should TURN allocations be released? TURN would not be used for 99% of
PeerConnections but it cannot be released until it is guaranteed that the o=
ffer
that used this TURN candidate would not be used to create any more PeerConn=
ections.
We would need a separate method to release the TURN candidates, which can b=
e
complicated by the fact that you can have several offers for a given
peerconnection in progress, each with its own TURN candidates that can have
overlapping lifetimes.<br>
<br>
I would actually think that the model where an offer object is created and =
each
PeerConnection is created&nbsp; using the offer object and the answer would=
 be
cleaner. <br>
<br>
One way to implement this, is to define a factory object is used to create
offers. SDP is extracted from an offer object. Once answer is received, a
method in the offer object is used to create the PeerConnection. Once no mo=
re
answers are expected offer object should be released.<br>
<br>
The other option, that would be less disruptive to the curretn API,&nbsp; i=
s
that instead of cloning peerconnection method, a method that returns refere=
nce
to the current offer is used. A cloned peerconnection can be created using =
this
offer. Once final answer is processed by the peerconnection, offer object i=
s released
by it. If multiple answers are expected, offer object can be retrieved usin=
g
the get offer method and stored separately until the moment when answers fo=
r
this particular offer a no longer expected.<br clear=3Dall>
_____________<br>
Roman Shpount<o:p></o:p></span></font></p>

</div>

</body>

</html>

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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On 05/07/2012 11:53 PM, Randell Jesup wrote:
> On 5/7/2012 5:32 PM, Harald Alvestrand wrote:
>> Richard, thank you for doing this! It's definitely the most
>> thought-through proposal I've seen on this question!
>
> Ditto!
>
>>>   5. The CPC object will inherit the local streams, local ICE
>>>      candidates, and local description of the PC.
>>>
>> when it inherits the local streams, do you assume that unless the JS
>> applies some setting, they will also inherit the "enabled/disabled"
>> state, but that the "enabled/disabled" state can be changed
>> independently for each PeerConnection's local streams after the cloning?
>
> I would assume that personally, and that typically after cloning you'd
> do "appropriate" tweaking to new (and old).

As the APIs are designed right now, this is not possible AFAIK. The 
"local streams" is more or less just a list of the streams that has been 
added to the PeerConnection (PC) using "addStream". So if you have one 
PC with one MediaStream added, then clones the PC, the result would be 
that that MediaStream now has two consumers. And if you disable a track, 
that track would be disabled for all consumers.

I think a more flexible solution would be that the new PC does _not_ 
inherit the streams. This way the application can choose to clone the 
MediaStream's attached to the first PC, and add the cloned MS's to the 
new PC, and thereby get independent control of e.g. enable/disable 
state. After all, it is the application that initiates the PC cloning, 
so it knows exactly when it happens and the context, so adding the req. 
to the app to add the stream(s) that should be added would not make it 
more complex IMO.

The possible problem I can see with this approach is that if the set of 
MS's added to the cloned PC is not exactly the same as for the original 
PC, the number of ICE connections needed might not match. But is that 
problem not already there, and must be handled by the ICE agent? The 
cloned PC may connect to an endpoint that does not support multiplexing 
of RTP-streams, or new streams might be added to the cloned PC at any 
time after cloning.

>
>>>   8. The local streams might multicast toward the remote targets
>>>      depending on the directionality attributes independently set for
>>>      each PC and clone.
>>>
>> So far, we've stayed away from anything that involves IP multicast. Are
>> you sure it's worth the potential complexity to try to use this here?
>
> Richard - do you mean "IP Multicast" (tm) or 'multicast' by sending the
> same encoded data to more than one destination in separate RTP streams?
>    (i.e. one encoding being used to simulcast to multiple receivers?)
>
> Simulcast is possible and useful (almost required for Mesh
> Conferencing); IP Multicast has been pretty much ruled out-of-scope IIRC.
>


From harald@alvestrand.no  Tue May  8 00:57:52 2012
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Subject: [rtcweb] SAVPF history (Re:  Final plea about SRTP)
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On 05/04/2012 07:45 PM, Roman Shpount wrote:
>
> On Fri, May 4, 2012 at 1:04 PM, Randell Jesup <randell-ietf@jesup.org 
> <mailto:randell-ietf@jesup.org>> wrote:
>
>     You forget that bid-down includes bid-downs by the JS or server
>     (which are not trusted in our model), not just by on-path attackers.
>
>
> If your session is initiated by HTTPS, using RTP should not be an 
> option (the same way as using HTTP from HTTPS is not normally an 
> option). If your session is HTTP, whole application can be spoofed, so 
> there is no security to begin with.
>
>     I used to work on hardware endpoints that have been using SAVPF
>     since 2004, with hundreds of thousands of units in the field.
>
>
> I thought SAVPF was only standardized in 2008 and AVPF was 
> standardized in 2006. AVPF was discussed for a while though, so I 
> would assumed you worked with something that implemented one of the 
> drafts...
The -00 version of the SAVPF draft is dated 19 October 2003.

According to 
https://datatracker.ietf.org/doc/draft-ietf-avt-profile-savpf/history/ 
publication was requested in February 2006, and it was approved by the 
IESG in November 2007. The publication delay was 3 months.

The technical changes that resulted from these 4 years of work can be 
seen here:

http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url1=draft-ietf-avt-profile-savpf-00.txt&url2=draft-ietf-avt-profile-savpf-12.txt 
<http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url1=draft-ietf-avt-profile-savpf-00.txt&url2=draft-ietf-avt-profile-savpf-12.txt>

                         Harald


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    On 05/04/2012 07:45 PM, Roman Shpount wrote:
    <blockquote
cite="mid:CAD5OKxtJzp-eA_9BpaX1ekt7LwNbQsJcyfEYytwTLXCffUZcGA@mail.gmail.com"
      type="cite"><br clear="all">
      <div class="gmail_quote">On Fri, May 4, 2012 at 1:04 PM, Randell
        Jesup <span dir="ltr">&lt;<a moz-do-not-send="true"
            href="mailto:randell-ietf@jesup.org" target="_blank">randell-ietf@jesup.org</a>&gt;</span>
        wrote:<br>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          You forget that bid-down includes bid-downs by the JS or
          server (which are not trusted in our model), not just by
          on-path attackers.</blockquote>
        <div><br>
          If your session is initiated by HTTPS, using RTP should not be
          an option (the same way as using HTTP from HTTPS is not
          normally an option). If your session is HTTP, whole
          application can be spoofed, so there is no security to begin
          with. <br>
          <br>
        </div>
        <blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
          0.8ex; border-left: 1px solid rgb(204, 204, 204);
          padding-left: 1ex;">
          <div class="im">I used to work on hardware endpoints that have
            been using SAVPF since 2004, with hundreds of thousands of
            units in the field.<br>
          </div>
        </blockquote>
        <div><br>
          I thought SAVPF was only standardized in 2008 and AVPF was
          standardized in 2006. AVPF was discussed for a while though,
          so I would assumed you worked with something that implemented
          one of the drafts...  <br>
        </div>
      </div>
    </blockquote>
    The -00 version of the SAVPF draft is dated 19 October 2003.<br>
    <br>
    According to <a
href="https://datatracker.ietf.org/doc/draft-ietf-avt-profile-savpf/history/">https://datatracker.ietf.org/doc/draft-ietf-avt-profile-savpf/history/</a>
    publication was requested in February 2006, and it was approved by
    the IESG in November 2007. The publication delay was 3 months.<br>
    <br>
    The technical changes that resulted from these 4 years of work can
    be seen here:<br>
    <br>
    <a
href="http://tools.ietf.org/rfcdiff?difftype=--hwdiff&amp;url1=draft-ietf-avt-profile-savpf-00.txt&amp;url2=draft-ietf-avt-profile-savpf-12.txt">http://tools.ietf.org/rfcdiff?difftype=--hwdiff&amp;url1=draft-ietf-avt-profile-savpf-00.txt&amp;url2=draft-ietf-avt-profile-savpf-12.txt</a><br>
    <br>
                            Harald<br>
    <br>
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</html>

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Thread-Topic: Consent freshness - interaction with SDP direction attributes
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Subject: [rtcweb] Consent freshness - interaction with SDP direction attributes
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I was thinking what the interaction between the SDP direction attributes
and the generation of consent freshness (described in
draft-muthu-behave-consent-freshness-00) should be, in response to some
questions posted by Harald. My thought is that the direction attributes
in SDP should have no impact on the generation of consent freshness. As
far as the media connection address and port are valid, the browser
should continue to generate consent freshness requests, irrespective of
whether the stream is sendonly, recvonly, sendrecv or inactive,
analogous to RTCP.

=20

This would:

- Eliminate the need for the browser to depend on the (untrusted) JS to
tell it when to stop/start generating consent requests.

- Keep the NAT bindings always alive.

=20

Comments?

=20

Muthu

=20


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>I was thinking what =
the interaction between the SDP direction attributes and the generation =
of consent freshness (described in =
draft-muthu-behave-consent-freshness-00) should be, in response to some =
questions posted by Harald. My thought is that the direction attributes =
in SDP should have no impact on the generation of consent freshness. As =
far as the media connection address and port are valid, the browser =
should continue to generate consent freshness requests, irrespective of =
whether the stream is sendonly, recvonly, sendrecv or inactive, =
analogous to RTCP.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>This =
would:<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New"'>- Eliminate the =
need for the browser to depend on the (untrusted) JS to tell it when to =
stop/start generating consent requests.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'>- Keep the NAT bindings always alive.<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:10.0pt;font-family:"Courier =
New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
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New"'>Comments?<o:p></o:p></span></p><p class=3DMsoNormal><span =
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New"'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
New"'>Muthu<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier =
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References: <CAD5OKxtSvdu9gMqfb3ptw5aQJt1NZKLJ1UB_vKRWDXCZurD+1w@mail.gmail.com> <BDA69428-93F2-475B-ABBB-5DE539671DD1@iii.ca> <CAD5OKxs+oZj47DrTSnvaLV7-jNEPOkxjZfJuC5F2fo71kB3-4g@mail.gmail.com> <BLU169-DS251D322307BC173FD221AE932F0@phx.gbl> <CAD5OKxvahkBEs6iVuuyrwuYXzcbKKPvVWL5rx02d6DOhtX_0Cg@mail.gmail.com> <4FA3754D.6020004@ericsson.com> <CAD5OKxs3zhxecnXCjsbKzeWNvyJCUy_31pnXKv+orT-T6-FtLg@mail.gmail.com> <4FA40C0F.3000702@jesup.org> <CAD5OKxtJzp-eA_9BpaX1ekt7LwNbQsJcyfEYytwTLXCffUZcGA@mail.gmail.com> <4FA8D1F6.4010103@alvestrand.no>
Date: Tue, 8 May 2012 09:34:03 -0400
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Subject: Re: [rtcweb] SAVPF history (Re: Final plea about SRTP)
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On Tue, May 8, 2012 at 3:57 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> **
> On 05/04/2012 07:45 PM, Roman Shpount wrote:
>
> I used to work on hardware endpoints that have been using SAVPF since
> 2004, with hundreds of thousands of units in the field.
>
>>
> I thought SAVPF was only standardized in 2008 and AVPF was standardized in
> 2006. AVPF was discussed for a while though, so I would assumed you worked
> with something that implemented one of the drafts...
>
> The -00 version of the SAVPF draft is dated 19 October 2003.
>
> According to
> https://datatracker.ietf.org/doc/draft-ietf-avt-profile-savpf/history/publication was requested in February 2006, and it was approved by the IESG
> in November 2007. The publication delay was 3 months.
>
> The technical changes that resulted from these 4 years of work can be seen
> here:
>
>
> http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url1=draft-ietf-avt-profile-savpf-00.txt&url2=draft-ietf-avt-profile-savpf-12.txt
>
>

I would say that something that changes to rfc4585 would be more relevant.
Regardless of the actual changes in the draft, my point is there are very
few actual SIP devices that implement SAVPF or AVPF. Specifying this as the
only profile supported by WebRTC will create yet another challenge to
legacy interop, since it will require not only processing of ICE messages,
and DTLS-SRTP re-encoding, but also RTCP re-generation.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, May 8, 2012 at 3:57 AM=
, Harald Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestra=
nd.no" target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">
<u></u>

 =20
   =20
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000">
    On 05/04/2012 07:45 PM, Roman Shpount wrote:
    <blockquote type=3D"cite">I used to work on hardware endpoints that hav=
e
            been using SAVPF since 2004, with hundreds of thousands of
            units in the field.<br>
          <div class=3D"gmail_quote"><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204,204,204);paddin=
g-left:1ex">
        </blockquote>
        <div><br>
          I thought SAVPF was only standardized in 2008 and AVPF was
          standardized in 2006. AVPF was discussed for a while though,
          so I would assumed you worked with something that implemented
          one of the drafts...=A0 <br>
        </div>
      </div>
    </blockquote>
    The -00 version of the SAVPF draft is dated 19 October 2003.<br>
    <br>
    According to <a href=3D"https://datatracker.ietf.org/doc/draft-ietf-avt=
-profile-savpf/history/" target=3D"_blank">https://datatracker.ietf.org/doc=
/draft-ietf-avt-profile-savpf/history/</a>
    publication was requested in February 2006, and it was approved by
    the IESG in November 2007. The publication delay was 3 months.<br>
    <br>
    The technical changes that resulted from these 4 years of work can
    be seen here:<br>
    <br>
    <a href=3D"http://tools.ietf.org/rfcdiff?difftype=3D--hwdiff&amp;url1=
=3Ddraft-ietf-avt-profile-savpf-00.txt&amp;url2=3Ddraft-ietf-avt-profile-sa=
vpf-12.txt" target=3D"_blank">http://tools.ietf.org/rfcdiff?difftype=3D--hw=
diff&amp;url1=3Ddraft-ietf-avt-profile-savpf-00.txt&amp;url2=3Ddraft-ietf-a=
vt-profile-savpf-12.txt</a><span class=3D"HOEnZb"><font color=3D"#888888"><=
br>

    =A0=A0=A0 </font></span><br></div></blockquote></div><br>I would say th=
at something that changes to rfc4585 would be more relevant. Regardless of =
the actual changes in the draft, my point is there are very few actual SIP =
devices that implement SAVPF or AVPF. Specifying this as the only profile s=
upported by WebRTC will create yet another challenge to legacy interop, sin=
ce it will require not only processing of ICE messages, and DTLS-SRTP re-en=
coding, but also RTCP re-generation.<br>
_____________<br>Roman Shpount<br>
<br>

--001517592bfcaa56f404bf86707d--

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Date: Tue, 8 May 2012 09:40:30 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] Proposal for PeerConnection cloning
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On Mon, May 7, 2012 at 10:10 PM, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

>  You raise a good point since some candidates may be released before
> cloning occurs.  So this is another bug in the procedure.  I would suggest
> that the configuration also be passed to the CPC and that if any of the
> required candidates are released before cloning occurs that they be
> reallocated.****
>
> ** **
>
> Alternately, we could recommend that candidates be kept allocated longer
> than usual and that if cloning is attempted after any of the candidates are
> released then the cloning procedure fails.****
>
> ** **
>
> The latter approach should be sufficient for SIP forking scenarios, but
> others may prefer the generality of the former.  Or maybe both?  Any votes?
> ****
>
> ** **
>

My suggestion was that instead of the clone method to have a method that
returns reference to the offer object. This separate object holds the
candidates while references to it exist. When this object is passed during
creation of peer connection, peer connection is initialized as a clone of
an existing peer connection This way application has the explicit control
of the ICE candidate lifetimes. I would think this would be the most
flexible.
_____________
Roman Shpount

--0015175cd0dabc3c2304bf868771
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<br clear=3D"all"><div class=3D"gmail_quote">On Mon, May 7, 2012 at 10:10 P=
M, Ejzak, Richard P (Richard) <span dir=3D"ltr">&lt;<a href=3D"mailto:richa=
rd.ejzak@alcatel-lucent.com" target=3D"_blank">richard.ejzak@alcatel-lucent=
.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 link=3D"blue" vlink=3D"#606420" lang=3D"EN-US">

<div>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">You raise a good point since =
some
candidates may be released before cloning occurs. =A0So this is another bug
in the procedure.=A0 I would suggest that the configuration also be passed
to the CPC and that if any of the required candidates are released before
cloning occurs that they be reallocated.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">Alternately, we could recomme=
nd that
candidates be kept allocated longer than usual and that if cloning is attem=
pted
after any of the candidates are released then the cloning procedure fails.<=
u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy">The latter approach should be=
 sufficient
for SIP forking scenarios, but others may prefer the generality of the form=
er.=A0
Or maybe both?=A0 Any votes?<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font color=3D"navy" face=3D"Arial"><span style=3D"f=
ont-size:10.0pt;font-family:Arial;color:navy"><u></u>=A0<u></u></span></fon=
t></p></div></div></blockquote><div><br>My suggestion was that instead of t=
he clone method to have a method that returns reference to the offer object=
. This separate object holds the candidates while references to it exist. W=
hen this object is passed during creation of peer connection, peer connecti=
on is initialized as a clone of an existing peer connection This way applic=
ation has the explicit control of the ICE candidate lifetimes. I would thin=
k this would be the most flexible.=A0 <br>
</div></div>_____________<br>Roman Shpount<br>

--0015175cd0dabc3c2304bf868771--

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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SAVPF history (Re: Final plea about SRTP)
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On 2012-05-08 15:34, Roman Shpount wrote:

> I would say that something that changes to rfc4585 would be more
> relevant. Regardless of the actual changes in the draft, my point is
> there are very few actual SIP devices that implement SAVPF or AVPF.
> Specifying this as the only profile supported by WebRTC will create yet
> another challenge to legacy interop, since it will require not only
> processing of ICE messages, and DTLS-SRTP re-encoding, but also RTCP
> re-generation.

Hi,

Here I have to protest that it increases the barrier. First of all the S
in SAVPF is for using SRTP. That is already agreed on in this WG.
Secondly the F stands for the feedback part. That consists of two parts:

1) New Timer Rules

2) Feedback RTCP packets

The second is actually negotiated on a per individual feedback format
basis. So only the first is what really what would be mandated by a
WebRTC requirement of SAVPF. From my perspective to get the RTP/RTCP
functionalities needed to give a good user experience we really need the
timing rules. When it comes to legacy interop a signalling gateway can
actually produce a SAVPF configuration that is fully interopable with an
SAVP end-point.

The configuration required to make this interoperable are: use a=rtcp-fb
to sett trr-int to 3.5 to 4 seconds and ensure that the same or at least
sufficient RTCP bandwidth is configured and that no feedback format are
agreed on being used.

So please don't be afraid of SAVPF.

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
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
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Subject: Re: [rtcweb] Question about JSEP createOffer
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Richard,

That is an interesting point. Could you give a short example of when such a
"full" re-OFFER would be used? There are a few ways we could handle this
case.

On Sun, May 6, 2012 at 2:40 PM, Ejzak, Richard P (Richard) <
richard.ejzak@alcatel-lucent.com> wrote:

>  I have a question about something I came across in the JSEP draft.****
>
> ** **
>
> 6.1.2 of JSEP draft 0 has the following text regarding =E2=80=9CcreateOff=
er=E2=80=9D:****
>
> ** **
>
>    As an offer, the generated SDP will contain the full set of****
>
>    capabilities supported by the session (as opposed to an answer, which*=
*
> **
>
>    will include only a specific negotiated subset to use); for each SDP**=
*
> *
>
>    line, the generation of the SDP must follow the appropriate process***=
*
>
>    for generating an offer. In the event createOffer is called after the*=
*
> **
>
>    session is established, createOffer will generate an offer that is****
>
>    compatible with the current session, incorporating any changes that***=
*
>
>    have been made to the session since the last complete offer-answer****
>
>    exchange, such as addition or removal of streams. If no changes have**=
*
> *
>
>    been made, the offer will be identical to the current local****
>
>    description.****
>
> ** **
>
> The first and last sentences might conflict if the SDP Answerer later
> wants to create a new Offer.  The local configuration based on the first
> offer/answer will contain only the selected media configuration, whereas =
it
> is desirable for the new Offer to contain the full set of capabilities of
> the (new) Offerer.  I think this text needs to change to reflect this cas=
e.
>  Am I missing something?****
>
> ** **
>
> Richard****
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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Richard,<div><br></div><div>That is an interesting point. Could you give a =
short example of when such a &quot;full&quot; re-OFFER would be used?=C2=A0=
There are a few ways we could handle this case.</div><div><br><div class=3D=
"gmail_quote">

On Sun, May 6, 2012 at 2:40 PM, Ejzak, Richard P (Richard) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:richard.ejzak@alcatel-lucent.com" target=3D"_blank"=
>richard.ejzak@alcatel-lucent.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">










<div lang=3D"EN-US" link=3D"blue" vlink=3D"#606420">

<div>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">I have a question about something I came across in the =
JSEP
draft.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">6.1.2 of JSEP draft 0 has the following text regarding =
=E2=80=9CcreateOffer=E2=80=9D:<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 As an offer, the generated SDP will contain the full set
of<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 capabilities supported by the session (as opposed to an
answer, which<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 will include only a specific negotiated subset to use);
for each SDP<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 line, the generation of the SDP must follow the
appropriate process<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 for generating an offer. In the event createOffer is
called after the<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 session is established, createOffer will generate an offer
that is<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 compatible with the current session, incorporating any
changes that<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 have been made to the session since the last complete
offer-answer<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 exchange, such as addition or removal of streams. If no
changes have<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 been made, the offer will be identical to the current
local<u></u><u></u></span></font></p>

<p><font face=3D"Courier New"><span style=3D"font-size:10.0pt">=C2=A0=C2=A0=
 description.<u></u><u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">The first and last sentences might conflict if the SDP
Answerer later wants to create a new Offer. =C2=A0The local configuration b=
ased
on the first offer/answer will contain only the selected media configuratio=
n,
whereas it is desirable for the new Offer to contain the full set of
capabilities of the (new) Offerer. =C2=A0I think this text needs to change =
to
reflect this case. =C2=A0Am I missing something?<span class=3D"HOEnZb"><fon=
t color=3D"#888888"><u></u><u></u></font></span></span></font></p><span cla=
ss=3D"HOEnZb"><font color=3D"#888888">

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial"><u></u>=C2=A0<u></u></span></font></p>

<p class=3D"MsoNormal"><font face=3D"Arial"><span style=3D"font-size:10.0pt=
;font-family:Arial">Richard<u></u><u></u></span></font></p>

</font></span></div>

</div>


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

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Subject: Re: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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This "SW encoding chews battery" claim is often mentioned, but in a video
chat, the power draw from the screen and network are often 10x the CPU
power draw, which is typically < 1W (for ARM). So there is very little
impact on talk time between HW and SW encode.

Mobile devices are often limited in terms of what resolutions they can
encode in software, because they have less overall compute power than PCs.
In my experience, that is the factor to consider, not battery life.

On Fri, May 4, 2012 at 6:56 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

>
> > I am more worried about the guy implementing a WebRTC security camera
> > that uses an embedded Linux kernel and a software video encoder. Each
> > unit might "produce" video 24x7. But there is no MPEG-LA licensed
> > browser or OS or encoder chip to fall back on. The whole product might
> > sell for less than it might cost him to license the codec.
>
> [BA]  We are rapidly moving toward inclusion of hardware video
> encode/decode by default on battery-powered devices (mobile handsets,
> tablets, etc.).
>
> Since software encode/decode is processor (and battery) intensive, the
> latest generation of webcams have video encode/decode built-in.  For
> example, see:
>
> http://www.ehomeupgrade.com/2011/06/02/skype-certified-webcams-tote-built-in-720p-h-264-video-encoders/
>
> When handled purely in software, it is quite difficult to enable video
> sessions of substantial duration without draining the battery.
>
> While my assumption is that hardware support for encode/decode will become
> less codec-specific over time,  that is typically not the case today.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--0003255630a262479404bf8791cc
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

This &quot;SW encoding chews battery&quot; claim is often mentioned, but in=
 a video chat, the power draw from the screen and network are often 10x the=
 CPU power draw, which is typically &lt; 1W (for ARM). So there is very lit=
tle impact on talk time between HW and SW encode.=C2=A0<div>

<br></div><div>Mobile devices are often limited in terms of what resolution=
s they can encode in software, because they have less overall compute power=
 than PCs. In my experience, that is the factor to consider, not battery li=
fe.<br>

<br><div class=3D"gmail_quote">On Fri, May 4, 2012 at 6:56 PM, Bernard Abob=
a <span dir=3D"ltr">&lt;<a href=3D"mailto:bernard_aboba@hotmail.com" target=
=3D"_blank">bernard_aboba@hotmail.com</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">




<div><div dir=3D"ltr">
<br><div><div class=3D"im">&gt; I am more worried about the guy implementin=
g a WebRTC security camera<br>&gt; that uses an embedded Linux kernel and a=
 software video encoder. Each<br>&gt; unit might &quot;produce&quot; video =
24x7. But there is no MPEG-LA licensed<br>

&gt; browser or OS or encoder chip to fall back on. The whole product might=
<br>&gt; sell for less than it might cost him to license the codec.<br><br>=
</div>[BA]=C2=A0 We are rapidly moving toward inclusion of hardware video e=
ncode/decode by default on battery-powered devices (mobile handsets, tablet=
s, etc.). <br>

<br>Since software encode/decode is processor (and battery) intensive, the =
latest generation of webcams have video encode/decode built-in.=C2=A0 For e=
xample, see:<br><a href=3D"http://www.ehomeupgrade.com/2011/06/02/skype-cer=
tified-webcams-tote-built-in-720p-h-264-video-encoders/" target=3D"_blank">=
http://www.ehomeupgrade.com/2011/06/02/skype-certified-webcams-tote-built-i=
n-720p-h-264-video-encoders/</a><br>

<br>When handled purely in software, it is quite difficult to enable video =
sessions of substantial duration without draining the battery.=C2=A0=C2=A0 =
<br><br>While my assumption is that hardware support for encode/decode will=
 become less codec-specific over time,=C2=A0 that is typically not the case=
 today.<br>

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

--0003255630a262479404bf8791cc--

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From: Justin Uberti <juberti@google.com>
Date: Tue, 8 May 2012 11:09:58 -0400
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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The conclusion on this thread seems to be that the right way to address
this problem is via PeerConnection cloning, and that no changes are needed
to the JSEP state machine. I'll work with Richard to figure out how we
should extend JSEP to support cloning.

On Thu, May 3, 2012 at 5:16 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >>>>You could also choose not to alert the remote user until the ICE
> procedures have finished - more or less using ICE with preconditions.
> >>>>
> >>>> Of course, that is going to trigger actions in STUN/TURN servers,
> even if the called party won't accept the call, but at least from a user
> >>>> experience perspective that is still better than lifting the hook,
> and having to wait for whatever-time-it-takes-for-ICE-to-finish seconds
> before one can start to talk.
> >>>
> >>> This also has a downside of disclosing user's IP to the caller without
> the user confirmation. For a lot of applications this can be serious
> security flaw.
> >>
> >> The client can still log when ICE procedures occur.
> >>
> >> Because, even if I wait until your phone starts to ring, most likely I
> would still get your IP address without user confirmation (speaking in SIP
> terms, phones normally don't ask for user permission before they send 18x
> with SDP), eventhough you would easier be made aware of that it happens.
> >
> > Another alternative is of course to do ICE with an SBC, and/or having an
> SBC doing ICE on behalf of you :)
> >
> >
> > This is true for SIP but was as far as I know was specifically designed
> around in WebRTC. WebRTC signaling server acts as B2BUA so any type of
> ringing notification goes through the web server and does not need to carry
> any type of client address information. The client address information is
> only provided
> > when SDP answer is sent or ICE negotiation is started.
>
> Assuming you are only going to make user confirmation (read: accept calls)
> in cases where you absolutely sure that the caller isn't someone trying to
> get your IP address.
>
> But, never the less, having a solution where I first have to give user
> confirmation, and then wait until I can start to talk, is probably
> something most people want to avoid. Depending upon, of course, how long
> the waiting time is.
>
> Regards,
>
> Christer
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--20cf300513f6f0f06604bf87c89e
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The conclusion on this thread seems to be that the right way to address thi=
s problem is via PeerConnection cloning, and that no changes are needed to =
the JSEP state machine. I&#39;ll work with Richard to figure out how we sho=
uld extend JSEP to support cloning.<br>

<br><div class=3D"gmail_quote">On Thu, May 3, 2012 at 5:16 PM, Christer Hol=
mberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsson.co=
m" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex">

<div class=3D"HOEnZb"><div class=3D"h5">Hi,<br>
<br>
&gt;&gt;&gt;&gt;You could also choose not to alert the remote user until th=
e ICE procedures have finished - more or less using ICE with preconditions.=
<br>
&gt;&gt;&gt;&gt;<br>
&gt;&gt;&gt;&gt; Of course, that is going to trigger actions in STUN/TURN s=
ervers, even if the called party won&#39;t accept the call, but at least fr=
om a user<br>
&gt;&gt;&gt;&gt; experience perspective that is still better than lifting t=
he hook, and having to wait for whatever-time-it-takes-for-ICE-to-finish se=
conds before one can start to talk.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; This also has a downside of disclosing user&#39;s IP to the ca=
ller without the user confirmation. For a lot of applications this can be s=
erious security flaw.<br>
&gt;&gt;<br>
&gt;&gt; The client can still log when ICE procedures occur.<br>
&gt;&gt;<br>
&gt;&gt; Because, even if I wait until your phone starts to ring, most like=
ly I would still get your IP address without user confirmation (speaking in=
 SIP terms, phones normally don&#39;t ask for user permission before they s=
end 18x with SDP), eventhough you would easier be made aware of that it hap=
pens.<br>


&gt;<br>
&gt; Another alternative is of course to do ICE with an SBC, and/or having =
an SBC doing ICE on behalf of you :)<br>
&gt;<br>
&gt;<br>
&gt; This is true for SIP but was as far as I know was specifically designe=
d around in WebRTC. WebRTC signaling server acts as B2BUA so any type of ri=
nging notification goes through the web server and does not need to carry a=
ny type of client address information. The client address information is on=
ly provided<br>


&gt; when SDP answer is sent or ICE negotiation is started.<br>
<br>
</div></div>Assuming you are only going to make user confirmation (read: ac=
cept calls) in cases where you absolutely sure that the caller isn&#39;t so=
meone trying to get your IP address.<br>
<br>
But, never the less, having a solution where I first have to give user conf=
irmation, and then wait until I can start to talk, is probably something mo=
st people want to avoid. Depending upon, of course, how long the waiting ti=
me is.<br>


<div class=3D"HOEnZb"><div class=3D"h5"><br>
Regards,<br>
<br>
Christer<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>

--20cf300513f6f0f06604bf87c89e--

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From: Roman Shpount <roman@telurix.com>
To: Justin Uberti <juberti@google.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Question about JSEP createOffer
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On Tue, May 8, 2012 at 10:46 AM, Justin Uberti <juberti@google.com> wrote:

> Richard,
>
> That is an interesting point. Could you give a short example of when such
> a "full" re-OFFER would be used? There are a few ways we could handle this
> case.
>
>
Actually I was the person who argued that "full" capabilities should be
present in original SIP O/A, when new offer is generated in an existing
session. The use case for this are B2BUA that connect an existing session
to a new call or another existing session. Typically, B2BUA would ask one
of the call parties for the offer (send an INVITE with no body in SIP), get
the new offer, and send this offer to the newly placed call. In order for
this to work, all the calling party capabilities should be listed in this
new offer, in particular all the codecs and all the communications types
(audio, video, text, etc), or call would not be successfully established.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Tue, May 8, 2012 at 10:46 A=
M, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com=
" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">

Richard,<div><br></div><div>That is an interesting point. Could you give a =
short example of when such a &quot;full&quot; re-OFFER would be used?=A0The=
re are a few ways we could handle this case.</div><div><br></div></blockquo=
te>

</div><br>Actually I was the person who argued that &quot;full&quot; capabi=
lities should be present in original SIP O/A, when new offer is generated i=
n an existing session. The use case for this are B2BUA that connect an exis=
ting session to a new call or another existing session. Typically, B2BUA wo=
uld ask one of the call parties for the offer (send an INVITE with no body =
in SIP), get the new offer, and send this offer to the newly placed call. I=
n order for this to work, all the calling party capabilities should be list=
ed in this new offer, in particular all the codecs and all the communicatio=
ns types (audio, video, text, etc), or call would not be successfully estab=
lished.<br>

_____________<br>Roman Shpount<br>
<br>

--e89a8fb20896afe13f04bf881a1e--

From martin.thomson@gmail.com  Tue May  8 08:40:48 2012
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From: Martin Thomson <martin.thomson@gmail.com>
To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Subject: Re: [rtcweb] Consent freshness - interaction with SDP direction attributes
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On 8 May 2012 03:20, Muthu Arul Mozhi Perumal (mperumal)
<mperumal@cisco.com> wrote:
> My thought is that the direction attributes in
> SDP should have no impact on the generation of consent freshness. As far as
> the media connection address and port are valid, the browser should continue
> to generate consent freshness requests, irrespective of whether the stream
> is sendonly, recvonly, sendrecv or inactive, analogous to RTCP.

I tend to agree.

Where I disagree: consent freshness does not require a new mechanism.
If a Binding request is used, it's trivial to maintain consent with
ICE-lite endpoints and I don't see any significant advantage in
avoiding a simple SHA-1.

> - Eliminate the need for the browser to depend on the (untrusted) JS to tell
> it when to stop/start generating consent requests.

This isn't a problem.  The browser needs to have consent freshness
when it sends a packet, simple.  The only possible concern is
clipping, where consent freshness has expired, causing the browser to
drop packets instead of sending them.  If consent freshness wasn't
maintained, then an unmute on a stream would force the browser to
refresh consent prior to sending packets.  Add packet loss and this
could be annoying.

--Martin

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On 7 May 2012 19:10, Ejzak, Richard P (Richard)
<richard.ejzak@alcatel-lucent.com> wrote:
> You raise a good point since some candidates may be released before cloning
> occurs.  So this is another bug in the procedure.  I would suggest that the
> configuration also be passed to the CPC and that if any of the required
> candidates are released before cloning occurs that they be reallocated.

Releasing a TURN candidate is a real possibility.  The same could
apply to local candidates (and server reflexive), though that will
depend on implementation.  Getting the same one back is unlikely for
TURN, though it might be possible for local.

In all these cases, answers that arrive late will fail.

> Alternately, we could recommend that candidates be kept allocated longer
> than usual and that if cloning is attempted after any of the candidates are
> released then the cloning procedure fails.

I'd be against making any such recommendation for browser
implementations.  The signaling application is more likely to know
best in this scenario.  In that case, it can make the call to hold
candidates open.  PR_ANSWER seems to have the right semantics.

--Martin

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+1

Many companies shipping in high volume pay the annual cap and have a
marginal cost at or below 10c each and the low volumes (<100K) are free
as noted.       That leaves non-profits (or have no workable freemium
model) shipping in the millions to be about the only segment with an
hurdle.

=20

From: Stephan Wenger [mailto:stewe@stewe.org]=20
Sent: Friday, May 04, 2012 12:09 PM
To: Dean Willis; Lorenzo Miniero
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Discussion on codec choices from a developer who
doesn't come to IETF

=20

=20

From: Dean Willis <dean.willis@softarmor.com>
Date: Friday, 4 May, 2012 16:10=20

[...]

[...]

	I am more worried about the guy implementing a WebRTC security
camera that uses an embedded Linux kernel and a software video encoder.
Each unit might "produce" video 24x7. But there is no MPEG-LA licensed
browser or OS or encoder chip to fall back on. The whole product might
sell for less than it might cost him to license the codec.

Really?  The MPEG-LA license terms for one encoder/decoder: $0 for the
first 100,000 units per year.  $0.20 for the next several millions,
after that $0.10 until the cap is reached.  And, the MPEG-LA license is
"take it or leave it" (at least for small fish like your hypothetical
camera guy), so even the lawyer cost for review should be minimal.

=20

No, I continue to believe that the MPEG-LA license is not a hurdle for
anyone but those giving stuff away for free, or having a business model
that disallows them to pay licensing fees.  It's the patents not covered
by that license that you may worry about.

=20

Stephan =20


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style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Many companies shipping in high volume pay the annual cap and have a =
marginal cost at or below 10c each and the low volumes (&lt;100K) are =
free as noted.&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp;That leaves =
non-profits (or have no workable freemium model) shipping in the =
millions to be about the only segment with an =
hurdle.<o:p></o:p></span></p><p class=3DMsoNormal><span =
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</b><span style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> =
Stephan Wenger [mailto:stewe@stewe.org] <br><b>Sent:</b> Friday, May 04, =
2012 12:09 PM<br><b>To:</b> Dean Willis; Lorenzo Miniero<br><b>Cc:</b> =
rtcweb@ietf.org<br><b>Subject:</b> Re: [rtcweb] Discussion on codec =
choices from a developer who doesn't come to =
IETF<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
><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:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>From: </span></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>Dean Willis &lt;<a =
href=3D"mailto:dean.willis@softarmor.com">dean.willis@softarmor.com</a>&g=
t;<br><b>Date: </b>Friday, 4 May, 2012 16:10 =
<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>[&#8230;]<o:p></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:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>[&#8230;]</span></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
><o:p></o:p></span></p></div><div><div><blockquote =
style=3D'margin-left:30.0pt;margin-top:5.0pt;margin-right:0in;margin-bott=
om:5.0pt'><p><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>I am more worried about the guy implementing a WebRTC security camera =
that uses an embedded Linux kernel and a software video encoder. Each =
unit might &quot;produce&quot; video 24x7. But there is no MPEG-LA =
licensed browser or OS or encoder chip to fall back on. The whole =
product might sell for less than it might cost him to license the =
codec.<o:p></o:p></span></p></blockquote></div></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:blue'>=
Really? &nbsp;The MPEG-LA license terms for one encoder/decoder: $0 for =
the first 100,000 units per year. &nbsp;$0.20 for the next several =
millions, after that $0.10 until the cap is reached. &nbsp;And, the =
MPEG-LA license is &quot;take it or leave it&quot; (at least for small =
fish like your hypothetical camera guy), so even the lawyer cost for =
review should be minimal.</span><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif"'><o:p></o:p>=
</span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:blue'>=
No, I continue to believe that the MPEG-LA license is not a hurdle for =
anyone but those giving stuff away for free, or having a business model =
that disallows them to pay licensing fees. &nbsp;It's the patents not =
covered by that license that you may worry about.</span><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif"'><o:p></o:p>=
</span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif"'><o:p>&nbsp;=
</o:p></span></p></div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:blue'>=
Stephan &nbsp;</span><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p></o:p></span></p></div></div></div></body></html>
------_=_NextPart_001_01CD2D38.C93C3173--

From tterriberry@mozilla.com  Tue May  8 09:48:45 2012
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Justin Uberti wrote:
> This "SW encoding chews battery" claim is often mentioned, but in a
> video chat, the power draw from the screen and network are often 10x the
> CPU power draw, which is typically < 1W (for ARM). So there is very
> little impact on talk time between HW and SW encode.
>
> Mobile devices are often limited in terms of what resolutions they can
> encode in software, because they have less overall compute power than
> PCs. In my experience, that is the factor to consider, not battery life.

Even if they could encode higher resolutions, they typically don't have 
the bandwidth to push out HD content at several megabits per second (or 
it would be prohibitively expensive if they did). So I think mobile 
videoconferencing resolutions around 360p are much more realistic, and 
perfectly suitable for software. At this point I'll just refer people to 
the guy who posted here from TI:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg03785.html

From randell-ietf@jesup.org  Tue May  8 10:54:01 2012
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Subject: Re: [rtcweb] SAVPF history (Re: Final plea about SRTP)
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On 5/8/2012 9:34 AM, Roman Shpount wrote:
>
> On Tue, May 8, 2012 at 3:57 AM, Harald Alvestrand <harald@alvestrand.no
> <mailto:harald@alvestrand.no>> wrote:
>
>     __
>     On 05/04/2012 07:45 PM, Roman Shpount wrote:
>>     I used to work on hardware endpoints that have been using SAVPF
>>     since 2004, with hundreds of thousands of units in the field.
>>
>>
>>     I thought SAVPF was only standardized in 2008 and AVPF was
>>     standardized in 2006. AVPF was discussed for a while though, so I
>>     would assumed you worked with something that implemented one of
>>     the drafts...
>     The -00 version of the SAVPF draft is dated 19 October 2003.
>
>     According to
>     https://datatracker.ietf.org/doc/draft-ietf-avt-profile-savpf/history/
>     publication was requested in February 2006, and it was approved by
>     the IESG in November 2007. The publication delay was 3 months.
>
>     The technical changes that resulted from these 4 years of work can
>     be seen here:
>
>     http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url1=draft-ietf-avt-profile-savpf-00.txt&url2=draft-ietf-avt-profile-savpf-12.txt
>     <http://tools.ietf.org/rfcdiff?difftype=--hwdiff&url1=draft-ietf-avt-profile-savpf-00.txt&url2=draft-ietf-avt-profile-savpf-12.txt>
>
>
> I would say that something that changes to rfc4585 would be more
> relevant. Regardless of the actual changes in the draft, my point is
> there are very few actual SIP devices that implement SAVPF or AVPF.

Perhaps, though maybe it's more common in video (SIP INFO for IDR 
requests "sucks dead gerbils through garden hoses" - old Amiga slang 
phrase).

> Specifying this as the only profile supported by WebRTC will create yet
> another challenge to legacy interop, since it will require not only
> processing of ICE messages, and DTLS-SRTP re-encoding, but also RTCP
> re-generation.

AVPF has the advantage of being backward-compatible with AVP in almost 
all cases.  We used SAVPF internally, but typically hid that from the 
outside world by advertising AVP (this was long before SDP cap-neg was 
finalized).  (Has anyone truly implemented it?  In open source?)  IIRC 
this is even spoken to in the spec.  (We used AVP/AVPF instead of 
SAVP/SAVPF because we used the non-standardized "best effort encryption" 
draft I worked on with Hadriel, as an alternative/stopgap until cap-neg 
"worked".)

-- 
Randell Jesup
randell-ietf@jesup.org

From keith.drage@alcatel-lucent.com  Tue May  8 11:35:58 2012
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From: "DRAGE, Keith (Keith)" <keith.drage@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 8 May 2012 20:35:52 +0200
Thread-Topic: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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Subject: Re: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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I suspect the key question is being missed here to make the arguments look =
more attractive.

Most mobile devices (at least those on 3GPP standards) trying to do anythin=
g other than RTCWEB with their screens and cameras will need to support H.2=
64 anyway.

So the discussion needs to be around what are the pros and cons of supporti=
ng both VP8 and H.264 on the same mobile, as opposed to only having to supp=
ort H.264.

Regards

Keith

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Timothy B. Terriberry
Sent: 08 May 2012 17:49
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Discussion on codec choices from a developer who does=
n't come to IETF

Justin Uberti wrote:
> This "SW encoding chews battery" claim is often mentioned, but in a
> video chat, the power draw from the screen and network are often 10x the
> CPU power draw, which is typically < 1W (for ARM). So there is very
> little impact on talk time between HW and SW encode.
>
> Mobile devices are often limited in terms of what resolutions they can
> encode in software, because they have less overall compute power than
> PCs. In my experience, that is the factor to consider, not battery life.

Even if they could encode higher resolutions, they typically don't have=20
the bandwidth to push out HD content at several megabits per second (or=20
it would be prohibitively expensive if they did). So I think mobile=20
videoconferencing resolutions around 360p are much more realistic, and=20
perfectly suitable for software. At this point I'll just refer people to=20
the guy who posted here from TI:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg03785.html
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From ted.ietf@gmail.com  Tue May  8 14:40:49 2012
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On Tue, May 8, 2012 at 11:35 AM, DRAGE, Keith (Keith)
<keith.drage@alcatel-lucent.com> wrote:
> Most mobile devices (at least those on 3GPP standards) trying to do anything other than RTCWEB with their screens and cameras will need to support H.264 anyway.
>

My faulty memory is that the availability of hardware acceleration for
certain audio codecs used to be restricted and that VoIP clients on
3GPP devices had no access to them.  Is that situation cleared up?
Can any installed application now access hardware accelerated codec
support?

regards,

Ted

From harald@alvestrand.no  Tue May  8 14:54:35 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Consent freshness - revisiting the RTCP option
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Just because I realized I didnt understand something, I ask.....

We rejected RTCP RR as a consent freshness mechanism because RR is 
trivial to fake.
But - now we have SRTP as mandatory-to-use, which means that all RTCP 
RRs are integrity protected, origin authenticated and replay protected 
(do I have that right?).

What is the reason why this is not sufficient protection to use RTCP RR 
as a consent freshness mechanism?

                         Harald


From ekr@rtfm.com  Tue May  8 15:00:29 2012
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Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
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On Tue, May 8, 2012 at 2:54 PM, Harald Alvestrand <harald@alvestrand.no> wrote:
> Just because I realized I didnt understand something, I ask.....
>
> We rejected RTCP RR as a consent freshness mechanism because RR is trivial
> to fake.
> But - now we have SRTP as mandatory-to-use, which means that all RTCP RRs
> are integrity protected, origin authenticated and replay protected (do I
> have that right?).
>
> What is the reason why this is not sufficient protection to use RTCP RR as a
> consent freshness mechanism?

This isn't a complete analysis, but if you are using SDES for key management,
then the site knows the SRTCP keys, so I don't *think* SRTCP is buying you
much. I haven't thought through this completely though, so maybe there is
still some additional value.

-Ekr

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Subject: Re: [rtcweb] Discussion on codec choices from a developer who doesn't come to IETF
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On 5/8/2012 2:35 PM, DRAGE, Keith (Keith) wrote:
> I suspect the key question is being missed here to make the arguments look more attractive.
>
> Most mobile devices (at least those on 3GPP standards) trying to do anything other than RTCWEB with their screens and cameras will need to support H.264 anyway.
>
> So the discussion needs to be around what are the pros and cons of supporting both VP8 and H.264 on the same mobile, as opposed to only having to support H.264.

This only has any real import if you're talking hardware 
implementations.  Hardware H.264 doesn't preclude software VP8 (or vice 
versa), and the software is available, and a trivial amount of flash 
space.  So I see nothing to discuss around "both VP8 and H.264 vs. just 
H.264 on mobile".

-- 
Randell Jesup
randell-ietf@jesup.org


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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Roman Shpount <roman@telurix.com>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Question about JSEP createOffer
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I agree with Richard & Roman. "Full" re-OFFER MUST supported required for c=
ouple of mid session scenario like


1)      The session is established with audio and then video shall be added=
 in the middle in case complete offer is provided in re-OFFER.

2)       It is possible to change the codec in the middle of the Session as=
 Roman mentioned.



From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Roman Shpount
Sent: Tuesday, May 08, 2012 9:03 PM
To: Justin Uberti
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Question about JSEP createOffer


On Tue, May 8, 2012 at 10:46 AM, Justin Uberti <juberti@google.com<mailto:j=
uberti@google.com>> wrote:
Richard,

That is an interesting point. Could you give a short example of when such a=
 "full" re-OFFER would be used? There are a few ways we could handle this c=
ase.


Actually I was the person who argued that "full" capabilities should be pre=
sent in original SIP O/A, when new offer is generated in an existing sessio=
n. The use case for this are B2BUA that connect an existing session to a ne=
w call or another existing session. Typically, B2BUA would ask one of the c=
all parties for the offer (send an INVITE with no body in SIP), get the new=
 offer, and send this offer to the newly placed call. In order for this to =
work, all the calling party capabilities should be listed in this new offer=
, in particular all the codecs and all the communications types (audio, vid=
eo, text, etc), or call would not be successfully established.
_____________
Roman Shpount

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<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1F497D">I agree with Richard &amp=
; Roman. &#8220;Full&#8221; re-OFFER MUST supported required for couple of =
mid session scenario like
<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><=
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;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">The session is es=
tablished with audio and then video shall be added in the middle in case co=
mplete offer is provided in re-OFFER.<o:p></o:p></span></p>
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;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span style=3D"font-size:11.0pt;font-family:=
&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D">&nbsp;It is possi=
ble to change the codec in the middle of the Session as Roman mentioned.<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><=
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D"><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"><o:p>&nbsp;</o:p></span><=
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<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>Roman Shpount<br>
<b>Sent:</b> Tuesday, May 08, 2012 9:03 PM<br>
<b>To:</b> Justin Uberti<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Question about JSEP createOffer<o:p></o:p></sp=
an></p>
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal"><br clear=3D"all">
<o:p></o:p></p>
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<p class=3D"MsoNormal">On Tue, May 8, 2012 at 10:46 AM, Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoNormal">Richard,<o:p></o:p></p>
<div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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<div>
<p class=3D"MsoNormal">That is an interesting point. Could you give a short=
 example of when such a &quot;full&quot; re-OFFER would be used?&nbsp;There=
 are a few ways we could handle this case.<o:p></o:p></p>
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<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
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<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Actually I was the person who argued that &quot;full&quot; capabilities sho=
uld be present in original SIP O/A, when new offer is generated in an exist=
ing session. The use case for this are B2BUA that connect an existing sessi=
on to a new call or another existing session.
 Typically, B2BUA would ask one of the call parties for the offer (send an =
INVITE with no body in SIP), get the new offer, and send this offer to the =
newly placed call. In order for this to work, all the calling party capabil=
ities should be listed in this new
 offer, in particular all the codecs and all the communications types (audi=
o, video, text, etc), or call would not be successfully established.<br>
_____________<br>
Roman Shpount<o:p></o:p></p>
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From pravindran@sonusnet.com  Tue May  8 23:34:46 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Justin Uberti <juberti@google.com>, Christer Holmberg <christer.holmberg@ericsson.com>
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Date: Wed, 9 May 2012 06:34:33 +0000
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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From Ranjit.Avasarala@Polycom.com  Tue May  8 23:49:24 2012
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 9 May 2012 14:49:13 +0800
Thread-Topic: Mapping between SIP and ROAP/JSEP
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Hi

Is there any interest in the coming up with a mapping between SIP and ROAP/=
JSEP protocols?  Though JSEP is a set of APIs as of now which map closely w=
ith ROAP APIs, ROAP has messages like OFFER,ANSWER and OK which can be mapp=
ed to SIP.

Comment?

Thanks

Regards
Ranjit


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<html xmlns:v=3D"urn:schemas-microsoft-com:vml" xmlns:o=3D"urn:schemas-micr=
osoft-com:office:office" xmlns:w=3D"urn:schemas-microsoft-com:office:word" =
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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue vli=
nk=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal>Hi<o:p></o:p></p=
><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Is there an=
y interest in the coming up with a mapping between SIP and ROAP/JSEP protoc=
ols?&nbsp; Though JSEP is a set of APIs as of now which map closely with RO=
AP APIs, ROAP has messages like OFFER,ANSWER and OK which can be mapped to =
SIP.<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMso=
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class=3DMsoNormal>Thanks<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:=
p></p><p class=3DMsoNormal>Regards<o:p></o:p></p><p class=3DMsoNormal>Ranji=
t<o:p></o:p></p><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div></body></ht=
ml>=

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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Mapping between SIP and ROAP/JSEP
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Ranjit,

I agree with you that Interworking document for JSEP to SIP is important fo=
r developing the interworking solution. I'll interested in contributing one=
 such draft. Such a draft will help in identify the gap in JSEP to support =
SIP interworking like UPDATE during early dialog.

Thanks
Partha

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Avasarala, Ranjit
Sent: Wednesday, May 09, 2012 12:19 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Mapping between SIP and ROAP/JSEP

Hi

Is there any interest in the coming up with a mapping between SIP and ROAP/=
JSEP protocols?  Though JSEP is a set of APIs as of now which map closely w=
ith ROAP APIs, ROAP has messages like OFFER,ANSWER and OK which can be mapp=
ed to SIP.

Comment?

Thanks

Regards
Ranjit


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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ranjit,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I agree with you that =
Interworking document for JSEP to SIP is important for developing the inter=
working solution. I&#8217;ll interested in contributing one such draft. Suc=
h a draft will help in identify the gap in
 JSEP to support SIP interworking like UPDATE during early dialog.<o:p></o:=
p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Partha<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
4.0pt">
<div>
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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>Avasarala, Ranjit<br>
<b>Sent:</b> Wednesday, May 09, 2012 12:19 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Mapping between SIP and ROAP/JSEP<o:p></o:p></span=
></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Hi<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Is there any interest in the coming up with a mappin=
g between SIP and ROAP/JSEP protocols?&nbsp; Though JSEP is a set of APIs a=
s of now which map closely with ROAP APIs, ROAP has messages like OFFER,ANS=
WER and OK which can be mapped to SIP.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Comment?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Thanks<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards<o:p></o:p></p>
<p class=3D"MsoNormal">Ranjit<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
</div>
</div>
</body>
</html>

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From: Harald Alvestrand <harald@alvestrand.no>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
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On 05/08/2012 11:59 PM, Eric Rescorla wrote:
> On Tue, May 8, 2012 at 2:54 PM, Harald Alvestrand<harald@alvestrand.no>  wrote:
>> Just because I realized I didnt understand something, I ask.....
>>
>> We rejected RTCP RR as a consent freshness mechanism because RR is trivial
>> to fake.
>> But - now we have SRTP as mandatory-to-use, which means that all RTCP RRs
>> are integrity protected, origin authenticated and replay protected (do I
>> have that right?).
>>
>> What is the reason why this is not sufficient protection to use RTCP RR as a
>> consent freshness mechanism?
> This isn't a complete analysis, but if you are using SDES for key management,
> then the site knows the SRTCP keys, so I don't *think* SRTCP is buying you
> much. I haven't thought through this completely though, so maybe there is
> still some additional value.
Ah - had forgotten that the attacker is assumed to observe the 
signalling path. Thanks.


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Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP
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On 05/09/2012 08:52 AM, Ravindran, Parthasarathi wrote:
>
> Ranjit,
>
> I agree with you that Interworking document for JSEP to SIP is 
> important for developing the interworking solution. I?ll interested in 
> contributing one such draft. Such a draft will help in identify the 
> gap in JSEP to support SIP interworking like UPDATE during early dialog.
>
Since JSEP is an API, and has been implemented, I suggest that the most 
reasonable form of documentation is an opensource SIP implementation on 
top of JSEP, not an internet-draft.

If someone wants to contribute such code, I'm sure we can host it as 
part of webrtc-samples, if that helps - the other piece would be a 
recipe for setting up some opensource SIP server to prove that it 
actually works.

Running code has the advantage that it is easy to test it.


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    On 05/09/2012 08:52 AM, Ravindran, Parthasarathi wrote:
    <blockquote
cite="mid:387F9047F55E8C42850AD6B3A7A03C6C16033818@inba-mail01.sonusnet.com"
      type="cite">
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        <p class="MsoNormal"><span style="color: rgb(31, 73, 125);">Ranjit,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="color: rgb(31, 73, 125);"><o:p> </o:p></span></p>
        <p class="MsoNormal"><span style="color: rgb(31, 73, 125);">I
            agree with you that Interworking document for JSEP to SIP is
            important for developing the interworking solution. I&#8217;ll
            interested in contributing one such draft. Such a draft will
            help in identify the gap in JSEP to support SIP interworking
            like UPDATE during early dialog.</span></p>
      </div>
    </blockquote>
    Since JSEP is an API, and has been implemented, I suggest that the
    most reasonable form of documentation is an opensource SIP
    implementation on top of JSEP, not an internet-draft.<br>
    <br>
    If someone wants to contribute such code, I'm sure we can host it as
    part of webrtc-samples, if that helps - the other piece would be a
    recipe for setting up some opensource SIP server to prove that it
    actually works.<br>
    <br>
    Running code has the advantage that it is easy to test it.<br>
    <br>
  </body>
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From pravindran@sonusnet.com  Wed May  9 00:15:26 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Mapping between SIP and ROAP/JSEP
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Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP
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Harald,

My guess is that running code already exists in the form of SIP over websoc=
ket. The point is not about the running code. The running code will not cov=
er all the offer/answer state machine issues which I'm talking about. The r=
unning code is just equivalent to proof of concept but will not cover all t=
he deployable solution and does not cover all the existing offer/answer .

Even though JSEP is projected as API, offer/pr-answer/answer state MUST be =
traversed in some magic way between two web-browsers for establishing the s=
ession and these states are not possible to directly map with RFC 3264 or R=
FC 3261 (SIP) usage. So, there is a need of interworking document. The poin=
t to be noted is that the interworking between JSEP to SIP shall occur at b=
rowser or at WebRTC server.

Thanks
Partha

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Wednesday, May 09, 2012 12:31 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP

On 05/09/2012 08:52 AM, Ravindran, Parthasarathi wrote:
Ranjit,

I agree with you that Interworking document for JSEP to SIP is important fo=
r developing the interworking solution. I'll interested in contributing one=
 such draft. Such a draft will help in identify the gap in JSEP to support =
SIP interworking like UPDATE during early dialog.
Since JSEP is an API, and has been implemented, I suggest that the most rea=
sonable form of documentation is an opensource SIP implementation on top of=
 JSEP, not an internet-draft.

If someone wants to contribute such code, I'm sure we can host it as part o=
f webrtc-samples, if that helps - the other piece would be a recipe for set=
ting up some opensource SIP server to prove that it actually works.

Running code has the advantage that it is easy to test it.

--_000_387F9047F55E8C42850AD6B3A7A03C6C16033869inbamail01sonus_
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<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Harald,<o:p></o:p></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">My guess is that runni=
ng code already exists in the form of SIP over websocket. The point is not =
about the running code. The running code will not cover all the offer/answe=
r state machine issues which I&#8217;m talking
 about. The running code is just equivalent to proof of concept but will no=
t cover all the deployable solution and does not cover all the existing off=
er/answer .
<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Even though JSEP is pr=
ojected as API, offer/pr-answer/answer state MUST be traversed in some magi=
c way between two web-browsers for establishing the session and these state=
s are not possible to directly map with
 RFC 3264 or RFC 3261 (SIP) usage. So, there is a need of interworking docu=
ment. The point to be noted is that the interworking between JSEP to SIP sh=
all occur at browser or at WebRTC server.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Thanks<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Partha<o:p></o:p></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D"><o:p>&nbsp;</o:p></spa=
n></p>
<div style=3D"border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in =
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<div>
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<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;;color:windowtext">From:</span></b><spa=
n style=3D"font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif=
&quot;;color:windowtext"> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ie=
tf.org]
<b>On Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> Wednesday, May 09, 2012 12:31 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Mapping between SIP and ROAP/JSEP<o:p></o:p></=
span></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">On 05/09/2012 08:52 AM, Ravindran, Parthasarathi wro=
te: <o:p>
</o:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">Ranjit,</span><o:p></o=
:p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">&nbsp;</span><o:p></o:=
p></p>
<p class=3D"MsoNormal"><span style=3D"color:#1F497D">I agree with you that =
Interworking document for JSEP to SIP is important for developing the inter=
working solution. I&#8217;ll interested in contributing one such draft. Suc=
h a draft will help in identify the gap in
 JSEP to support SIP interworking like UPDATE during early dialog.</span><o=
:p></o:p></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:12.0pt;font-family:&quot;Times New Roman&quot;,&quot;serif&quot;">Since=
 JSEP is an API, and has been implemented, I suggest that the most reasonab=
le form of documentation is an opensource SIP implementation
 on top of JSEP, not an internet-draft.<br>
<br>
If someone wants to contribute such code, I'm sure we can host it as part o=
f webrtc-samples, if that helps - the other piece would be a recipe for set=
ting up some opensource SIP server to prove that it actually works.<br>
<br>
Running code has the advantage that it is easy to test it.<o:p></o:p></span=
></p>
</div>
</div>
</body>
</html>

--_000_387F9047F55E8C42850AD6B3A7A03C6C16033869inbamail01sonus_--

From magnus.westerlund@ericsson.com  Wed May  9 00:31:17 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
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On 2012-05-08 23:54, Harald Alvestrand wrote:
> Just because I realized I didnt understand something, I ask.....
> 
> We rejected RTCP RR as a consent freshness mechanism because RR is 
> trivial to fake.
> But - now we have SRTP as mandatory-to-use, which means that all RTCP 
> RRs are integrity protected, origin authenticated and replay protected 
> (do I have that right?).
> 
> What is the reason why this is not sufficient protection to use RTCP RR 
> as a consent freshness mechanism?
> 

I agree that SRTP will not really buy you much. Even using DTLS-SRTP
there are attack vectors for the case where the attacker is willing to
start in the attacked network and then move his signalling and crypto
state to an off-path source address spoofing consent freshness
generating bot. Thus leaving the stream alive for a long time.

The issue with RTCP is the amount of entropy in each RTCP packets that
are hard to guess given an off-path attacker and which the media sender
verifying consent can determine correctly.

Revisiting the discussion from September last year on this topic.
Especially my estimate of entropy sources in RTCP that can't be guessed:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01327.html

The only two fields that an off-path attacker can't easily spoof in an
RTCP RR are:
- last SR (LSR)
- Extended highest sequence number

And here the first one is the one that is more valuable as it is the
senders RTCP SR NTP timestamp field. Yes, it will increase as time goes
one, but the lower bits of this field when the actual RTCP SR was sent
is difficult to spoof correctly.

The sequence number is also a bit difficult to spoof if you don't see
the packets being sent. However, as this is not a field copied directly
from the latest received RTCP SR packet and thus need to be tolerant to
packet losses etc and thus accept a window of the from the latest
actually sent any sent the last few seconds.

I do think we can have consent freshness based on that three RTCP
packets in a time window of 30 seconds or some number of reporting
intervals that do not contain a clear error an actual receiver wouldn't
do, would keep the media stream alive. I would probably also add in
cases I get 3 RTCP packets in a row that matches security but contains
errors would directly kill the stream. Here actually SRTCP does help.
Because you can use SRTCP to ensure that the denial of service vector on
this second rule can't be triggered by someone outside of the security
context.

However, the question about consent freshness I think needs to consider
also if this RTCP rule is in fact reasonable for legacy interworking
cases. For a gateway responding to STUN messages are probably easier
than in addition to being able to do ICE initially also need a full RTP
stack to generate RTCP reports for any legacy end-point that doesn't
implement RTCP. This may however be a mote point depending on what
requirements on congestion control we put on the legacy interworking
gateway.

My personal perspective is that we actually need to consider all the
aspects and see if in the end there is any advantage of either using
RTCP or only using STUN.

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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Thread-Topic: [rtcweb] Consent freshness - revisiting the RTCP option
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Harald Alvestrand" <harald@alvestrand.no>, "Eric Rescorla" <ekr@rtfm.com>
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Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
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More reasons why RTCP may not be suitable for consent freshness:
1. RTCP (as described in RFC3550) is receiver based, so the browser
can't explicitly request for consent. If consent freshness fails (for
e.g, RTCP packets temporarily lost because of network flapping), the
browser would have to wait for the peer to send RTCP before it can start
transmitting media. Worst, if the peer isn't an active sender it may not
send any RTCP RR until it receives media. It could send a bare minimum
RTCP RR (an RR with RC=3D0 and SDES with CNAME), but that has zero
entropy:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg01497.html

2. There are still some endpoints that don't send / pay attention to
RTCP:
http://www.ietf.org/mail-archive/web/rai/current/msg01257.html

For these cases an intermediary like an SBC would have to manufacture
them. Manufacturing them for just consent freshness would be expensive
compared to generating STUN request/response.

Muthu

|-----Original Message-----
|From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Harald Alvestrand
|Sent: Wednesday, May 09, 2012 12:26 PM
|To: Eric Rescorla
|Cc: rtcweb@ietf.org
|Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
|
|On 05/08/2012 11:59 PM, Eric Rescorla wrote:
|> On Tue, May 8, 2012 at 2:54 PM, Harald
Alvestrand<harald@alvestrand.no>  wrote:
|>> Just because I realized I didnt understand something, I ask.....
|>>
|>> We rejected RTCP RR as a consent freshness mechanism because RR is
trivial
|>> to fake.
|>> But - now we have SRTP as mandatory-to-use, which means that all
RTCP RRs
|>> are integrity protected, origin authenticated and replay protected
(do I
|>> have that right?).
|>>
|>> What is the reason why this is not sufficient protection to use RTCP
RR as a
|>> consent freshness mechanism?
|> This isn't a complete analysis, but if you are using SDES for key
management,
|> then the site knows the SRTCP keys, so I don't *think* SRTCP is
buying you
|> much. I haven't thought through this completely though, so maybe
there is
|> still some additional value.
|Ah - had forgotten that the attacker is assumed to observe the
|signalling path. Thanks.
|
|_______________________________________________
|rtcweb mailing list
|rtcweb@ietf.org
|https://www.ietf.org/mailman/listinfo/rtcweb

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<div class=3D"WordSection1">
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">I think it is useful to have such a draft which provides a mappin=
g between SIP and ROAP/JSEP protocols.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">Previously Cullen has an expired draft related to this:<o:p></o:p=
></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><a href=3D"http://tools.ietf.org/html/draft-jennings-rtcweb-signa=
ling-gateway-00">http://tools.ietf.org/html/draft-jennings-rtcweb-signaling=
-gateway-00</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">Some time ago, I submitted a draft to map ROAP with XMPP/Jingle.<=
o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><a href=3D"http://datatracker.ietf.org/doc/draft-li-rtcweb-roap-j=
ingle-interworking/">http://datatracker.ietf.org/doc/draft-li-rtcweb-roap-j=
ingle-interworking/</a><o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">I need to update it to reflect the JSEP concept.<o:p></o:p></span=
></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">I would appreciate if anybody has interest to work with me about =
this.<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">Thanks,<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">Kind Regards<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1F497D">Kepeng<o: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=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:=CB=
=CE=CC=E5">=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:=CB=CE=CC=E5"> rtcweb-=
bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
</span><b><span style=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5">=B4=FA=
=B1=ED </span></b><span lang=3D"EN-US" style=3D"font-size:10.0pt;font-famil=
y:=CB=CE=CC=E5">Avasarala, Ranjit<br>
</span><b><span style=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5">=B7=A2=
=CB=CD=CA=B1=BC=E4<span lang=3D"EN-US">:</span></span></b><span lang=3D"EN-=
US" style=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5"> 2012</span><span s=
tyle=3D"font-size:10.0pt;font-family:=CB=CE=CC=E5">=C4=EA<span lang=3D"EN-U=
S">5</span>=D4=C2<span lang=3D"EN-US">9</span>=C8=D5<span lang=3D"EN-US">
 14:49<br>
</span><b>=CA=D5=BC=FE=C8=CB<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"=
> [rtcweb] Mapping between SIP and ROAP/JSEP<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">Hi<o:p></o:p></span></p>
<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">Is there any interest in the co=
ming up with a mapping between SIP and ROAP/JSEP protocols?&nbsp; Though JS=
EP is a set of APIs as of now which map closely with ROAP APIs, ROAP has me=
ssages like OFFER,ANSWER and OK which can
 be mapped to SIP.<o:p></o:p></span></p>
<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">Comment?<o:p></o:p></span></p>
<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">Thanks<o:p></o:p></span></p>
<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">Regards<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Ranjit<o:p></o:p></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><o:p>&nbsp;</o:p></span></p>
</div>
</body>
</html>

--_000_34966E97BE8AD64EAE9D3D6E4DEE36F2035816DEszxeml525mbschi_--

From harald@alvestrand.no  Wed May  9 02:59:51 2012
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From: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Consent freshness - revisiting the RTCP option
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On 05/09/2012 09:34 AM, Muthu Arul Mozhi Perumal (mperumal) wrote:
> More reasons why RTCP may not be suitable for consent freshness:
> 1. RTCP (as described in RFC3550) is receiver based, so the browser
> can't explicitly request for consent. If consent freshness fails (for
> e.g, RTCP packets temporarily lost because of network flapping), the
> browser would have to wait for the peer to send RTCP before it can start
> transmitting media. Worst, if the peer isn't an active sender it may not
> send any RTCP RR until it receives media. It could send a bare minimum
> RTCP RR (an RR with RC=0 and SDES with CNAME), but that has zero
> entropy:
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg01497.html
Of course endpoints are expected to be RFC 3550 compliant, which 
includes obeying the diktat of section 6.3.5: "Every SSRC that hasn't 
sent an RTP or RTCP packet for 5*5 seconds is dead".
> 2. There are still some endpoints that don't send / pay attention to
> RTCP:
> http://www.ietf.org/mail-archive/web/rai/current/msg01257.html
>
> For these cases an intermediary like an SBC would have to manufacture
> them. Manufacturing them for just consent freshness would be expensive
> compared to generating STUN request/response.

For non-RTCP entities, we need an RTCP-generating gateway in order to 
interwork.
End of story.
(My opinion.)



From ted.ietf@gmail.com  Wed May  9 08:52:17 2012
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Subject: [rtcweb] Strawman agenda and documents deadline for interim
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Howdy,

We would like document updates for the interim to be complete by June
4th, so that folks have time to review them before the interim
meeting.  Slides associated with the documents should be in by June
9th.  Below is the current strawman agenda; please comment.

regards,

Ted, Cullen, Magnus

Day 1

- Use Cases (2 H)
* New use cases needing additional discussion if not resolved on
mailing list.
* Use cases that require cloning (yes or no?)


- WebRTC's RTP Usage (3 H)
* Topologies that need to be supported
- what can be at one end of peer connection and what can be at other end
* Walk through of all the pieces suggested
- how to map media streams to rtp session.
 -- example: describe how retransmission SSRCs relate to main SSRCs
- mapping between RTP and what happens in JS

- JSEP
* Discussion of State Machine
* Interactions allowed
* SDP specification
 -- mapping between SDP and what happens in JS

- Security
* Continued Consent
* Identity and IdP
* Details of Security Architecture

From juberti@google.com  Wed May  9 19:54:16 2012
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--e89a8f2357c733e8e604bfa5bc47
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I would be willing to review a JSEP-XMPP draft.

On Wed, May 9, 2012 at 4:18 AM, Likepeng <likepeng@huawei.com> wrote:

>  I think it is useful to have such a draft which provides a mapping
> between SIP and ROAP/JSEP protocols.****
>
> ** **
>
> Previously Cullen has an expired draft related to this:****
>
> http://tools.ietf.org/html/draft-jennings-rtcweb-signaling-gateway-00****
>
> ** **
>
> Some time ago, I submitted a draft to map ROAP with XMPP/Jingle.****
>
> http://datatracker.ietf.org/doc/draft-li-rtcweb-roap-jingle-interworking/=
*
> ***
>
> ** **
>
> I need to update it to reflect the JSEP concept.****
>
> ** **
>
> I would appreciate if anybody has interest to work with me about this.***=
*
>
> ** **
>
> Thanks,****
>
> Kind Regards****
>
> Kepeng****
>
> *=E5=8F=91=E4=BB=B6=E4=BA=BA:* rtcweb-bounces@ietf.org [mailto:rtcweb-bou=
nces@ietf.org] *=E4=BB=A3=E8=A1=A8 *Avasarala,
> Ranjit
> *=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4:* 2012=E5=B9=B45=E6=9C=889=E6=97=A5=
 14:49
> *=E6=94=B6=E4=BB=B6=E4=BA=BA:* rtcweb@ietf.org
> *=E4=B8=BB=E9=A2=98:* [rtcweb] Mapping between SIP and ROAP/JSEP****
>
> ** **
>
> Hi****
>
> ** **
>
> Is there any interest in the coming up with a mapping between SIP and
> ROAP/JSEP protocols?  Though JSEP is a set of APIs as of now which map
> closely with ROAP APIs, ROAP has messages like OFFER,ANSWER and OK which
> can be mapped to SIP.****
>
> ** **
>
> Comment?****
>
> ** **
>
> Thanks****
>
> ** **
>
> Regards****
>
> Ranjit****
>
> ** **
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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I would be willing to review a JSEP-XMPP draft.<br><br><div class=3D"gmail_=
quote">On Wed, May 9, 2012 at 4:18 AM, Likepeng <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:likepeng@huawei.com" target=3D"_blank">likepeng@huawei.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 lang=3D"ZH-CN" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">I think it is useful to have such a draft which provides a mappin=
g between SIP and ROAP/JSEP protocols.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">Previously Cullen has an expired draft related to this:<u></u><u>=
</u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><a href=3D"http://tools.ietf.org/html/draft-jennings-rtcweb-signa=
ling-gateway-00" target=3D"_blank">http://tools.ietf.org/html/draft-jenning=
s-rtcweb-signaling-gateway-00</a><u></u><u></u></span></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">Some time ago, I submitted a draft to map ROAP with XMPP/Jingle.<=
u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><a href=3D"http://datatracker.ietf.org/doc/draft-li-rtcweb-roap-j=
ingle-interworking/" target=3D"_blank">http://datatracker.ietf.org/doc/draf=
t-li-rtcweb-roap-jingle-interworking/</a><u></u><u></u></span></p>


<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">I need to update it to reflect the JSEP concept.<u></u><u></u></s=
pan></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">I would appreciate if anybody has interest to work with me about =
this.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">Thanks,<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">Kind Regards<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US" style=3D"font-size:10.5pt;color=
:#1f497d">Kepeng<u></u><u></u></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>=E5=8F=91=E4=BB=B6=E4=BA=BA<span lang=3D"EN=
-US">:</span></span></b><span lang=3D"EN-US"> <a href=3D"mailto:rtcweb-boun=
ces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a> [mailto:<a href=
=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.o=
rg</a>]
</span><b><span>=E4=BB=A3=E8=A1=A8 </span></b><span lang=3D"EN-US">Avasaral=
a, Ranjit<br>
</span><b><span>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span lang=3D"EN-US">:<=
/span></span></b><span lang=3D"EN-US"> 2012</span><span>=E5=B9=B4<span lang=
=3D"EN-US">5</span>=E6=9C=88<span lang=3D"EN-US">9</span>=E6=97=A5<span lan=
g=3D"EN-US">
 14:49<br>
</span><b>=E6=94=B6=E4=BB=B6=E4=BA=BA<span lang=3D"EN-US">:</span></b><span=
 lang=3D"EN-US"> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcwe=
b@ietf.org</a><br>
</span><b>=E4=B8=BB=E9=A2=98<span lang=3D"EN-US">:</span></b><span lang=3D"=
EN-US"> [rtcweb] Mapping between SIP and ROAP/JSEP<u></u><u></u></span></sp=
an></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Hi<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Is there any interest in the co=
ming up with a mapping between SIP and ROAP/JSEP protocols?=C2=A0 Though JS=
EP is a set of APIs as of now which map closely with ROAP APIs, ROAP has me=
ssages like OFFER,ANSWER and OK which can
 be mapped to SIP.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Comment?<u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Thanks<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Regards<u></u><u></u></span></p=
>
<p class=3D"MsoNormal"><span lang=3D"EN-US">Ranjit<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span lang=3D"EN-US"><u></u>=C2=A0<u></u></span></p>
</div></div></div>
</div>

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

--e89a8f2357c733e8e604bfa5bc47--

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Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP
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On 5/9/12 8:53 PM, Justin Uberti wrote:
> I would be willing to review a JSEP-XMPP draft.

Same here.

Peter


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On Wed, May 9, 2012 at 2:24 AM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

>  I agree with Richard & Roman. =E2=80=9CFull=E2=80=9D re-OFFER MUST suppo=
rted required
> for couple of mid session scenario like ****
>
> ** **
>
> **1)      **The session is established with audio and then video shall be
> added in the middle in case complete offer is provided in re-OFFER.
>
You don't need a complete offer here, just a "full" offer for the video m=
=3D
section. This should already work today.

> ****
>
> **2)      ** It is possible to change the codec in the middle of the
> Session as Roman mentioned.
>
This also doesn't require a complete offer. The application would need to
remember what codecs are available so that it could change the offered
codec to something other than what it is currently using, but it would not
need to resend all codecs.

I agree Roman's case of a B2BUA seems like it would need a full offer,
although the idea of connecting an existing call to a new destination seems
like an uncommon case; the application could simply start a new call (i.e.
create a new PeerConnection), right?

If we decide we need to support this, the simplest way of handling this
would be to simply extend the |hints| parameter in createOffer to have some
sort of "complete" flag.


> ****
>
> ** **
>
> ** **
>
> ** **
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Roman Shpount
> *Sent:* Tuesday, May 08, 2012 9:03 PM
> *To:* Justin Uberti
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Question about JSEP createOffer****
>
> ** **
>
>
> ****
>
> On Tue, May 8, 2012 at 10:46 AM, Justin Uberti <juberti@google.com> wrote=
:
> ****
>
> Richard,****
>
> ** **
>
> That is an interesting point. Could you give a short example of when such
> a "full" re-OFFER would be used? There are a few ways we could handle thi=
s
> case.****
>
> ** **
>
>
> Actually I was the person who argued that "full" capabilities should be
> present in original SIP O/A, when new offer is generated in an existing
> session. The use case for this are B2BUA that connect an existing session
> to a new call or another existing session. Typically, B2BUA would ask one
> of the call parties for the offer (send an INVITE with no body in SIP), g=
et
> the new offer, and send this offer to the newly placed call. In order for
> this to work, all the calling party capabilities should be listed in this
> new offer, in particular all the codecs and all the communications types
> (audio, video, text, etc), or call would not be successfully established.
> _____________
> Roman Shpount****
>

--e89a8f2347b54a00e904bfa5da3b
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<br><br><div class=3D"gmail_quote">On Wed, May 9, 2012 at 2:24 AM, Ravindra=
n, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusne=
t.com" target=3D"_blank">pravindran@sonusnet.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px=
 #ccc solid;padding-left:1ex">







<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I agree with Richard &amp=
; Roman. =E2=80=9CFull=E2=80=9D re-OFFER MUST supported required for couple=
 of mid session scenario like
<u></u><u></u></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"><u></u>=C2=A0<u></u></spa=
n></p>
<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>1)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">The session is estab=
lished with audio and then video shall be added in the middle in case compl=
ete offer is provided in re-OFFER.</span></p>

</div></div></blockquote><div>You don&#39;t need a complete offer here, jus=
t a &quot;full&quot; offer for the video m=3D section. This should already =
work today.=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple"><div><p><span style=3D"f=
ont-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;colo=
r:#1f497d"><u></u><u></u></span></p>
<p><u></u><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&=
quot;sans-serif&quot;;color:#1f497d"><span>2)<span style=3D"font:7.0pt &quo=
t;Times New Roman&quot;">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span></span></span><u></u><span style=3D"font-size:11.0pt;font-family:&qu=
ot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0It is possible=
 to change the codec in the middle of the Session as Roman mentioned.</span=
></p>

</div></div></blockquote><div>This also doesn&#39;t require a complete offe=
r. The application would need to remember what codecs are available so that=
 it could change the offered codec to something other than what it is curre=
ntly using, but it would not need to resend all codecs.=C2=A0</div>

<div><br></div><div>I agree Roman&#39;s case of a B2BUA seems like it would=
 need a full offer, although the idea of connecting an existing call to a n=
ew destination seems like an uncommon case; the application could simply st=
art a new call (i.e. create a new PeerConnection), right?</div>

<div><br></div><div>If we decide we need to support this, the simplest way =
of handling this would be to simply extend the |hints| parameter in createO=
ffer to have some sort of &quot;complete&quot; flag.</div><div>=C2=A0<br></=
div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><p><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&qu=
ot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u></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"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal" style=3D"margin-left:.25in"><span style=3D"font-size=
:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497=
d"><u></u>=C2=A0<u></u></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"><u></u>=C2=A0<u></u></spa=
n></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;"> <a href=
=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.o=
rg</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank"=
>rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> Tuesday, May 08, 2012 9:03 PM<br>
<b>To:</b> Justin Uberti<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Question about JSEP createOffer<u></u><u></u><=
/span></p>
</div>
</div><div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal"><br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Tue, May 8, 2012 at 10:46 AM, Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">Richard,<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">That is an interesting point. Could you give a short=
 example of when such a &quot;full&quot; re-OFFER would be used?=C2=A0There=
 are a few ways we could handle this case.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Actually I was the person who argued that &quot;full&quot; capabilities sho=
uld be present in original SIP O/A, when new offer is generated in an exist=
ing session. The use case for this are B2BUA that connect an existing sessi=
on to a new call or another existing session.
 Typically, B2BUA would ask one of the call parties for the offer (send an =
INVITE with no body in SIP), get the new offer, and send this offer to the =
newly placed call. In order for this to work, all the calling party capabil=
ities should be listed in this new
 offer, in particular all the codecs and all the communications types (audi=
o, video, text, etc), or call would not be successfully established.<br>
_____________<br>
Roman Shpount<u></u><u></u></p>
</div></div></div>
</div>
</div>

</blockquote></div><br>

--e89a8f2347b54a00e904bfa5da3b--

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Date: Thu, 10 May 2012 12:42:56 +0700
From: Glen Zorn <glenzorn@gmail.com>
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Subject: Re: [rtcweb] Mapping between SIP and ROAP/JSEP
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On 05/10/2012 10:00 AM, Peter Saint-Andre wrote:
> On 5/9/12 8:53 PM, Justin Uberti wrote:
>> I would be willing to review a JSEP-XMPP draft.
> Same here.

Me, too.

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


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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Question about JSEP createOffer
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From: Justin Uberti <juberti@google.com>
Date: Thu, 10 May 2012 09:48:37 -0400
Message-ID: <CAOJ7v-1J+GLRea=3r4Ees5SadQ60grogACEbofoZ0Jh5VrpY6g@mail.gmail.com>
To: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Question about JSEP createOffer
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On Thu, May 10, 2012 at 4:10 AM, Ravindran, Parthasarathi <
pravindran@sonusnet.com> wrote:

>  Justin,****
>
> ** **
>
> The new media type/codec change by the answerer requires the complete set
> of codec from offerer. It is not appropriate to remember the previously
> offered codec & capabilities as it is subject to vary in the mid-session.
> For example, it is possible for the offerer to remove webcam in the middl=
e
> of the audio session wherein answerer should not expect that video is
> supported by offerer as it was offered initially.****
>
> ** **
>
> Please read inline for more comments.****
>
> ** **
>
> Thanks****
>
> Partha****
>
> ** **
>
> *From:* Justin Uberti [mailto:juberti@google.com]
> *Sent:* Thursday, May 10, 2012 8:32 AM
> *To:* Ravindran, Parthasarathi
> *Cc:* Roman Shpount; rtcweb@ietf.org
>
> *Subject:* Re: [rtcweb] Question about JSEP createOffer****
>
>  ** **
>
> ** **
>
> On Wed, May 9, 2012 at 2:24 AM, Ravindran, Parthasarathi <
> pravindran@sonusnet.com> wrote:****
>
> I agree with Richard & Roman. =E2=80=9CFull=E2=80=9D re-OFFER MUST suppor=
ted required for
> couple of mid session scenario like ****
>
>  ****
>
> 1)      The session is established with audio and then video shall be
> added in the middle in case complete offer is provided in re-OFFER.****
>
> You don't need a complete offer here, just a "full" offer for the video m=
=3D
> section. This should already work today. ****
>
> <partha>  IIUC for audio session, each time re-offer MUST contain =E2=80=
=9Cfull=E2=80=9D
> offer for video section </partha>
>

If you have an audio session, and you add video, the offer to add video
will include a "full" offer for video. But changes just to the audio
session itself will obviously not include anything about video. Maybe I
don't understand what you are suggesting.

> ****
>
>  2)       It is possible to change the codec in the middle of the Session
> as Roman mentioned.****
>
>  This also doesn't require a complete offer. The application would need
> to remember what codecs are available so that it could change the offered
> codec to something other than what it is currently using, but it would no=
t
> need to resend all codecs. ****
>
> <partha> Your proposal is limiting as the answerer has no choice to the
> select the preferred codec </partha>
>

I thought the offerer wanted to change the codec. Why would the answerer
need to make a selection? If the 'answerer' wants to use a different codec,
it should make an offer of its own.

****
>
> ** **
>
> I agree Roman's case of a B2BUA seems like it would need a full offer,
> although the idea of connecting an existing call to a new destination see=
ms
> like an uncommon case; the application could simply start a new call (i.e=
.
> create a new PeerConnection), right?****
>
> ** **
>
> If we decide we need to support this, the simplest way of handling this
> would be to simply extend the |hints| parameter in createOffer to have so=
me
> sort of "complete" flag.****
>
> <partha> your API proposal looks good  as it is possible to re-use the
> existing or offer full based on the application needs. But the applicatio=
n
> developer  MUST aware of the implication as it calls for interop issue in
> case of federation scenario if the full offer is not sent across. </parth=
a>
>

The application developer only needs to do this if supporting these edge
cases is relevant to their application.

> ****
>
>  ****
>
>   ****
>
>  ****
>
>  ****
>
> *From:* rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *Roman Shpount
> *Sent:* Tuesday, May 08, 2012 9:03 PM
> *To:* Justin Uberti
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Question about JSEP createOffer****
>
>  ****
>
>
> ****
>
> On Tue, May 8, 2012 at 10:46 AM, Justin Uberti <juberti@google.com> wrote=
:
> ****
>
> Richard,****
>
>  ****
>
> That is an interesting point. Could you give a short example of when such
> a "full" re-OFFER would be used? There are a few ways we could handle thi=
s
> case.****
>
>  ****
>
>
> Actually I was the person who argued that "full" capabilities should be
> present in original SIP O/A, when new offer is generated in an existing
> session. The use case for this are B2BUA that connect an existing session
> to a new call or another existing session. Typically, B2BUA would ask one
> of the call parties for the offer (send an INVITE with no body in SIP), g=
et
> the new offer, and send this offer to the newly placed call. In order for
> this to work, all the calling party capabilities should be listed in this
> new offer, in particular all the codecs and all the communications types
> (audio, video, text, etc), or call would not be successfully established.
> _____________
> Roman Shpount****
>
>  ** **
>

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Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Thu, May 10, 2012 at 4:10 AM, Ravindr=
an, Parthasarathi <span dir=3D"ltr">&lt;<a href=3D"mailto:pravindran@sonusn=
et.com" target=3D"_blank">pravindran@sonusnet.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 lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Justin,<u></u><u></u></sp=
an></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></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">The new media type/codec =
change by the answerer requires the complete set of codec from offerer. It =
is not appropriate to remember the previously offered codec
 &amp; capabilities as it is subject to vary in the mid-session. For exampl=
e, it is possible for the offerer to remove webcam in the middle of the aud=
io session wherein answerer should not expect that video is supported by of=
ferer as it was offered initially.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">Please read inline for mo=
re comments.<u></u><u></u></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"><u></u>=C2=A0<u></u></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">Thanks<u></u><u></u></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">Partha<u></u><u></u></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"><u></u>=C2=A0<u></u></spa=
n></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;"> Justin U=
berti [mailto:<a href=3D"mailto:juberti@google.com" target=3D"_blank">juber=
ti@google.com</a>]
<br>
<b>Sent:</b> Thursday, May 10, 2012 8:32 AM<br>
<b>To:</b> Ravindran, Parthasarathi<br>
<b>Cc:</b> Roman Shpount; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_bla=
nk">rtcweb@ietf.org</a></span></p><div class=3D"im"><br>
<b>Subject:</b> Re: [rtcweb] Question about JSEP createOffer<u></u><u></u><=
/div><p></p>
</div>
</div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><u></u>=C2=A0<u></u><=
/p>
<div>
<p class=3D"MsoNormal">On Wed, May 9, 2012 at 2:24 AM, Ravindran, Parthasar=
athi &lt;<a href=3D"mailto:pravindran@sonusnet.com" target=3D"_blank">pravi=
ndran@sonusnet.com</a>&gt; wrote:<u></u><u></u></p><div class=3D"im">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">I agree with Richard &amp=
; Roman. =E2=80=9CFull=E2=80=9D re-OFFER MUST supported required for couple=
 of mid session scenario
 like </span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>
<p><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;;color:#1f497d">1)</span><span style=3D"font-size:7.0pt;color=
:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d">The session is established with audio and=
 then video shall be added in the middle in case complete offer is provided=
 in re-OFFER.</span><u></u><u></u></p>


</div>
</div>
</div><div><div class=3D"im">
<p class=3D"MsoNormal">You don&#39;t need a complete offer here, just a &qu=
ot;full&quot; offer for the video m=3D section. This should already work to=
day.=C2=A0<u></u><u></u></p>
</div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;partha&gt;=C2=
=A0 IIUC for audio session, each time re-offer MUST contain =E2=80=9Cfull=
=E2=80=9D offer for video section &lt;/partha&gt;</span></p>

</div></div></div></div></div></blockquote><div><br></div><div>If you have =
an audio session, and you add video, the offer to add video will include a =
&quot;full&quot; offer for video. But changes just to the audio session its=
elf will obviously not include anything about video. Maybe I don&#39;t unde=
rstand what you are suggesting.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:=
0in 0in 0in 4.0pt">

<div><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-famil=
y:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u><=
/span></p>
</div><div class=3D"im">
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sa=
ns-serif&quot;;color:#1f497d">2)</span><span style=3D"font-size:7.0pt;color=
:#1f497d">=C2=A0=C2=A0=C2=A0=C2=A0=C2=A0
</span><span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quo=
t;sans-serif&quot;;color:#1f497d">=C2=A0It is possible to change the codec =
in the middle of the Session as Roman mentioned.</span><u></u><u></u></p>
</div>
</div>
</blockquote>
</div><div><div class=3D"im">
<p class=3D"MsoNormal">This also doesn&#39;t require a complete offer. The =
application would need to remember what codecs are available so that it cou=
ld change the offered codec to something other than what it is currently us=
ing, but it would not need to resend all
 codecs.=C2=A0<u></u><u></u></p>
</div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;partha&gt; Your=
 proposal is limiting as the answerer has no choice to the select the prefe=
rred codec &lt;/partha&gt;</span></p>

</div></div></div></div></div></blockquote><div><br></div><div>I thought th=
e offerer wanted to change the codec. Why would the answerer need to make a=
 selection? If the &#39;answerer&#39; wants to use a different codec, it sh=
ould make an offer of its own.</div>

<div><br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex=
;border-left:1px #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"b=
lue" vlink=3D"purple"><div><div style=3D"border:none;border-left:solid blue=
 1.5pt;padding:0in 0in 0in 4.0pt">

<div><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-famil=
y:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u><=
/span></p>
</div><div class=3D"im">
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I agree Roman&#39;s case of a B2BUA seems like it wo=
uld need a full offer, although the idea of connecting an existing call to =
a new destination seems like an uncommon case; the application could simply=
 start a new call (i.e. create a new PeerConnection),
 right?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div><div><div class=3D"im">
<p class=3D"MsoNormal">If we decide we need to support this, the simplest w=
ay of handling this would be to simply extend the |hints| parameter in crea=
teOffer to have some sort of &quot;complete&quot; flag.<u></u><u></u></p>


</div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;partha&gt; your=
 API proposal looks good=C2=A0 as it is possible to re-use the existing or =
offer full based on the application needs. But the application developer=C2=
=A0
 MUST aware of the implication as it calls for interop issue in case of fed=
eration scenario if the full offer is not sent across. &lt;/partha&gt;</spa=
n></p></div></div></div></div></div></blockquote><div><br></div><div>The ap=
plication developer only needs to do this if supporting these edge cases is=
 relevant to their application.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div lang=3D"EN-US" link=3D"blue" vlink=3D"p=
urple"><div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:=
0in 0in 0in 4.0pt">

<div><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-famil=
y:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u><=
/span></p>
</div><div class=3D"im">
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<blockquote style=3D"border:none;border-left:solid #cccccc 1.0pt;padding:0i=
n 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in">
<div>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></p>
<p class=3D"MsoNormal" style=3D"margin-left:.25in">
<span style=3D"font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-=
serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></u></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">=C2=A0</span><u></u><u></=
u></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;">
<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces=
@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"=
_blank">rtcweb-bounces@ietf.org</a>]
<b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> Tuesday, May 08, 2012 9:03 PM<br>
<b>To:</b> Justin Uberti<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf=
.org</a><br>
<b>Subject:</b> Re: [rtcweb] Question about JSEP createOffer</span><u></u><=
u></u></p>
</div>
</div>
<div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
<p class=3D"MsoNormal"><br clear=3D"all">
<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">On Tue, May 8, 2012 at 10:46 AM, Justin Uberti &lt;<=
a href=3D"mailto:juberti@google.com" target=3D"_blank">juberti@google.com</=
a>&gt; wrote:<u></u><u></u></p>
<p class=3D"MsoNormal">Richard,<u></u><u></u></p>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">That is an interesting point. Could you give a short=
 example of when such a &quot;full&quot; re-OFFER would be used?=C2=A0There=
 are a few ways we could handle this case.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><br>
Actually I was the person who argued that &quot;full&quot; capabilities sho=
uld be present in original SIP O/A, when new offer is generated in an exist=
ing session. The use case for this are B2BUA that connect an existing sessi=
on to a new call or another existing session.
 Typically, B2BUA would ask one of the call parties for the offer (send an =
INVITE with no body in SIP), get the new offer, and send this offer to the =
newly placed call. In order for this to work, all the calling party capabil=
ities should be listed in this new
 offer, in particular all the codecs and all the communications types (audi=
o, video, text, etc), or call would not be successfully established.<br>
_____________<br>
Roman Shpount<u></u><u></u></p>
</div>
</div>
</div>
</div>
</div>
</blockquote>
</div></div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>
</div>

</blockquote></div><br>

--20cf300513f6a10a7104bfaee1c5--

From fluffy@iii.ca  Thu May 10 07:13:47 2012
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericss on.se> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com>
To: Roman Shpount <roman@telurix.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On May 3, 2012, at 2:34 PM, Roman Shpount wrote:

> On Thu, May 3, 2012 at 4:29 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> You could also choose not to alert the remote user until the ICE =
procedures have finished - more or less using ICE with preconditions.
>=20
> Of course, that is going to trigger actions in STUN/TURN servers, even =
if the called party won't accept the call, but at least from a user =
experience perspective that is still better than lifting the hook, and =
having to wait for whatever-time-it-takes-for-ICE-to-finish seconds =
before one can start to talk.
>=20
>=20
> This also has a downside of disclosing user's IP to the caller without =
the user confirmation. For a lot of applications this can be serious =
security flaw.=20
> _____________

The initial media flow could be forced to provide only relay ICE =
candidates to help mitigate this. (I think we decided awhile back that =
was one of the things we would provide support for in the API). Later =
with the user answers or consents in same way, the ICE can move the =
media to a non relay flow.=20





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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <BEC86C2A-B983-4230-A997-A39CF4205CCD@iii.ca>
Date: Thu, 10 May 2012 12:40:21 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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--047d7b33d550a34e5b04bfb146c5
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On Thu, May 10, 2012 at 10:13 AM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On May 3, 2012, at 2:34 PM, Roman Shpount wrote:
>
> > On Thu, May 3, 2012 at 4:29 PM, Christer Holmberg <
> christer.holmberg@ericsson.com> wrote:
> > You could also choose not to alert the remote user until the ICE
> procedures have finished - more or less using ICE with preconditions.
> >
> > Of course, that is going to trigger actions in STUN/TURN servers, even
> if the called party won't accept the call, but at least from a user
> experience perspective that is still better than lifting the hook, and
> having to wait for whatever-time-it-takes-for-ICE-to-finish seconds before
> one can start to talk.
> >
> >
> > This also has a downside of disclosing user's IP to the caller without
> the user confirmation. For a lot of applications this can be serious
> security flaw.
> > _____________
>
> The initial media flow could be forced to provide only relay ICE
> candidates to help mitigate this. (I think we decided awhile back that was
> one of the things we would provide support for in the API). Later with the
> user answers or consents in same way, the ICE can move the media to a non
> relay flow.
>
> Just to make this clear, in this scenario, once application receives the
offer, it will generate a provisional answer with only relay candidates,
send this answer to the calling party, completes the ICE process on the
called party side, and then alert the called party. Once called user
accepts the call, application will send the final answer with relay,
reflexive and local candidates. This can also be implemented with final
answer with only relay candidates by the called party before user accepts
the the call, and O/A in the opposite direction once the call is accepted.
There is still a slight chance of clipping the media from the calling party
towards the called party since the answer might not arrive to the calling
party or ICE process from the calling party to the called party might not
complete before the called party answers. To avoid this, application will
need to send a separate message via signaling or data channel once the ICE
process on the calling side completes and only alert the called party once
this signaling message is received. What it boils down to is that
application should setup the media channel (provide relay candidates only
from the called side, if IP privacy is a concern), send a separate
signaling message when it is done, and then alert the user. This will
significantly slow down the actual call setup and will use a bunch of extra
resources, but will solve the clipping problem.
_____________
Roman Shpount

--047d7b33d550a34e5b04bfb146c5
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br><div class=3D"gmail_quote">On Thu, May 10, 2012 at 10:13 AM, Cullen Jen=
nings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_bla=
nk">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote=
" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On May 3, 2012, at 2:34 PM, Roman Shpount wrote:<br>
<br>
&gt; On Thu, May 3, 2012 at 4:29 PM, Christer Holmberg &lt;<a href=3D"mailt=
o:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt; wr=
ote:<br>
&gt; You could also choose not to alert the remote user until the ICE proce=
dures have finished - more or less using ICE with preconditions.<br>
&gt;<br>
&gt; Of course, that is going to trigger actions in STUN/TURN servers, even=
 if the called party won&#39;t accept the call, but at least from a user ex=
perience perspective that is still better than lifting the hook, and having=
 to wait for whatever-time-it-takes-for-ICE-to-finish seconds before one ca=
n start to talk.<br>

&gt;<br>
&gt;<br>
&gt; This also has a downside of disclosing user&#39;s IP to the caller wit=
hout the user confirmation. For a lot of applications this can be serious s=
ecurity flaw.<br>
&gt; _____________<br>
<br>
</div></div>The initial media flow could be forced to provide only relay IC=
E candidates to help mitigate this. (I think we decided awhile back that wa=
s one of the things we would provide support for in the API). Later with th=
e user answers or consents in same way, the ICE can move the media to a non=
 relay flow.<br>

<br></blockquote><div>Just to make this clear, in this scenario, once appli=
cation receives the offer, it will generate a provisional answer with only =
relay candidates, send this answer to the calling party, completes the ICE =
process on the called party side, and then alert the called party. Once cal=
led user accepts the call, application will send the final answer with rela=
y, reflexive and local candidates. This can also be implemented with final =
answer with only relay candidates by the called party before user accepts t=
he the call, and O/A in the opposite direction once the call is accepted. T=
here is still a slight chance of clipping the media from the calling party =
towards the called party since the answer might not arrive to the calling p=
arty or ICE process from the calling party to the called party might not co=
mplete before the called party answers. To avoid this, application will nee=
d to send a separate message via signaling or data channel once the ICE pro=
cess on the calling side completes and only alert the called party once thi=
s signaling message is received. What it boils down to is that application =
should setup the media channel (provide relay candidates only from the call=
ed side, if IP privacy is a concern), send a separate signaling message whe=
n it is done, and then alert the user. This will significantly slow down th=
e actual call setup and will use a bunch of extra resources, but will solve=
 the clipping problem.<br>
</div><div>_____________<br>Roman Shpount<br>=A0
<br></div></div><br>

--047d7b33d550a34e5b04bfb146c5--

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Date: Thu, 10 May 2012 12:53:58 -0400
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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Question about JSEP createOffer
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--e89a8ff24e1153975a04bfb177fb
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On Thu, May 10, 2012 at 9:48 AM, Justin Uberti <juberti@google.com> wrote:

>
> ****
>>
>> ** **
>>
>> I agree Roman's case of a B2BUA seems like it would need a full offer,
>> although the idea of connecting an existing call to a new destination seems
>> like an uncommon case; the application could simply start a new call (i.e.
>> create a new PeerConnection), right?****
>>
>> ** **
>>
>> If we decide we need to support this, the simplest way of handling this
>> would be to simply extend the |hints| parameter in createOffer to have some
>> sort of "complete" flag.****
>>
>> <partha> your API proposal looks good  as it is possible to re-use the
>> existing or offer full based on the application needs. But the application
>> developer  MUST aware of the implication as it calls for interop issue in
>> case of federation scenario if the full offer is not sent across. </partha>
>>
>
> The application developer only needs to do this if supporting these edge
> cases is relevant to their application.
>
>> ****
>>
>>
If you inter-operate with anything SIP, you need to do a full new offer
every time you get a re-INVITE without a body. The reason for this is you
are working with an external system. Thus you cannot guess if this
re-INVITE was generated by an existing call or some sort of third party
call control application which will need a full offer. This is not
something that is specific to SIP only. Every time you will implement any
type of interconnect between disjoint systems you will deal with a similar
problem.
_____________
Roman Shpount

--e89a8ff24e1153975a04bfb177fb
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<br clear=3D"all"><div class=3D"gmail_quote">On Thu, May 10, 2012 at 9:48 A=
M, Justin Uberti <span dir=3D"ltr">&lt;<a href=3D"mailto:juberti@google.com=
" target=3D"_blank">juberti@google.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">
<div class=3D"gmail_quote"><div class=3D"im"><br><blockquote class=3D"gmail=
_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:=
1ex"><div link=3D"blue" vlink=3D"purple" lang=3D"EN-US"><div><div style=3D"=
border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt">


<div><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-famil=
y:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u><=
/span></p>
</div><div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">I agree Roman&#39;s case of a B2BUA seems like it wo=
uld need a full offer, although the idea of connecting an existing call to =
a new destination seems like an uncommon case; the application could simply=
 start a new call (i.e. create a new PeerConnection),
 right?<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=A0<u></u></p>
</div>
</div><div><div>
<p class=3D"MsoNormal">If we decide we need to support this, the simplest w=
ay of handling this would be to simply extend the |hints| parameter in crea=
teOffer to have some sort of &quot;complete&quot; flag.<u></u><u></u></p>



</div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&q=
uot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d">&lt;partha&gt; your=
 API proposal looks good=A0 as it is possible to re-use the existing or off=
er full based on the application needs. But the application developer=A0
 MUST aware of the implication as it calls for interop issue in case of fed=
eration scenario if the full offer is not sent across. &lt;/partha&gt;</spa=
n></p></div></div></div></div></div></blockquote><div><br></div></div>
<div>The application developer only needs to do this if supporting these ed=
ge cases is relevant to their application.</div><div class=3D"im">

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div link=3D"blue" vlink=3D"purple" lang=3D"=
EN-US"><div><div style=3D"border:none;border-left:solid blue 1.5pt;padding:=
0in 0in 0in 4.0pt">


<div><div><p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-famil=
y:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u><u></u><=
/span></p>
</div><br></div></div></div></div></blockquote></div></div></blockquote></d=
iv><br>If you inter-operate with anything SIP, you need to do a full new of=
fer every time you get a re-INVITE without a body. The reason for this is y=
ou are working with an external system. Thus you cannot guess if this re-IN=
VITE was generated by an existing call or some sort of third party call con=
trol application which will need a full offer. This is not something that i=
s specific to SIP only. Every time you will implement any type of interconn=
ect between disjoint systems you will deal with a similar problem.<br>
_____________<br>Roman Shpount<br>
<br>

--e89a8ff24e1153975a04bfb177fb--

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From: Martin Thomson <martin.thomson@gmail.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 10 May 2012 07:13, Cullen Jennings <fluffy@iii.ca> wrote:
> The initial media flow could be forced to provide only relay ICE candidat=
es to help mitigate this. (I think we decided awhile back that was one of t=
he things we would provide support for in the API). Later with the user ans=
wers or consents in same way, the ICE can move the media to a non relay flo=
w.

That's right.  But you pretty much have to leave this in the hands of
the application.

Unless you want to provide an authenticated indication of user answer
on the media path so that when the user picks up the phone you can
provide other local candidates...

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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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I'm a very confused on what type of forking (with ICE) use case people =
want to satisfy.=20

I agree with you that PRANSWER does make things more complicated but I =
presented at several of the meetings about how things just don't work =
without it or something like it so I don't see how to avoid it.=20


On May 9, 2012, at 12:34 AM, Ravindran, Parthasarathi wrote:

> Hi Justin,
> =20
> Sorry for the delay in reply. There are two issues associated with =
this scenario:
> =20
> Issue 1) SIP Forking : This is solved by peerconnection cloning
> =20
> Issue 2)  UPDATE support within SIP early dialog in the non-forking =
scenario. Here, the cloning will not solve issue as the answerer =
originated the new offer during SDP_PRANSWER state . Issue 2 is the =
title of this mail thread.
> =20
> IMO, SDP_PRANSWER will complicate the offer/answer state machine.
> =20
> Thanks
> Partha
> =20
> =20
> =20
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf Of Justin Uberti
> Sent: Tuesday, May 08, 2012 8:40 PM
> To: Christer Holmberg
> Cc: Randell Jesup; rtcweb@ietf.org
> Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in =
JSEP
> =20
> The conclusion on this thread seems to be that the right way to =
address this problem is via PeerConnection cloning, and that no changes =
are needed to the JSEP state machine. I'll work with Richard to figure =
out how we should extend JSEP to support cloning.
>=20
> On Thu, May 3, 2012 at 5:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> >>>>You could also choose not to alert the remote user until the ICE =
procedures have finished - more or less using ICE with preconditions.
> >>>>
> >>>> Of course, that is going to trigger actions in STUN/TURN servers, =
even if the called party won't accept the call, but at least from a user
> >>>> experience perspective that is still better than lifting the =
hook, and having to wait for whatever-time-it-takes-for-ICE-to-finish =
seconds before one can start to talk.
> >>>
> >>> This also has a downside of disclosing user's IP to the caller =
without the user confirmation. For a lot of applications this can be =
serious security flaw.
> >>
> >> The client can still log when ICE procedures occur.
> >>
> >> Because, even if I wait until your phone starts to ring, most =
likely I would still get your IP address without user confirmation =
(speaking in SIP terms, phones normally don't ask for user permission =
before they send 18x with SDP), eventhough you would easier be made =
aware of that it happens.
> >
> > Another alternative is of course to do ICE with an SBC, and/or =
having an SBC doing ICE on behalf of you :)
> >
> >
> > This is true for SIP but was as far as I know was specifically =
designed around in WebRTC. WebRTC signaling server acts as B2BUA so any =
type of ringing notification goes through the web server and does not =
need to carry any type of client address information. The client address =
information is only provided
> > when SDP answer is sent or ICE negotiation is started.
>=20
> Assuming you are only going to make user confirmation (read: accept =
calls) in cases where you absolutely sure that the caller isn't someone =
trying to get your IP address.
>=20
> But, never the less, having a solution where I first have to give user =
confirmation, and then wait until I can start to talk, is probably =
something most people want to avoid. Depending upon, of course, how long =
the waiting time is.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> 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: Cullen Jennings <fluffy@iii.ca>
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Date: Thu, 10 May 2012 13:06:32 -0600
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001337@ESESSCMS0356.eemea.ericss on.se> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001339@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxvFYy-9XAE5GG0n58xX2HKRX3o5qkTZ2o6rugakaM77Ew@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400133A@ESESSCMS0356.eemea.ericsson.se> <CAOJ7v-1Ue=q4MAh0csiDLvRFfMGaJ4vWrDxhir-S-OjSTsymaw@mail.gmail.com>
To: Justin Uberti <juberti@google.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Uh, I don't think we are ready to design anything yet - I am still 100% =
confused on what the use case we are trying to solve is. Is it forking =
on at the browser, in intermediate node, in SIP, parallel, sequential or =
what? I want to see the end user use cases we need to deal with. The =
example Randell gave seems reasonable but seem like it would best be =
handled with just multiple independent PeerConnection. The combination =
of requiring ICE and SIP parallel forking is very complicated - =
particularly without Reliable provisions responses and all the =
complexity that ensues.

So before we go down this path, lets get clearer on what we are trying =
to accomplish.

Cullen

On May 8, 2012, at 9:09 AM, Justin Uberti wrote:

> The conclusion on this thread seems to be that the right way to =
address this problem is via PeerConnection cloning, and that no changes =
are needed to the JSEP state machine. I'll work with Richard to figure =
out how we should extend JSEP to support cloning.
>=20
> On Thu, May 3, 2012 at 5:16 PM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:
> Hi,
>=20
> >>>>You could also choose not to alert the remote user until the ICE =
procedures have finished - more or less using ICE with preconditions.
> >>>>
> >>>> Of course, that is going to trigger actions in STUN/TURN servers, =
even if the called party won't accept the call, but at least from a user
> >>>> experience perspective that is still better than lifting the =
hook, and having to wait for whatever-time-it-takes-for-ICE-to-finish =
seconds before one can start to talk.
> >>>
> >>> This also has a downside of disclosing user's IP to the caller =
without the user confirmation. For a lot of applications this can be =
serious security flaw.
> >>
> >> The client can still log when ICE procedures occur.
> >>
> >> Because, even if I wait until your phone starts to ring, most =
likely I would still get your IP address without user confirmation =
(speaking in SIP terms, phones normally don't ask for user permission =
before they send 18x with SDP), eventhough you would easier be made =
aware of that it happens.
> >
> > Another alternative is of course to do ICE with an SBC, and/or =
having an SBC doing ICE on behalf of you :)
> >
> >
> > This is true for SIP but was as far as I know was specifically =
designed around in WebRTC. WebRTC signaling server acts as B2BUA so any =
type of ringing notification goes through the web server and does not =
need to carry any type of client address information. The client address =
information is only provided
> > when SDP answer is sent or ICE negotiation is started.
>=20
> Assuming you are only going to make user confirmation (read: accept =
calls) in cases where you absolutely sure that the caller isn't someone =
trying to get your IP address.
>=20
> But, never the less, having a solution where I first have to give user =
confirmation, and then wait until I can start to talk, is probably =
something most people want to avoid. Depending upon, of course, how long =
the waiting time is.
>=20
> Regards,
>=20
> Christer
> _______________________________________________
> 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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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <4FA2E264.4040207@jesup.org> <CAD5OKxs3AqvbookWYPDgOuAkjPZGVUdSEhqz6P_p9E4h_ZHJGQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001339@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxvFYy-9XAE5GG0n58xX2HKRX3o5qkTZ2o6rugakaM77Ew@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400133A@ESESSCMS0356.eemea.ericsson.se> <CAOJ7v-1Ue=q4MAh0csiDLvRFfMGaJ4vWrDxhir-S-OjSTsymaw@mail.gmail.com> <90DEBA9A-E8D2-4662-9A41-95B361FDAF05@iii.ca>
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From: Roman Shpount <roman@telurix.com>
To: Cullen Jennings <fluffy@iii.ca>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On Thu, May 10, 2012 at 3:06 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> Uh, I don't think we are ready to design anything yet - I am still 100%
> confused on what the use case we are trying to solve is. Is it forking on
> at the browser, in intermediate node, in SIP, parallel, sequential or what?
> I want to see the end user use cases we need to deal with. The example
> Randell gave seems reasonable but seem like it would best be handled with
> just multiple independent PeerConnection. The combination of requiring ICE
> and SIP parallel forking is very complicated - particularly without
> Reliable provisions responses and all the complexity that ensues.
>
> So before we go down this path, lets get clearer on what we are trying to
> accomplish.
>
>
One use case is interop with SIP with parallel forking, reliable
provisioning responses, UPDATE, and ICE. The big question regarding this
use case is availability of anything that supports parallel forking,
reliable provisioning and UPDATE. I actually think ICE makes implementation
of this use case easier, not harder.

The real end user use case is any type of service where calling destination
is a small group of people. This allows to initiate a single offer and send
it to all the members of the group. In most cases only one person will
reply, but, in a smaller set of cases, several people from the group will
answer. In the later case, you need a mechanism to clone the original peer
connection (or to create additional peer connections based on the original
offer) and deal with additional answers independently.

This is a very common use case for any type of service where people use
multiple devices. For instance if we add voice/video calling to a web mail
program, I can be logged in from multiple devices. It can also be a shared
account (like support@acme.com) where several different people are logged
in to the same account. When someone places a call to a shared account,
several people can answer at the same time.

I know this can be addressed by sending a separate message  and reversing
the call setup direction (each accepting party creates an offer and sends
it back), but this causes another set of issues related to slow call setup
and media clipping.
_____________
Roman Shpount

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<br><div class=3D"gmail_quote">On Thu, May 10, 2012 at 3:06 PM, Cullen Jenn=
ings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blan=
k">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
Uh, I don&#39;t think we are ready to design anything yet - I am still 100%=
 confused on what the use case we are trying to solve is. Is it forking on =
at the browser, in intermediate node, in SIP, parallel, sequential or what?=
 I want to see the end user use cases we need to deal with. The example Ran=
dell gave seems reasonable but seem like it would best be handled with just=
 multiple independent PeerConnection. The combination of requiring ICE and =
SIP parallel forking is very complicated - particularly without Reliable pr=
ovisions responses and all the complexity that ensues.<br>

<br>
So before we go down this path, lets get clearer on what we are trying to a=
ccomplish.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"></font></span><br></blockquo=
te></div><br>One use case is interop with SIP with parallel forking, reliab=
le provisioning responses, UPDATE, and ICE. The big question regarding this=
 use case is availability of anything that supports parallel forking, relia=
ble provisioning and UPDATE. I actually think ICE makes implementation of t=
his use case easier, not harder.<br>
<br>The real end user use case is any type of service where calling destina=
tion is a small group of people. This allows to initiate a single offer and=
 send it to all the members of the group. In most cases only one person wil=
l reply, but, in a smaller set of cases, several people from the group will=
 answer. In the later case, you need a mechanism to clone the original peer=
 connection (or to create additional peer connections based on the original=
 offer) and deal with additional answers independently. <br>
<br>This is a very common use case for any type of service where people use=
 multiple devices. For instance if we add voice/video calling to a web mail=
 program, I can be logged in from multiple devices. It can also be a shared=
 account (like <a href=3D"mailto:support@acme.com">support@acme.com</a>) wh=
ere several different people are logged in to the same account. When someon=
e places a call to a shared account, several people can answer at the same =
time.<br>
<br>I know this can be addressed by sending a separate message=A0 and rever=
sing the call setup direction (each accepting party creates an offer and se=
nds it back), but this causes another set of issues related to slow call se=
tup and media clipping.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br>

--e89a8ff24e1124777404bfb3f206--

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Date: Thu, 10 May 2012 12:53:13 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: Cullen Jennings <fluffy@iii.ca>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 10 May 2012 12:05, Cullen Jennings <fluffy@iii.ca> wrote:
> I'm a very confused on what type of forking (with ICE) use case people want to satisfy.

Here's the use case as I understand it:

create offer and send to some thing
the offer actually hits multiple things
multiple answers arrive
want to establish sessions with all of them

That's the use case.  Adding it to the use cases draft (or not) is
probably something that needs to be debated more.

--Martin

p.s. How the offer reaches multiple things doesn't matter much.  I see
too many emails on these lists that include the letters S, I and P
together.

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To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On May 10, 2012, at 1:51 PM, Roman Shpount wrote:

>=20
> On Thu, May 10, 2012 at 3:06 PM, Cullen Jennings <fluffy@iii.ca> =
wrote:
>=20
> Uh, I don't think we are ready to design anything yet - I am still =
100% confused on what the use case we are trying to solve is. Is it =
forking on at the browser, in intermediate node, in SIP, parallel, =
sequential or what? I want to see the end user use cases we need to deal =
with. The example Randell gave seems reasonable but seem like it would =
best be handled with just multiple independent PeerConnection. The =
combination of requiring ICE and SIP parallel forking is very =
complicated - particularly without Reliable provisions responses and all =
the complexity that ensues.
>=20
> So before we go down this path, lets get clearer on what we are trying =
to accomplish.
>=20
>=20
> One use case is interop with SIP with parallel forking, reliable =
provisioning responses, UPDATE, and ICE. The big question regarding this =
use case is availability of anything that supports parallel forking, =
reliable provisioning and UPDATE. I actually think ICE makes =
implementation of this use case easier, not harder.
>=20
> The real end user use case is any type of service where calling =
destination is a small group of people. This allows to initiate a single =
offer and send it to all the members of the group. In most cases only =
one person will reply, but, in a smaller set of cases, several people =
from the group will answer. In the later case, you need a mechanism to =
clone the original peer connection (or to create additional peer =
connections based on the original offer) and deal with additional =
answers independently.=20
>=20
> This is a very common use case for any type of service where people =
use multiple devices. For instance if we add voice/video calling to a =
web mail program, I can be logged in from multiple devices. It can also =
be a shared account (like support@acme.com) where several different =
people are logged in to the same account. When someone places a call to =
a shared account, several people can answer at the same time.
>=20
> I know this can be addressed by sending a separate message  and =
reversing the call setup direction (each accepting party creates an =
offer and sends it back), but this causes another set of issues related =
to slow call setup and media clipping.
> _____________
> Roman Shpount
>=20

So let me try and make sure I understand this uses case. The web browser =
sends some signaling to a server that convert it to a sip call to the =
fork@example.com address. This sends it to the example.com sip server =
that parallel forks it to two SIP UASs  that are so two phones ring at =
the same time. Now both theses phones are going to negotiate ICE by =
using reliably provisional responses.=20

What products actually do parallel forking like this and return =
provisional reliable responses with ICE. I'm sort of questioning this is =
a common use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX =
systems I have seen that support ringing multiple phones at the same =
time don't seem to do this way. They manage to hide the parallel forking =
with a B2BUA model and make it look like sequential forking.=20



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From: Justin Uberti <juberti@google.com>
Date: Thu, 10 May 2012 16:56:07 -0400
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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The other use case is the one Partha originally mentioned, and recently
resurfaced:

Issue 2)  UPDATE support within SIP early dialog in the non-forking
scenario. Here, the cloning will not solve issue as the answerer originated
the new offer during SDP_PRANSWER state . Issue 2 is the title of this mail
thread.

which I think expands to:
1. local PC1 sends offer, forks to remote PC1 and PC2
2. remote PC1 receives offer, sends pranswer
3. local PC1 applies this pranswer
4. remote PC2 receives offer, sends pranswer
5. local PC1 applies this pranswer
6. remote PC1 sends update (i.e. new offer)
7. local PC1 does ???

Richard mentioned that this could be solved by cloning, and I think I agree.
local PC1 can clone at step 5, and create local PC2. local PC2 stays in the
pranswer state; local PC1 applies the previous pranswer as final, and then
handles the new offer (update).


On Thu, May 10, 2012 at 3:53 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 10 May 2012 12:05, Cullen Jennings <fluffy@iii.ca> wrote:
> > I'm a very confused on what type of forking (with ICE) use case people
> want to satisfy.
>
> Here's the use case as I understand it:
>
> create offer and send to some thing
> the offer actually hits multiple things
> multiple answers arrive
> want to establish sessions with all of them
>
> That's the use case.  Adding it to the use cases draft (or not) is
> probably something that needs to be debated more.
>
> --Martin
>
> p.s. How the offer reaches multiple things doesn't matter much.  I see
> too many emails on these lists that include the letters S, I and P
> together.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div>The other use case is the one Partha originally mentioned, and recentl=
y resurfaced:</div><div><br></div><div><p class=3D"MsoNormal" style><span s=
tyle=3D"font-size:11pt;font-family:Calibri,sans-serif;color:rgb(31,73,125)"=
>Issue 2)=C2=A0 UPDATE support within SIP early dialog in the non-forking s=
cenario. Here, the cloning will not solve issue as the answerer originated =
the new offer during SDP_PRANSWER state . Issue 2 is the title of this mail=
 thread.</span></p>

<div><br></div><div>which I think expands to:</div><div>1. local PC1 sends =
offer, forks to remote PC1 and PC2</div><div>2. remote PC1 receives offer, =
sends pranswer</div><div>3. local PC1 applies this pranswer</div><div>
4. remote PC2 receives offer, sends pranswer</div>
<div>5. local PC1 applies this pranswer</div><div>6. remote PC1 sends updat=
e (i.e. new offer)</div><div>7. local PC1 does ???=C2=A0</div><div><br></di=
v><div>Richard mentioned that this could be solved by cloning, and I think =
I agree.</div>

<div>local PC1 can clone at step 5, and create local PC2. local PC2 stays i=
n the pranswer state; local PC1 applies the previous pranswer as final, and=
 then handles the new offer (update).</div><div><br></div><br><div class=3D=
"gmail_quote">

On Thu, May 10, 2012 at 3:53 PM, Martin Thomson <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gma=
il.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div class=3D"im">On 10 May 2012 12:05, Cullen Jennings &lt;<a href=3D"mail=
to:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt; I&#39;m a very confused on what type of forking (with ICE) use case pe=
ople want to satisfy.<br>
<br>
</div>Here&#39;s the use case as I understand it:<br>
<br>
create offer and send to some thing<br>
the offer actually hits multiple things<br>
multiple answers arrive<br>
want to establish sessions with all of them<br>
<br>
That&#39;s the use case. =C2=A0Adding it to the use cases draft (or not) is=
<br>
probably something that needs to be debated more.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
--Martin<br>
</font></span><br>
p.s. How the offer reaches multiple things doesn&#39;t matter much. =C2=A0I=
 see<br>
too many emails on these lists that include the letters S, I and P<br>
together.<br>
<div class=3D"HOEnZb"><div class=3D"h5">___________________________________=
____________<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>

--00235429db9079ac4c04bfb4da15--

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I was looking more for the actually sort of end user use case where this =
might happen.=20


On May 10, 2012, at 1:53 PM, Martin Thomson wrote:

> On 10 May 2012 12:05, Cullen Jennings <fluffy@iii.ca> wrote:
>> I'm a very confused on what type of forking (with ICE) use case =
people want to satisfy.
>=20
> Here's the use case as I understand it:
>=20
> create offer and send to some thing
> the offer actually hits multiple things
> multiple answers arrive
> want to establish sessions with all of them
>=20
> That's the use case.  Adding it to the use cases draft (or not) is
> probably something that needs to be debated more.
>=20
> --Martin
>=20
> p.s. How the offer reaches multiple things doesn't matter much.  I see
> too many emails on these lists that include the letters S, I and P
> together.


From martin.thomson@gmail.com  Thu May 10 14:07:37 2012
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On 10 May 2012 13:56, Cullen Jennings <fluffy@iii.ca> wrote:
> I was looking more for the actually sort of end user use case where this might happen.

I ring you.
Your mobile and home phone both ring.
You answer your mobile, your daughter answers the home phone.
I get to talk to both of you.

Of course, that's not such a great outcome.  I can hear both of you,
but you can't hear each other unless I have really bad echo
cancellation.

I tend to think of this as solving a really interesting signaling
problem that doesn't really have any practical application.  If you
want to talk to both people, then you should signal the creation of a
conference call.

--Martin

From ibc@aliax.net  Thu May 10 14:13:04 2012
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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Thu, 10 May 2012 23:12:41 +0200
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2012/5/10 Cullen Jennings <fluffy@iii.ca>:
> The Nortel, Cisco, Avaya, Siemens etc enterprises PBX systems I have seen=
 that support ringing multiple phones at the same time don't seem to do thi=
s way. They manage to hide the parallel forking with a B2BUA model and make=
 it look like sequential forking.

Hi Cullen, I strongly prefer that WebRTC use cases are based on real
standards (i.e. pure SIP proxies) instead of non-standard devices
(like B2BUA's that are defined nowhere).

At least for me, the use case in which a "call" arrives to two
different devices and both generate direct (early-media) to the origin
is a mandatory use case for WebRTC, and I don't think that we should
discard this use case by design in any way.

Regards.

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

From roman@telurix.com  Thu May 10 14:23:33 2012
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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <4FA2E264.4040207@jesup.org> <7F2072F1E0DE894DA4B517B93C6A05852C44001338@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001339@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxvFYy-9XAE5GG0n58xX2HKRX3o5qkTZ2o6rugakaM77Ew@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400133A@ESESSCMS0356.eemea.ericsson.se> <CAOJ7v-1Ue=q4MAh0csiDLvRFfMGaJ4vWrDxhir-S-OjSTsymaw@mail.gmail.com> <90DEBA9A-E8D2-4662-9A41-95B361FDAF05@iii.ca> <CAD5OKxvXstf7GOcZc7qdFB6GPvZyzBP8cv+JQcDK6rzbsmA6-Q@mail.gmail.com> <F6B988EE-8847-4AC6-806C-A9B96A075BDB@iii.ca>
Date: Thu, 10 May 2012 17:23:30 -0400
Message-ID: <CAD5OKxsw+mJ5Ou7_p+nJU55iwuh1xnNrF0YEeST_sBUFWLwLHA@mail.gmail.com>
From: Roman Shpount <roman@telurix.com>
To: Cullen Jennings <fluffy@iii.ca>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On Thu, May 10, 2012 at 4:55 PM, Cullen Jennings <fluffy@iii.ca> wrote:

> > One use case is interop with SIP with parallel forking, reliable
> provisioning responses, UPDATE, and ICE. The big question regarding this
> use case is availability of anything that supports parallel forking,
> reliable provisioning and UPDATE. I actually think ICE makes implementation
> of this use case easier, not harder.
>
> So let me try and make sure I understand this uses case. The web browser
> sends some signaling to a server that convert it to a sip call to the
> fork@example.com address. This sends it to the example.com sip server
> that parallel forks it to two SIP UASs  that are so two phones ring at the
> same time. Now both theses phones are going to negotiate ICE by using
> reliably provisional responses.
>
> What products actually do parallel forking like this and return
> provisional reliable responses with ICE. I'm sort of questioning this is a
> common use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX
> systems I have seen that support ringing multiple phones at the same time
> don't seem to do this way. They manage to hide the parallel forking with a
> B2BUA model and make it look like sequential forking.
>
>
Let's say a standard SIP proxy (call it opensips for specificity) placing a
forked call to two ICE enabled IP phones. Both phones answer with 200 OK.

_____________
Roman Shpount

--047d7b33d55039e28404bfb53bca
Content-Type: text/html; charset=ISO-8859-1
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<br><div class=3D"gmail_quote">On Thu, May 10, 2012 at 4:55 PM, Cullen Jenn=
ings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" target=3D"_blan=
k">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote"=
 style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class=3D"HOEnZb"><div class=3D"h5">
&gt; One use case is interop with SIP with parallel forking, reliable provi=
sioning responses, UPDATE, and ICE. The big question regarding this use cas=
e is availability of anything that supports parallel forking, reliable prov=
isioning and UPDATE. I actually think ICE makes implementation of this use =
case easier, not harder.<br>

<br>
</div></div>So let me try and make sure I understand this uses case. The we=
b browser sends some signaling to a server that convert it to a sip call to=
 the <a href=3D"mailto:fork@example.com">fork@example.com</a> address. This=
 sends it to the <a href=3D"http://example.com" target=3D"_blank">example.c=
om</a> sip server that parallel forks it to two SIP UASs =A0that are so two=
 phones ring at the same time. Now both theses phones are going to negotiat=
e ICE by using reliably provisional responses.<br>

<br>
What products actually do parallel forking like this and return provisional=
 reliable responses with ICE. I&#39;m sort of questioning this is a common =
use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX systems I h=
ave seen that support ringing multiple phones at the same time don&#39;t se=
em to do this way. They manage to hide the parallel forking with a B2BUA mo=
del and make it look like sequential forking.<br>

<br>

</blockquote></div><br>Let&#39;s say a standard SIP proxy (call it opensips=
 for specificity) placing a forked call to two ICE enabled IP phones. Both =
phones answer with 200 OK.<br><br>_____________<br>Roman Shpount<br>
<br>

--047d7b33d55039e28404bfb53bca--

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To: Roman Shpount <roman@telurix.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On May 10, 2012, at 3:23 PM, Roman Shpount wrote:

>=20
> On Thu, May 10, 2012 at 4:55 PM, Cullen Jennings <fluffy@iii.ca> =
wrote:
> > One use case is interop with SIP with parallel forking, reliable =
provisioning responses, UPDATE, and ICE. The big question regarding this =
use case is availability of anything that supports parallel forking, =
reliable provisioning and UPDATE. I actually think ICE makes =
implementation of this use case easier, not harder.
>=20
> So let me try and make sure I understand this uses case. The web =
browser sends some signaling to a server that convert it to a sip call =
to the fork@example.com address. This sends it to theexample.com sip =
server that parallel forks it to two SIP UASs  that are so two phones =
ring at the same time. Now both theses phones are going to negotiate ICE =
by using reliably provisional responses.
>=20
> What products actually do parallel forking like this and return =
provisional reliable responses with ICE. I'm sort of questioning this is =
a common use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX =
systems I have seen that support ringing multiple phones at the same =
time don't seem to do this way. They manage to hide the parallel forking =
with a B2BUA model and make it look like sequential forking.
>=20
>=20
> Let's say a standard SIP proxy (call it opensips for specificity) =
placing a forked call to two ICE enabled IP phones. Both phones answer =
with 200 OK.

With a standard SIP proxy, only one of the 200 would be sent back to the =
web browser.=20=

From roman@telurix.com  Thu May 10 15:40:42 2012
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Date: Thu, 10 May 2012 18:40:39 -0400
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From: Roman Shpount <roman@telurix.com>
To: Cullen Jennings <fluffy@iii.ca>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On Thu, May 10, 2012 at 6:29 PM, Cullen Jennings <fluffy@iii.ca> wrote:

>
> On May 10, 2012, at 3:23 PM, Roman Shpount wrote:
>
> >
> > On Thu, May 10, 2012 at 4:55 PM, Cullen Jennings <fluffy@iii.ca> wrote:
> > > One use case is interop with SIP with parallel forking, reliable
> provisioning responses, UPDATE, and ICE. The big question regarding this
> use case is availability of anything that supports parallel forking,
> reliable provisioning and UPDATE. I actually think ICE makes implementation
> of this use case easier, not harder.
> >
> > So let me try and make sure I understand this uses case. The web browser
> sends some signaling to a server that convert it to a sip call to the
> fork@example.com address. This sends it to theexample.com sip server that
> parallel forks it to two SIP UASs  that are so two phones ring at the same
> time. Now both theses phones are going to negotiate ICE by using reliably
> provisional responses.
> >
> > What products actually do parallel forking like this and return
> provisional reliable responses with ICE. I'm sort of questioning this is a
> common use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX
> systems I have seen that support ringing multiple phones at the same time
> don't seem to do this way. They manage to hide the parallel forking with a
> B2BUA model and make it look like sequential forking.
> >
> >
> > Let's say a standard SIP proxy (call it opensips for specificity)
> placing a forked call to two ICE enabled IP phones. Both phones answer with
> 200 OK.
>
> With a standard SIP proxy, only one of the 200 would be sent back to the
> web browser.


AFAIK a standard proxy will forward multiple 200 OK responses (see section
16.7 of RFC 3261). We have the same problem with 183 responses from two
phones anyway.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Thu, May 10, 2012 at 6:29 P=
M, Cullen Jennings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@iii.ca" t=
arget=3D"_blank">fluffy@iii.ca</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<div class=3D"im"><br>
On May 10, 2012, at 3:23 PM, Roman Shpount wrote:<br>
<br>
&gt;<br>
&gt; On Thu, May 10, 2012 at 4:55 PM, Cullen Jennings &lt;<a href=3D"mailto=
:fluffy@iii.ca">fluffy@iii.ca</a>&gt; wrote:<br>
&gt; &gt; One use case is interop with SIP with parallel forking, reliable =
provisioning responses, UPDATE, and ICE. The big question regarding this us=
e case is availability of anything that supports parallel forking, reliable=
 provisioning and UPDATE. I actually think ICE makes implementation of this=
 use case easier, not harder.<br>

&gt;<br>
</div>&gt; So let me try and make sure I understand this uses case. The web=
 browser sends some signaling to a server that convert it to a sip call to =
the <a href=3D"mailto:fork@example.com">fork@example.com</a> address. This =
sends it to <a href=3D"http://theexample.com" target=3D"_blank">theexample.=
com</a> sip server that parallel forks it to two SIP UASs =A0that are so tw=
o phones ring at the same time. Now both theses phones are going to negotia=
te ICE by using reliably provisional responses.<br>

<div class=3D"im">&gt;<br>
&gt; What products actually do parallel forking like this and return provis=
ional reliable responses with ICE. I&#39;m sort of questioning this is a co=
mmon use case. The Nortel, Cisco, Avaya, Siemens etc enterprises PBX system=
s I have seen that support ringing multiple phones at the same time don&#39=
;t seem to do this way. They manage to hide the parallel forking with a B2B=
UA model and make it look like sequential forking.<br>

&gt;<br>
&gt;<br>
&gt; Let&#39;s say a standard SIP proxy (call it opensips for specificity) =
placing a forked call to two ICE enabled IP phones. Both phones answer with=
 200 OK.<br>
<br>
</div>With a standard SIP proxy, only one of the 200 would be sent back to =
the web browser. </blockquote><div><br>AFAIK a standard proxy will forward =
multiple 200 OK responses (see section 16.7 of RFC 3261). We have the same =
problem with 183 responses from two phones anyway.<br>
</div></div>_____________<br>Roman Shpount<br>

--e89a8ff2437129f22904bfb64ffe--

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To: Martin Thomson <martin.thomson@gmail.com>
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On May 10, 2012, at 3:07 PM, Martin Thomson wrote:

> On 10 May 2012 13:56, Cullen Jennings <fluffy@iii.ca> wrote:
>> I was looking more for the actually sort of end user use case where =
this might happen.
>=20
> I ring you.
> Your mobile and home phone both ring.
> You answer your mobile, your daughter answers the home phone.
> I get to talk to both of you.
>=20
> Of course, that's not such a great outcome.  I can hear both of you,
> but you can't hear each other unless I have really bad echo
> cancellation.
>=20
> I tend to think of this as solving a really interesting signaling
> problem that doesn't really have any practical application.  If you
> want to talk to both people, then you should signal the creation of a
> conference call.
>=20
> --Martin

That an good use cases, except the people need to be able to hear each =
other. It gets called lots of things but some PBXs call it bridged line =
appearance. Variation of it have been discussed in bliss and =
draft-ietf-bliss-shared-appearances. It really comes down to where you =
think the conferencing should happen. If it does not happened on the the =
callers phone, seems like existing jsep idea should be able to do this.  =
(given the callee is the phone that is most likely out of administrative =
control of the other two phones that set up bridged line a appearance, =
assuming that phone can and is willing to do the conferencing seems like =
a bad architecture for this problem). Again lot of systems do this today =
in SIP and seem to do it without needing forcing the callers phones to =
deal with multiple simultaneous calls signaled in some way I don't even =
know how to signal in SIP.=20



From stefan.lk.hakansson@ericsson.com  Fri May 11 06:01:04 2012
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Subject: [rtcweb] New use-case proposed
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I've been sketching on a new use-case. The use-case is intended to
derive requirements that enable low complex central nodes for multi
party sessions, which in turn leads to requirements regarding
(essentially) keying solution for SRTP and the possibility for the app
to control/set things in the media configuration (in other words, for JSEP):


Multi-party with low complexity central node
==============================================

A geographically spread (charity, non-profit, school) organization
offers its members multi-party video communication via a shared virtual
room that participants can enter and leave at any time. At times there
are many persons in the virtual room (20+), at other times very few
(3-5, or even 0-1).

The application will control if few participants (a subset) are shown at
a higher fidelity or if many (all) are shown at a lower fidelity or a
mix of some few at high fidelity and the rest at much lower fidelity
given the existing bandwidth limitations between participants.

Since there are at times many persons in the virtual room, it is not
feasible to set up a mesh. The bandwidth needed by a participant exceeds
what many members have access to, especially in their uplinks, so
instead a central media node is deployed. In addition, the organization
does not have access to much funds, but has to rely on cheap hardware
(often donated) operated by volunteers.

  From this use-case a high level requirement saying something like

"It must be possible to set up media streams and encryption in such
a way that processing in a central node is minimized (no transcoding
required, no RTP packet rewriting, no decryption/re-encryption for every
outgoing flow - just simple forwarding)."

can be derived. This can in turn be broken down into more detailed
requirements such as:

- The keying solution must allow each participants to encrypt to
multiple receivers without any decryption+encryption in the middle node

- RTP sessions that have SSRCs from multiple PeerConnections being
interconneted must be supported. From a given end-point all SSRCs will
come over one PC, but the full path will be different towards different
sets of SSRCs.

- Signalling must be capable of establishing a common set of receiver
configurations over all participants.

- The API must allow for the above, i.e.:
-- The app must be able to chose a keying solution that enables
encryption to multiple receivers (if the app can have any control over
keying method used)
-- The app must be able to control SSRCs to use for outgoing streams
-- The app must be able to control the receiver configuration the
browser uses

Is this something we should consider adding to the use-case document?

Br,
Stefan

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From: Justin Uberti <juberti@google.com>
Date: Fri, 11 May 2012 10:24:12 -0400
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New use-case proposed
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SRTP encryption is cheap, practically free on today's CPUs. I agree with
the general use case but I don't agree with the conclusion that we need to
require that the middlebox does not rewrite packets or perform crypto,
mainly since it will increase system complexity (and the system is already
quite complex).

On Fri, May 11, 2012 at 9:01 AM, Stefan Hakansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> I've been sketching on a new use-case. The use-case is intended to
> derive requirements that enable low complex central nodes for multi
> party sessions, which in turn leads to requirements regarding
> (essentially) keying solution for SRTP and the possibility for the app
> to control/set things in the media configuration (in other words, for
> JSEP):
>
>
> Multi-party with low complexity central node
> ==============================**================
>
> A geographically spread (charity, non-profit, school) organization
> offers its members multi-party video communication via a shared virtual
> room that participants can enter and leave at any time. At times there
> are many persons in the virtual room (20+), at other times very few
> (3-5, or even 0-1).
>
> The application will control if few participants (a subset) are shown at
> a higher fidelity or if many (all) are shown at a lower fidelity or a
> mix of some few at high fidelity and the rest at much lower fidelity
> given the existing bandwidth limitations between participants.
>
> Since there are at times many persons in the virtual room, it is not
> feasible to set up a mesh. The bandwidth needed by a participant exceeds
> what many members have access to, especially in their uplinks, so
> instead a central media node is deployed. In addition, the organization
> does not have access to much funds, but has to rely on cheap hardware
> (often donated) operated by volunteers.
>
>  From this use-case a high level requirement saying something like
>
> "It must be possible to set up media streams and encryption in such
> a way that processing in a central node is minimized (no transcoding
> required, no RTP packet rewriting, no decryption/re-encryption for every
> outgoing flow - just simple forwarding)."
>
> can be derived. This can in turn be broken down into more detailed
> requirements such as:
>
> - The keying solution must allow each participants to encrypt to
> multiple receivers without any decryption+encryption in the middle node
>
> - RTP sessions that have SSRCs from multiple PeerConnections being
> interconneted must be supported. From a given end-point all SSRCs will
> come over one PC, but the full path will be different towards different
> sets of SSRCs.
>
> - Signalling must be capable of establishing a common set of receiver
> configurations over all participants.
>
> - The API must allow for the above, i.e.:
> -- The app must be able to chose a keying solution that enables
> encryption to multiple receivers (if the app can have any control over
> keying method used)
> -- The app must be able to control SSRCs to use for outgoing streams
> -- The app must be able to control the receiver configuration the
> browser uses
>
> Is this something we should consider adding to the use-case document?
>
> Br,
> Stefan
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--20cf3074b596b5b1a404bfc37e64
Content-Type: text/html; charset=UTF-8
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SRTP encryption is cheap, practically free on today&#39;s CPUs. I agree wit=
h the general use case but I don&#39;t agree with the conclusion that we ne=
ed to require that the middlebox does not rewrite packets or perform crypto=
, mainly since it will increase system complexity (and the system is alread=
y quite complex).<br>

<br><div class=3D"gmail_quote">On Fri, May 11, 2012 at 9:01 AM, Stefan Haka=
nsson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakansson@ericss=
on.com" target=3D"_blank">stefan.lk.hakansson@ericsson.com</a>&gt;</span> w=
rote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">I&#39;ve been sketching on a new use-case. T=
he use-case is intended to<br>
derive requirements that enable low complex central nodes for multi<br>
party sessions, which in turn leads to requirements regarding<br>
(essentially) keying solution for SRTP and the possibility for the app<br>
to control/set things in the media configuration (in other words, for JSEP)=
:<br>
<br>
<br>
Multi-party with low complexity central node<br>
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D<u></u>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D<br>
<br>
A geographically spread (charity, non-profit, school) organization<br>
offers its members multi-party video communication via a shared virtual<br>
room that participants can enter and leave at any time. At times there<br>
are many persons in the virtual room (20+), at other times very few<br>
(3-5, or even 0-1).<br>
<br>
The application will control if few participants (a subset) are shown at<br=
>
a higher fidelity or if many (all) are shown at a lower fidelity or a<br>
mix of some few at high fidelity and the rest at much lower fidelity<br>
given the existing bandwidth limitations between participants.<br>
<br>
Since there are at times many persons in the virtual room, it is not<br>
feasible to set up a mesh. The bandwidth needed by a participant exceeds<br=
>
what many members have access to, especially in their uplinks, so<br>
instead a central media node is deployed. In addition, the organization<br>
does not have access to much funds, but has to rely on cheap hardware<br>
(often donated) operated by volunteers.<br>
<br>
=C2=A0From this use-case a high level requirement saying something like<br>
<br>
&quot;It must be possible to set up media streams and encryption in such<br=
>
a way that processing in a central node is minimized (no transcoding<br>
required, no RTP packet rewriting, no decryption/re-encryption for every<br=
>
outgoing flow - just simple forwarding).&quot;<br>
<br>
can be derived. This can in turn be broken down into more detailed<br>
requirements such as:<br>
<br>
- The keying solution must allow each participants to encrypt to<br>
multiple receivers without any decryption+encryption in the middle node<br>
<br>
- RTP sessions that have SSRCs from multiple PeerConnections being<br>
interconneted must be supported. From a given end-point all SSRCs will<br>
come over one PC, but the full path will be different towards different<br>
sets of SSRCs.<br>
<br>
- Signalling must be capable of establishing a common set of receiver<br>
configurations over all participants.<br>
<br>
- The API must allow for the above, i.e.:<br>
-- The app must be able to chose a keying solution that enables<br>
encryption to multiple receivers (if the app can have any control over<br>
keying method used)<br>
-- The app must be able to control SSRCs to use for outgoing streams<br>
-- The app must be able to control the receiver configuration the<br>
browser uses<br>
<br>
Is this something we should consider adding to the use-case document?<br>
<br>
Br,<br>
Stefan<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>
</blockquote></div><br>

--20cf3074b596b5b1a404bfc37e64--

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From: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 11 May 2012 09:26:24 -0500
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Support for multiple registered contacts and (parallel) forking are pretty =
fundamental to SIP.  This has little to do with conferencing.  Some systems=
 (not all) may support only serial forking or may serialize media interacti=
ons associated with parallel forking, but forking remains widely deployed a=
nd useful.

The point is not to support simultaneous media flow with all forked targets=
 (your talking to me and my daughter at the same time in Martin's example),=
 but to support the ability to negotiate ICE and DTLS independently with ea=
ch target before one of them "answers", to minimize the potential for clipp=
ing.

True parallel forking has been around for a long time and is supported in s=
ome (not all) existing systems.  We do not mandate ICE support (for example=
) because it is commonly deployed, but because it is useful; the same princ=
iple should apply here.  I have not heard that simultaneous alerting of mul=
tiple registered contacts is not useful, but just that some existing system=
s find ways of serializing the media interactions.  That is relatively easi=
er to do when you don't support ICE and DTLS since media can flow almost as=
 soon as you signal the SDP changes.

But things are different with ICE and DTLS, particularly if you want to avo=
id media intermediaries.  Significant clipping is a real possibility if we =
have to restart ICE/DTLS after the "winner" responds with 200 OK.

This is not a separate use case, but a given aspect of interworking with SI=
P.

Richard=20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Cullen Jennings
Sent: Thursday, May 10, 2012 5:48 PM
To: Martin Thomson
Cc: Randell Jesup; rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP


On May 10, 2012, at 3:07 PM, Martin Thomson wrote:

> On 10 May 2012 13:56, Cullen Jennings <fluffy@iii.ca> wrote:
>> I was looking more for the actually sort of end user use case where this=
 might happen.
>=20
> I ring you.
> Your mobile and home phone both ring.
> You answer your mobile, your daughter answers the home phone.
> I get to talk to both of you.
>=20
> Of course, that's not such a great outcome.  I can hear both of you,
> but you can't hear each other unless I have really bad echo
> cancellation.
>=20
> I tend to think of this as solving a really interesting signaling
> problem that doesn't really have any practical application.  If you
> want to talk to both people, then you should signal the creation of a
> conference call.
>=20
> --Martin

That an good use cases, except the people need to be able to hear each othe=
r. It gets called lots of things but some PBXs call it bridged line appeara=
nce. Variation of it have been discussed in bliss and draft-ietf-bliss-shar=
ed-appearances. It really comes down to where you think the conferencing sh=
ould happen. If it does not happened on the the callers phone, seems like e=
xisting jsep idea should be able to do this.  (given the callee is the phon=
e that is most likely out of administrative control of the other two phones=
 that set up bridged line a appearance, assuming that phone can and is will=
ing to do the conferencing seems like a bad architecture for this problem).=
 Again lot of systems do this today in SIP and seem to do it without needin=
g forcing the callers phones to deal with multiple simultaneous calls signa=
led in some way I don't even know how to signal in SIP.=20


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

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To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:

> But things are different with ICE and DTLS, particularly if you want =
to avoid media intermediaries.  Significant clipping is a real =
possibility if we have to restart ICE/DTLS after the "winner" responds =
with 200 OK.
>=20
> This is not a separate use case, but a given aspect of interworking =
with SIP.

The whole problem of clipping audio due to the delay between the 200OK =
and the media actually flowing raises a flag that we are triggering =
upper layers of the stack on the wrong event.=20

Why is anyone sending media before the media channel is fully =
established? Why tell the user (or IVR etc) that the call is answered =
before we have a working media channel?

Neil


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Date: Fri, 11 May 2012 18:21:27 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] New use-case proposed
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I propose to reject this use case because it requires yet another 
security redesign.
The clue is here:

>
> - The keying solution must allow each participants to encrypt to
> multiple receivers without any decryption+encryption in the middle node
>

This means that each participant must use the same encryption keys to 
all other participants; this in turn means that when someone leaves the 
group, all participants must change their encryption keys; it also means 
that as long as shared keys are used for authentication, all 
participants can impersonate all other participants.

In fact, this solution has most of the issues (except for the network 
layer deployment issue) that leads me to strongly advocate leaving 
multicast out of scope for this effort.

                Harald



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--e89a8ff2445b8dd38104bfc5329b
Content-Type: text/plain; charset=ISO-8859-1

+1

I have provided the CPU requirements for doing SRTP re-encryption before.
Bottom line, on the real conferencing service encryption adds about 10%
extra CPU load. There is no benefit in compromising security to avoid
re-encrypting.
_____________
Roman Shpount


On Fri, May 11, 2012 at 12:21 PM, Harald Alvestrand <harald@alvestrand.no>wrote:

> I propose to reject this use case because it requires yet another security
> redesign.
> The clue is here:
>
>
>> - The keying solution must allow each participants to encrypt to
>> multiple receivers without any decryption+encryption in the middle node
>>
>>
> This means that each participant must use the same encryption keys to all
> other participants; this in turn means that when someone leaves the group,
> all participants must change their encryption keys; it also means that as
> long as shared keys are used for authentication, all participants can
> impersonate all other participants.
>
> In fact, this solution has most of the issues (except for the network
> layer deployment issue) that leads me to strongly advocate leaving
> multicast out of scope for this effort.
>
>               Harald
>
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--e89a8ff2445b8dd38104bfc5329b
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+1<br><br>I have provided the CPU requirements for doing SRTP re-encryption=
 before. Bottom line, on the real conferencing service encryption adds abou=
t 10% extra CPU load. There is no benefit in compromising security to avoid=
 re-encrypting.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br><br><div class=3D"gmail_quote">On Fri, May 11, 2012 at 12:21 PM, Harald=
 Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" t=
arget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid=
;padding-left:1ex">
I propose to reject this use case because it requires yet another security =
redesign.<br>
The clue is here:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
- The keying solution must allow each participants to encrypt to<br>
multiple receivers without any decryption+encryption in the middle node<br>
<br>
</blockquote>
<br>
This means that each participant must use the same encryption keys to all o=
ther participants; this in turn means that when someone leaves the group, a=
ll participants must change their encryption keys; it also means that as lo=
ng as shared keys are used for authentication, all participants can imperso=
nate all other participants.<br>

<br>
In fact, this solution has most of the issues (except for the network layer=
 deployment issue) that leads me to strongly advocate leaving multicast out=
 of scope for this effort.<br>
<br>
 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Harald<br>
<br>
<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>
</blockquote></div><br>

--e89a8ff2445b8dd38104bfc5329b--

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On 5/10/2012 6:47 PM, Cullen Jennings wrote:
> On May 10, 2012, at 3:07 PM, Martin Thomson wrote:
>
>> On 10 May 2012 13:56, Cullen Jennings<fluffy@iii.ca>  wrote:
>>> I was looking more for the actually sort of end user use case where this might happen.
>> I ring you.
>> Your mobile and home phone both ring.
>> You answer your mobile, your daughter answers the home phone.
>> I get to talk to both of you.
>>
>> Of course, that's not such a great outcome.  I can hear both of you,
>> but you can't hear each other unless I have really bad echo
>> cancellation.
>>
>> I tend to think of this as solving a really interesting signaling
>> problem that doesn't really have any practical application.  If you
>> want to talk to both people, then you should signal the creation of a
>> conference call.
>>
>> --Martin
> That an good use cases, except the people need to be able to hear each other. It gets called lots of things but some PBXs call it bridged line appearance. Variation of it have been discussed in bliss and draft-ietf-bliss-shared-appearances. It really comes down to where you think the conferencing should happen. If it does not happened on the the callers phone, seems like existing jsep idea should be able to do this.  (given the callee is the phone that is most likely out of administrative control of the other two phones that set up bridged line a appearance, assuming that phone can and is willing to do the conferencing seems like a bad architecture for this problem). Again lot of systems do this today in SIP and seem to do it without needing forcing the callers phones to deal with multiple simultaneous calls signaled in some way I don't even know how to signal in SIP.

I'll note I've built systems that simply handle in-phone conferencing; 
while there are some annoyances it's most certainly doable.  If the 
caller acts as the MCU you get OWD1+jitter+etc+encode+OWD2 
(OWD=one-way-delay) between the two leaf nodes (though this is no 
different than a normal conference).  Or your app, after the initial 
second answer tells the second answerer to invite the first into a mesh 
conference, if that works - and it may not, so you need to mix or 
forward if you want a conference.   (Forwarding the RTP traffic works at 
the cost of bandwidth and complexity, but simpler media processing and 
less leaf-to-leaf delay.)

If this *is* already a mesh conference you've joined (different usecase 
than above), no mixing or forwarding is needed - you render all the 
streams, and you send media to all the answerers.  This is exactly what 
you want.

However, if two people answer, you do not have to have a conference.  
First of all, the devices rtcweb is intended for generally have enough 
UI available that the user can manage multiple sessions reasonably, 
unlike POTS-type phones.  You can handle it much like a call-waiting 
scenario, though the second answerer might be confused unless you send 
some sort of hold/please-wait message (which, this not being POTS 
devices, you can).

In other cases/applications, the answers may be automated, and the 
channels may not be used immediately - think the VR example, where your 
OFFER is forwarded to any person who comes in "hearing" range, enters 
the room, whatever.  This may be a form of partial-mesh conference 
managed by the server.  These channels may be opened before they're 
actually needed in order to hide setup delay.

My point is to show that the possible uses of parallel forking exist, 
and don't always map to SIP or to classic telephony.  If cloning is 
supported, then these scenarios work.

I'm open to alternative proposals that cover this class of use-cases, if 
they make either "standard" apps simpler to a worthwhile degree, or if 
they make the implementation of webrtc itself noticably simpler, while 
still making these uses possible.  Alternatives may have downsides in 
some cases, in extra call setup delay/clipping or requiring re-invites 
or requiring special application protocols to request a offerer send a 
new invite, and added complexity to apps that wish to handle this.

-- 
Randell Jesup
randell-ietf@jesup.org



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I agree as well since based on our testing for a simple media bridge
that SRTP re-encryption adds a similar amount of CPU load (around 10%).=20

=20

Henry

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Roman Shpount
Sent: May-11-12 12:26 PM
To: Harald Alvestrand
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] New use-case proposed

=20

+1

I have provided the CPU requirements for doing SRTP re-encryption
before. Bottom line, on the real conferencing service encryption adds
about 10% extra CPU load. There is no benefit in compromising security
to avoid re-encrypting.
_____________
Roman Shpount



On Fri, May 11, 2012 at 12:21 PM, Harald Alvestrand
<harald@alvestrand.no> wrote:

I propose to reject this use case because it requires yet another
security redesign.
The clue is here:


- The keying solution must allow each participants to encrypt to
multiple receivers without any decryption+encryption in the middle node


This means that each participant must use the same encryption keys to
all other participants; this in turn means that when someone leaves the
group, all participants must change their encryption keys; it also means
that as long as shared keys are used for authentication, all
participants can impersonate all other participants.

In fact, this solution has most of the issues (except for the network
layer deployment issue) that leads me to strongly advocate leaving
multicast out of scope for this effort.

              Harald


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

=20


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I agree as well since based on our testing for a simple =
media
bridge that SRTP re-encryption adds a similar amount of CPU load (around =
10%). <o:p></o:p></span></p>

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

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

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

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<p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> rtcweb-bounces@ietf.org
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Roman Shpount<br>
<b>Sent:</b> May-11-12 12:26 PM<br>
<b>To:</b> Harald Alvestrand<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] New use-case proposed<o:p></o:p></span></p>

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</div>

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

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>+1<br>
<br>
I have provided the CPU requirements for doing SRTP re-encryption =
before.
Bottom line, on the real conferencing service encryption adds about 10% =
extra
CPU load. There is no benefit in compromising security to avoid =
re-encrypting.<br
clear=3Dall>
_____________<br>
Roman Shpount<br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal>On Fri, May 11, 2012 at 12:21 PM, Harald Alvestrand =
&lt;<a
href=3D"mailto:harald@alvestrand.no" =
target=3D"_blank">harald@alvestrand.no</a>&gt;
wrote:<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'>I propose to reject =
this use
case because it requires yet another security redesign.<br>
The clue is here:<o:p></o:p></p>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><br>
- The keying solution must allow each participants to encrypt to<br>
multiple receivers without any decryption+encryption in the middle =
node<o:p></o:p></p>

<p class=3DMsoNormal><br>
This means that each participant must use the same encryption keys to =
all other
participants; this in turn means that when someone leaves the group, all
participants must change their encryption keys; it also means that as =
long as
shared keys are used for authentication, all participants can =
impersonate all
other participants.<br>
<br>
In fact, this solution has most of the issues (except for the network =
layer
deployment issue) that leads me to strongly advocate leaving multicast =
out of
scope for this effort.<br>
<br>
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; Harald<br>
<br>
<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><o:p></=
o:p></p>

</div>

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

</div>

</div>

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>> Let's say a standard SIP proxy (call it opensips for specificity) 
>> placing a forked call to two ICE enabled IP phones. Both phones answer 
>> with 200 OK.
> With a standard SIP proxy, only one of the 200 would be sent back to the 
> web browser.

No, I do not think that is the case.  RFC 6026 modified parts of RFC 
3261, but the part that says that a proxy MUST sent multiple 2xx responses 
to  INVITE was kept as-is.  Specifically, RFC 6026 does not modify the 
behaviour of RFC 3261, Section 16.7, Step 5 that says:

         After a final response has been sent on the server transaction,
         the following responses MUST be forwarded immediately:

         -  Any 2xx response to an INVITE request

If I am misreading the specification, please let me know.

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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 11 May 2012 15:19:25 -0500
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Neil,
That's the point.  This discussion is with reference to parallel forking.  =
To avoid clipping, you need to establish the media channels with the regist=
ered targets before they are alerted because you don't know which one will =
answer.  This requires MediaConnection cloning so you can properly support =
parallel forking and do ICE and DTLS with each target in parallel before 20=
0 OK from one of the targets.  Delaying delivery of answer indication to th=
e caller (as you seem to imply) does not help.

Richard

-----Original Message-----
From: Neil Stratford [mailto:neils@vipadia.com] On Behalf Of Neil Stratford
Sent: Friday, May 11, 2012 10:05 AM
To: Ejzak, Richard P (Richard)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP


On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:

> But things are different with ICE and DTLS, particularly if you want to a=
void media intermediaries.  Significant clipping is a real possibility if w=
e have to restart ICE/DTLS after the "winner" responds with 200 OK.
>=20
> This is not a separate use case, but a given aspect of interworking with =
SIP.

The whole problem of clipping audio due to the delay between the 200OK and =
the media actually flowing raises a flag that we are triggering upper layer=
s of the stack on the wrong event.=20

Why is anyone sending media before the media channel is fully established? =
Why tell the user (or IVR etc) that the call is answered before we have a w=
orking media channel?

Neil


From harald@alvestrand.no  Fri May 11 13:52:20 2012
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Subject: Re: [rtcweb] New use-case proposed
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On 05/11/2012 06:38 PM, GÃ¶ran Eriksson AP wrote:
>
> 11 maj 2012 kl. 18:21 skrev "Harald Alvestrand"<harald@alvestrand.no>:
>
>> I propose to reject this use case because it requires yet another
>> security redesign.
>> The clue is here:
>>
>>> - The keying solution must allow each participants to encrypt to
>>> multiple receivers without any decryption+encryption in the middle node
>>>
>> This means that each participant must use the same encryption keys to
>> all other participants; this in turn means that when someone leaves the
>> group, all participants must change their encryption keys; it also means
>> that as long as shared keys are used for authentication, all
>> participants can impersonate all other participants.
>>
>> In fact, this solution has most of the issues (except for the network
>> layer deployment issue) that leads me to strongly advocate leaving
>> multicast out of scope for this effort.
> You mean for this phase of webrtc std efforts or for ever?
Until we've got an usage scenario with a deployment story that I find 
credible.

The point I'm pretty vehement on - that lack of multicast support should 
not be a blocker for finalizing the specs we are working on.

               Harald


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From dwing@cisco.com  Fri May 11 15:24:58 2012
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From: "Dan Wing" <dwing@cisco.com>
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> Personally, I would like to us to ensure we have at least have a good
> foundation for multicast later.

EKT, draft-ietf-avt-srtp-ekt, does that, and works with the initial
keying being Security Descriptions (should we allow that in RTCWEB) 
and with DTLS-SRTP.

-d




From marshall.eubanks@gmail.com  Fri May 11 18:11:41 2012
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On Fri, May 11, 2012 at 6:24 PM, Dan Wing <dwing@cisco.com> wrote:
>> Personally, I would like to us to ensure we have at least have a good
>> foundation for multicast later.
>
> EKT, draft-ietf-avt-srtp-ekt, does that, and works with the initial
> keying being Security Descriptions (should we allow that in RTCWEB)
> and with DTLS-SRTP.

Note : This draft expired last week. I presume that is not truly
indicative of its status.

Regards
Marshall

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

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> -----Original Message-----
> From: Marshall Eubanks [mailto:marshall.eubanks@gmail.com]
> Sent: Friday, May 11, 2012 6:12 PM
> To: Dan Wing
> Cc: G=F6ran Eriksson AP; Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] New use-case proposed
>=20
> On Fri, May 11, 2012 at 6:24 PM, Dan Wing <dwing@cisco.com> wrote:
> >> Personally, I would like to us to ensure we have at least have a
> good
> >> foundation for multicast later.
> >
> > EKT, draft-ietf-avt-srtp-ekt, does that, and works with the initial
> > keying being Security Descriptions (should we allow that in RTCWEB)
> > and with DTLS-SRTP.
>=20
> Note : This draft expired last week.  I presume that is not truly
> indicative of its status.

We are integrating the changes described in=20
http://www.ietf.org/mail-archive/web/avt/current/msg15232.html,
which I summarized in my RTCWEB presentation, starting on
slide 28 of =
http://www.ietf.org/proceedings/83/slides/slides-83-rtcweb-3.pdf

In that deck, slide 28 shows the problem of interworking DTLS-SRTP
with Security Descriptions -- namely, that something needs to=20
decrypt and re-encrypt SRTP packets in one direction.  This is shown
with flames coming out of that box.  (Its CPU is on fire.)

Slide 31 explains how EKT helps that interoperation.

Slide 33 shows how the interoperation, where the box no longer
has to decrypt/re-encrypt SRTP packets.

It was asked if we can do that interworking, between Security
Descriptions and DTLS-SRTP-EKT, without the media flowing through
a box.  Yes, we can -- except for EKT key changes (that is, key
changes from the WEBRTC side).  I have a design that will=20
handle that.  But folks still need to digest the interop model
and decide if allowing SDESC on the WEBRTC side is worth the
permanent interoperation burden (on all WEBRTC endpoints) and
worth the permanent reduction of identity and of media=20
encryption security.

-d


> Regards
> Marshall
>=20
> >
> > -d
> >
> >
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb


From michawe@ifi.uio.no  Sat May 12 01:58:32 2012
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On May 11, 2012, at 11:31 PM, G=F6ran Eriksson AP wrote:

>
>
> 11 maj 2012 kl. 22:52 skrev "Harald Alvestrand" =20
> <harald@alvestrand.no>:
>
>> On 05/11/2012 06:38 PM, G=F6ran Eriksson AP wrote:
>>>
>>> 11 maj 2012 kl. 18:21 skrev "Harald =20
>>> Alvestrand"<harald@alvestrand.no>:
>>>
>>>> I propose to reject this use case because it requires yet another
>>>> security redesign.
>>>> The clue is here:
>>>>
>>>>> - The keying solution must allow each participants to encrypt to
>>>>> multiple receivers without any decryption+encryption in the =20
>>>>> middle node
>>>>>
>>>> This means that each participant must use the same encryption =20
>>>> keys to
>>>> all other participants; this in turn means that when someone =20
>>>> leaves the
>>>> group, all participants must change their encryption keys; it =20
>>>> also means
>>>> that as long as shared keys are used for authentication, all
>>>> participants can impersonate all other participants.
>>>>
>>>> In fact, this solution has most of the issues (except for the =20
>>>> network
>>>> layer deployment issue) that leads me to strongly advocate leaving
>>>> multicast out of scope for this effort.
>>> You mean for this phase of webrtc std efforts or for ever?
>> Until we've got an usage scenario with a deployment story that I find
>> credible.
>>
>> The point I'm pretty vehement on - that lack of multicast support =20
>> should
>> not be a blocker for finalizing the specs we are working on.
>>
>
> I am pretty sensitive to such consequence as well! Having said that, =20=

> enabling innovation in this kind of application category is =20
> important  for the long term evolution of the web platform.
>
> Personally, I would like to us to ensure we have at least have a =20
> good foundation for multicast later.

+1 (to both parts of this)

Cheers
Michael


From bossiel@yahoo.fr  Sun May 13 22:14:43 2012
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---888064777-1348475897-1336972478=:22604
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Hi all,=0A=0AWe were working on WebRTC since 3 months and we'd like to shar=
e with you our open source HTML5 SIP client (codename: sipml5).=0AFor more =
information: http://www.sipml5.org=0ACode source: http://code.google.com/p/=
sipml5/=0AI hop it could help other developers.=0A=0ARegards,=0A
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<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div>Hi all,<br><br>W=
e were working on WebRTC since 3 months and we'd like to share with you our=
 open source HTML5 SIP client (codename: sipml5).<br>For more information: =
http://www.sipml5.org<br>Code source: http://code.google.com/p/sipml5/<br>I=
 hop it could help other developers.<br><br>Regards,</div></div></body></ht=
ml>
---888064777-1348475897-1336972478=:22604--

From magnus.westerlund@ericsson.com  Mon May 14 01:35:52 2012
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] New use-case proposed
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On 2012-05-11 18:21, Harald Alvestrand wrote:
> I propose to reject this use case because it requires yet another 
> security redesign.
> The clue is here:
> 
>>
>> - The keying solution must allow each participants to encrypt to
>> multiple receivers without any decryption+encryption in the middle node
>>
> 
> This means that each participant must use the same encryption keys to 
> all other participants; this in turn means that when someone leaves the 
> group, all participants must change their encryption keys; it also means 
> that as long as shared keys are used for authentication, all 
> participants can impersonate all other participants.

Actually EKT solves this in a very reasonable way. Each participant
announces his key using EKT. When a participant leaves all the other
participants uses EKT to provide a new key for the media they send. Thus
no need for synchronized key roll over anything.

When it comes to the security aspect this I would say it is part of the
model. SRTP uses symmetric keys, thus anyone can spoof anyone that are
inside the session and having access to the crypt context. In
centralized conferencing it is the central node that enforces the
relation between each other. That can be done both using independent
crypto contexts and having shared one.

And the arguments around the complexity is not that it on a per stream
individual basis is horrible expensive. It is the argument about needing
to have the functionality doing decryption and N-1 encryptions for each
packet being forwarded to all the others in a N number of participants
conference. The argument is to have no need for any media flow
encryption and decryption only be part of the key-management.

> 
> In fact, this solution has most of the issues (except for the network 
> layer deployment issue) that leads me to strongly advocate leaving 
> multicast out of scope for this effort.
> 

I am not arguing for including IP multicast support. The issue this use
case brings up is that one of RTP support for multi-party sessions. This
use case makes it clear that a particular level of RTP functionality
support is needed. I think almost the same is needed for centralized RTP
mixer topologies where the middle node uses its own SSRCs to forward
other participants media streams and transfer RTCP feedback messages
when not possible to handle in the mixer.

Another aspect that we need to bring up here is if there is going to be
any support for a WebRTC browser to receive a media stream from A and be
capable of forwarding it to B. Without the same features as needed for
this use case we must mandate that a browser doing this supports media
transcoding. Otherwise congestion control and media control aspects will
not be supported enabling media to be delivered. Transcoding obviously
gives a significant hit on quality unless the media bit-rate is
significantly higher than without transcoding.

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 pravindran@sonusnet.com  Mon May 14 01:39:28 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Cullen Jennings <fluffy@iii.ca>
Thread-Topic: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Cc: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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Cullen,

PRANSWER ISSUE:
In case PRANSWER is designed for interop with SIP forking with ICE handling=
,=20
then PRANSWER is not required because SIP (RFC 3261) never allows multiple
answer within the same dialog. In case of each forked response (18x/200 wit=
h=20
new To-Tag), SIP UA creates new dialog and those dialog has to be handled=20
separately. How grouping of those SIP dialogs works together is a implement=
ation
issue in SIP. So, there is no need of new O/A state (PRANSWER) just to make=
=20
SIP interop happen. Please let me know any other usecase which compels=20
PRANSWER state in JSEP.

FORKING ISSUE:
I agree with you that most of the deployed SIP end-point products does not=
=20
Support parallel forking and its next hop B2BUA (PBX, SBC, GW) takes care
of the handling. Having said that, some of the well deployed SIP entity in=
=20
service provider (SP) network is based on SIP forking which compels to=20
support forking in actual End-point or B2BUA. I know that it is possible
to design JSEP without forking support and interop with SIP is possible whi=
ch=20
will be equivalent to H.323 to SIP gateway development wherein H.323 never
supports forking. Now the question is whether JSEP API supports forking=20
or not. I'm lean towards forking because the same peerconnection cloning AP=
I=20
may extend for browser media mixing later stage. If you explain about my=20
mixing API idea is wrong, then I'll agree with you that cloning of peer=20
connection is not required.

Thanks
Partha     =20

>-----Original Message-----
>From: Cullen Jennings [mailto:fluffy@iii.ca]
>Sent: Friday, May 11, 2012 12:36 AM
>To: Ravindran, Parthasarathi
>Cc: Justin Uberti; Christer Holmberg; Randell Jesup; rtcweb@ietf.org
>Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in
>JSEP
>
>
>I'm a very confused on what type of forking (with ICE) use case people
>want to satisfy.
>
>I agree with you that PRANSWER does make things more complicated but I
>presented at several of the meetings about how things just don't work
>without it or something like it so I don't see how to avoid it.
>
>
>On May 9, 2012, at 12:34 AM, Ravindran, Parthasarathi wrote:
>
>> Hi Justin,
>>
>> Sorry for the delay in reply. There are two issues associated with
>this scenario:
>>
>> Issue 1) SIP Forking : This is solved by peerconnection cloning
>>
>> Issue 2)  UPDATE support within SIP early dialog in the non-forking
>scenario. Here, the cloning will not solve issue as the answerer
>originated the new offer during SDP_PRANSWER state . Issue 2 is the
>title of this mail thread.
>>
>> IMO, SDP_PRANSWER will complicate the offer/answer state machine.
>>
>> Thanks
>> Partha
>>
>>
>>
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Justin Uberti
>> Sent: Tuesday, May 08, 2012 8:40 PM
>> To: Christer Holmberg
>> Cc: Randell Jesup; rtcweb@ietf.org
>> Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in
>> JSEP
>>
>> The conclusion on this thread seems to be that the right way to
>address this problem is via PeerConnection cloning, and that no changes
>are needed to the JSEP state machine. I'll work with Richard to figure
>out how we should extend JSEP to support cloning.
>>
>> On Thu, May 3, 2012 at 5:16 PM, Christer Holmberg
><christer.holmberg@ericsson.com> wrote:
>> Hi,
>>
>> >>>>You could also choose not to alert the remote user until the ICE
>procedures have finished - more or less using ICE with preconditions.
>> >>>>
>> >>>> Of course, that is going to trigger actions in STUN/TURN servers,
>> >>>> even if the called party won't accept the call, but at least from
>a user experience perspective that is still better than lifting the
>hook, and having to wait for whatever-time-it-takes-for-ICE-to-finish
>seconds before one can start to talk.
>> >>>
>> >>> This also has a downside of disclosing user's IP to the caller
>without the user confirmation. For a lot of applications this can be
>serious security flaw.
>> >>
>> >> The client can still log when ICE procedures occur.
>> >>
>> >> Because, even if I wait until your phone starts to ring, most
>likely I would still get your IP address without user confirmation
>(speaking in SIP terms, phones normally don't ask for user permission
>before they send 18x with SDP), eventhough you would easier be made
>aware of that it happens.
>> >
>> > Another alternative is of course to do ICE with an SBC, and/or
>> > having an SBC doing ICE on behalf of you :)
>> >
>> >
>> > This is true for SIP but was as far as I know was specifically
>> > designed around in WebRTC. WebRTC signaling server acts as B2BUA so
>any type of ringing notification goes through the web server and does
>not need to carry any type of client address information. The client
>address information is only provided when SDP answer is sent or ICE
>negotiation is started.
>>
>> Assuming you are only going to make user confirmation (read: accept
>calls) in cases where you absolutely sure that the caller isn't someone
>trying to get your IP address.
>>
>> But, never the less, having a solution where I first have to give user
>confirmation, and then wait until I can start to talk, is probably
>something most people want to avoid. Depending upon, of course, how long
>the waiting time is.
>>
>> Regards,
>>
>> Christer
>> _______________________________________________
>> 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  Mon May 14 01:48:33 2012
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On 2012-05-14 09:49, Magnus Westerlund wrote:
> On 2012-05-11 18:21, Harald Alvestrand wrote:
>> I propose to reject this use case because it requires yet another 
>> security redesign.
>> The clue is here:
>>
>>>
>>> - The keying solution must allow each participants to encrypt to
>>> multiple receivers without any decryption+encryption in the middle node
>>>
>>
>> This means that each participant must use the same encryption keys to 
>> all other participants; this in turn means that when someone leaves the 
>> group, all participants must change their encryption keys; it also means 
>> that as long as shared keys are used for authentication, all 
>> participants can impersonate all other participants.
> 
> Actually EKT solves this in a very reasonable way. Each participant
> announces his key using EKT. When a participant leaves all the other
> participants uses EKT to provide a new key for the media they send. Thus
> no need for synchronized key roll over anything.

To make it clear, there are two levels of security here. A first one of
each participant changing keys after a participant has left and no
longer gets the traffic. If that is done without changing the EKT key
then that participant could decrypt the key, but then the attacker must
get to the media which no longer is being forwarded to it.

The second level which is recommend is to have the central node change
the EKT key by triggering DTLS based rekeying of the EKT. Then it is
impossible for the participant that left to eavesdrop on the conference.

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 tim@phonefromhere.com  Mon May 14 02:25:52 2012
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To: "Ejzak, Richard P (Richard)" <richard.ejzak@alcatel-lucent.com>
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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:

> Support for multiple registered contacts and (parallel) forking are =
pretty fundamental to SIP.  This has little to do with conferencing.  =
Some systems (not all) may support only serial forking or may serialize =
media interactions associated with parallel forking, but forking remains =
widely deployed and useful.


I have a feeling that it is mostly useful because SIP is managing both =
the signalling and the identity.=20
In most WebRTC cases, the real identity of the user will be established =
with a web protocol (perhaps OAuth).
The identity used in any browser based legacy signalling (e.g. SIP) will =
be a second-order derived or temporary identity.

In that case there is no need for SIP to use the same  temporary =
identity for both my phones, which (I think)
renders the forking case significantly less useful.

Tim.
=20=

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References: <387F9047F55E8C42850AD6B3A7A03C6C0E23B102@inba-mail01.sonusnet.com> <6F428EFD2B8C2F49A2FB1317291A76C1136029362A@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <4FA13816.6050003@alvestrand.no> <CAD5OKxsOAeTRdCgj2g4BY8maeG1n9nzCv29g8kaFPVZ4tf8C5w@mail.gmail.com> <4FA15898.1040204@alvestrand.no> <7F2072F1E0DE894DA4B517B93C6A05852C44001329@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxsbTSpMPg30P4DSN6UasCx6na43tpZ5yT2ct6SLMpd9xQ@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132C@ESESSCMS0356.eemea.ericsson.se> <6F428EFD2B8C2F49A2FB1317291A76C1136047B753@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400132F@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuy8_Ras3X-8tcY1qaLRTVJ-z-JuXQaf0NKaRwN_dU18Q@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C442AC0DA@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxuv7T6YrEXEeVECKbGtRg4feoRYTgvC57yqWXTBK43ypg@mail.gmail.com> <4FA2B447.7010504@jesup.org> <4FA2E264.4040207@jesup.org> <CAD5OKxuq14_DfSJuFEHq_ZEubbOYt7NFTcXWpLND8ggAr37QhA@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C44001339@ESESSCMS0356.eemea.ericsson.se> <CAD5OKxvFYy-9XAE5GG0n58xX2HKRX3o5qkTZ2o6rugakaM77Ew@mail.gmail.com> <7F2072F1E0DE894DA4B517B93C6A05852C4400133A@ESESSCMS0356.eemea.ericsson.se> <CAOJ7v-1Ue=q4MAh0csiDLvRFfMGaJ4vWrDxhir-S-OjSTsymaw@mail.gmail.com> <387F9047F55E8C42850AD6B3A7A03C6C160337F1@inba-mail01.sonusnet.com> <6E35F409-3B3F-4D92-91F2-451E0910CFDE@iii.ca> <CABkgnnVO5UyhHjiWVPsLYAoXSKPsfn6Fnb10hQr_6-kgq0FC3g@mail.gmail.com> <F0D1114C-0B6C-4AED-BDD3-4E3D6A219485@iii.ca> <CABkgnnWNLOb_xLUfOoZ5d7sKVsJDvjmNhmRqFcupO4S4p8Nrfw@mail.gmail.com> <219EC9F7-87B8-4E7B-8D6F-58A5B9773D29@iii.ca> <6F428EFD2B8C2F49A2FB1317291A76C1136D546819@USNAVSXCHMBSA1.ndc.alcatel-lucent.com> <BDEE1968-C29C-45D1-9D88-59723C63125C@phonefromhere.com>
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On Mon, May 14, 2012 at 5:25 AM, Tim Panton <tim@phonefromhere.com> wrote:

>
> On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:
>
> > Support for multiple registered contacts and (parallel) forking are
> pretty fundamental to SIP.  This has little to do with conferencing.  Some
> systems (not all) may support only serial forking or may serialize media
> interactions associated with parallel forking, but forking remains widely
> deployed and useful.
>
>
> I have a feeling that it is mostly useful because SIP is managing both the
> signalling and the identity.
> In most WebRTC cases, the real identity of the user will be established
> with a web protocol (perhaps OAuth).
> The identity used in any browser based legacy signalling (e.g. SIP) will
> be a second-order derived or temporary identity.
>
> In that case there is no need for SIP to use the same  temporary identity
> for both my phones, which (I think)
> renders the forking case significantly less useful.
>
>
I strongly disagree with this. I can still be calling somebody via, let's
say, their email address and end up talking to two people using two
different devices. I think there is always going to be the situation where
one identity (email address, support link on the web site) will map to
multiple physical end points. It is possible to resolve this identity to
end point mapping using some sort of HTTP signaling before media connection
is setup and then to generate offers for each end point, but this would
make signaling more complex, and would slow down the call connect process
by at least half the round trip time.
_____________
Roman Shpount

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<br clear=3D"all"><div class=3D"gmail_quote">On Mon, May 14, 2012 at 5:25 A=
M, Tim Panton <span dir=3D"ltr">&lt;<a href=3D"mailto:tim@phonefromhere.com=
" target=3D"_blank">tim@phonefromhere.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">
<div class=3D"im"><br>
On 11 May 2012, at 15:26, Ejzak, Richard P (Richard) wrote:<br>
<br>
&gt; Support for multiple registered contacts and (parallel) forking are pr=
etty fundamental to SIP. =A0This has little to do with conferencing. =A0Som=
e systems (not all) may support only serial forking or may serialize media =
interactions associated with parallel forking, but forking remains widely d=
eployed and useful.<br>

<br>
<br>
</div>I have a feeling that it is mostly useful because SIP is managing bot=
h the signalling and the identity.<br>
In most WebRTC cases, the real identity of the user will be established wit=
h a web protocol (perhaps OAuth).<br>
The identity used in any browser based legacy signalling (e.g. SIP) will be=
 a second-order derived or temporary identity.<br>
<br>
In that case there is no need for SIP to use the same =A0temporary identity=
 for both my phones, which (I think)<br>
renders the forking case significantly less useful.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"></font></span><br></blockquo=
te><div><br>I strongly disagree with this. I can still be calling somebody =
via, let&#39;s say, their email address and end up talking to two people us=
ing two different devices. I think there is always going to be the situatio=
n where one identity (email address, support link on the web site) will map=
 to multiple physical end points. It is possible to resolve this identity t=
o end point mapping using some sort of HTTP signaling before media connecti=
on is setup and then to generate offers for each end point, but this would =
make signaling more complex, and would slow down the call connect process b=
y at least half the round trip time.<br>
</div></div>_____________<br>Roman Shpount<br>

--047d7b1631bf3c779c04c0037d32--

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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 14 May 2012 11:45, Roman Shpount <roman@telurix.com> wrote:
> It is possible to resolve this identity to end
> point mapping using some sort of HTTP signaling

I think that this was the point.  Your application has some concept of
identity that maps to one or more registered endpoints.  It can work
out the signaling such that all devices get notified of a session
request without the need for this complex cloning mess.

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--047d7b15b017b124f704c00519e0
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On Mon, May 14, 2012 at 4:31 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 14 May 2012 11:45, Roman Shpount <roman@telurix.com> wrote:
> > It is possible to resolve this identity to end
> > point mapping using some sort of HTTP signaling
>
> I think that this was the point.  Your application has some concept of
> identity that maps to one or more registered endpoints.  It can work
> out the signaling such that all devices get notified of a session
> request without the need for this complex cloning mess.
>

I am not sure the cloning is a "complex mess". If ICE is the requirement
(so you know remote IP/port of each party) you can map each remote party to
the source IP/port. Then all you need to do is reference counting for local
sockets/TURN allocations, and you are done implementing it. PRANSWER would
probably require as much coding but would not solve all the problems that
cloning will.
_____________
Roman Shpount

--047d7b15b017b124f704c00519e0
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<div class=3D"gmail_quote">On Mon, May 14, 2012 at 4:31 PM, Martin Thomson =
<span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" target=3D=
"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">
<div class=3D"im">On 14 May 2012 11:45, Roman Shpount &lt;<a href=3D"mailto=
:roman@telurix.com">roman@telurix.com</a>&gt; wrote:<br>
&gt; It is possible to resolve this identity to end<br>
&gt; point mapping using some sort of HTTP signaling<br>
<br>
</div>I think that this was the point. =A0Your application has some concept=
 of<br>
identity that maps to one or more registered endpoints. =A0It can work<br>
out the signaling such that all devices get notified of a session<br>
request without the need for this complex cloning mess.<br>
</blockquote></div><br>I am not sure the cloning is a &quot;complex mess&qu=
ot;. If ICE is the requirement (so you know remote IP/port of each party) y=
ou can map each remote party to the source IP/port. Then all you need to do=
 is reference counting for local sockets/TURN allocations, and you are done=
 implementing it. PRANSWER would probably require as much coding but would =
not solve all the problems that cloning will.<br clear=3D"all">
_____________<br>Roman Shpount<br>
<br>

--047d7b15b017b124f704c00519e0--

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Subject: Re: [rtcweb] SDP_PRANSWER followed by SDP_OFFER scenario in JSEP
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On 5/14/2012 4:31 PM, Martin Thomson wrote:
> On 14 May 2012 11:45, Roman Shpount<roman@telurix.com>  wrote:
>> It is possible to resolve this identity to end
>> point mapping using some sort of HTTP signaling
>
> I think that this was the point.  Your application has some concept of
> identity that maps to one or more registered endpoints.  It can work
> out the signaling such that all devices get notified of a session
> request without the need for this complex cloning mess.

That isn't the point of cloning (getting notified).  It's the server in 
this case that knows about the multiple endpoints.  (For that matter, 
one could be a WebRTC client, and another could be a WebRTC gateway to 
PSTN, and a third could be a network-based answering service where a 
pickup from an actual endpoint would abort the playback/recording - 
another case of a forked ANSWER after a final ANSWER where cloning is 
useful.)

-- 
Randell Jesup
randell-ietf@jesup.org

From oscar.ohlsson@ericsson.com  Tue May 15 11:27:31 2012
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From: Oscar Ohlsson <oscar.ohlsson@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 15 May 2012 20:27:27 +0200
Thread-Topic: [rtcweb] New use-case proposed
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Subject: Re: [rtcweb] New use-case proposed
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Hi,

An interesting variant of Stefan's use case is when the central media=20
node is provided by an untrusted 3rd-party. The application provider and=20
the users trust the 3rd party to deliver the media as expected, but they=20
do not necessarily trust it with cleartext media, be that audio, video=20
or data.

The benefit is that the application provider can be less picky when=20
selecting the 3rd-party provider(s). Having a large pool to choose from=20
makes it easier to switch provider or select the cheapest or=20
geographically closest one.

         +---+      +------------+      +---+
         | A |<---->|            |<---->| B |
         +---+      | Translator |      +---+
         +---+      |(untrusted) |      +---+
         | C |<---->|            |<---->| D |
         +---+      +------------+      +---+


A requirement for this to work is that group keys are supported. This=20
allows the central media node to simply forward the media without=20
decrypting and re-encrypting it for each recipient. The fact that the=20
central media node becomes less complex and can achieve higher=20
throughput is of course beneficial, but it's not the main purpose.

Another requirement is that the group key can be distributed by the=20
application provider (or some other trusted party) without the central=20
media node learning its value. This can easily be achieved in=20
DTLS-SRTP + EKT by using the a=3Ddtls-srtp-host attribute. The host=20
specified in the attribute performs an initial DTLS-SRTP handshake with=20
each party and transmits the shared EKT key (i.e. the group key).

                    +------------+=20
                    | Key Server |
   |-----EKTkey---- | (trusted)  | ----EKTkey-----|
   |                |            |                |
   |                +------------+                |
   |                                              |
   |     +---+      +------------+      +---+     |
   |---> | A |<---->|            |<---->| B | <---|
   |     +---+      | Translator |      +---+     |
   |     +---+      |(untrusted) |      +---+     |
   |---> | C |<---->|            |<---->| D | <---|
         +---+      +------------+      +---+


Is the above something that this group would like to support or=20
investigate further?

Best regards,

Oscar Ohlsson

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Stefan Hakansson LK
> Sent: Friday, May 11, 2012 3:01 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] New use-case proposed
>=20
> I've been sketching on a new use-case. The use-case is=20
> intended to derive requirements that enable low complex=20
> central nodes for multi party sessions, which in turn leads=20
> to requirements regarding
> (essentially) keying solution for SRTP and the possibility=20
> for the app to control/set things in the media configuration=20
> (in other words, for JSEP):
>=20
>=20
> Multi-party with low complexity central node=20
> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>=20
> A geographically spread (charity, non-profit, school)=20
> organization offers its members multi-party video=20
> communication via a shared virtual room that participants can=20
> enter and leave at any time. At times there are many persons=20
> in the virtual room (20+), at other times very few (3-5, or even 0-1).
>=20
> The application will control if few participants (a subset)=20
> are shown at a higher fidelity or if many (all) are shown at=20
> a lower fidelity or a mix of some few at high fidelity and=20
> the rest at much lower fidelity given the existing bandwidth=20
> limitations between participants.
>=20
> Since there are at times many persons in the virtual room, it=20
> is not feasible to set up a mesh. The bandwidth needed by a=20
> participant exceeds what many members have access to,=20
> especially in their uplinks, so instead a central media node=20
> is deployed. In addition, the organization does not have=20
> access to much funds, but has to rely on cheap hardware=20
> (often donated) operated by volunteers.
>=20
>   From this use-case a high level requirement saying something like
>=20
> "It must be possible to set up media streams and encryption=20
> in such a way that processing in a central node is minimized=20
> (no transcoding required, no RTP packet rewriting, no=20
> decryption/re-encryption for every outgoing flow - just=20
> simple forwarding)."
>=20
> can be derived. This can in turn be broken down into more=20
> detailed requirements such as:
>=20
> - The keying solution must allow each participants to encrypt=20
> to multiple receivers without any decryption+encryption in=20
> the middle node
>=20
> - RTP sessions that have SSRCs from multiple PeerConnections=20
> being interconneted must be supported. From a given end-point=20
> all SSRCs will come over one PC, but the full path will be=20
> different towards different sets of SSRCs.
>=20
> - Signalling must be capable of establishing a common set of=20
> receiver configurations over all participants.
>=20
> - The API must allow for the above, i.e.:
> -- The app must be able to chose a keying solution that=20
> enables encryption to multiple receivers (if the app can have=20
> any control over keying method used)
> -- The app must be able to control SSRCs to use for outgoing streams
> -- The app must be able to control the receiver configuration=20
> the browser uses
>=20
> Is this something we should consider adding to the use-case document?
>=20
> Br,
> Stefan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From Milan.Young@nuance.com  Tue May 15 16:22:12 2012
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From: "Young, Milan" <Milan.Young@nuance.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: TCP vs UDP for media
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Subject: [rtcweb] TCP vs UDP for media
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Most of the use cases within WebRTC center around human to human communicat=
ion, and for such cases RTP transport makes sense.  But there are at least =
a few use cases where one of the endpoints could be a machine.  Since machi=
nes have the ability to buffer and take a quick nap on the job, the "realti=
me" requirement isn't as critical.  Often, in such cases, the more importan=
t consideration is whether all of the media was transmitted.

I'm relatively new to WebRTC, so I apologize if I'm rehashing settled issue=
s.  But I'm wondering if any thought has been given to TCP as a media trans=
port.  If framing and timeouts could be worked out, would other folks besid=
es myself find this useful?
=20
Thanks


From giles@thaumas.net  Tue May 15 16:38:35 2012
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Subject: Re: [rtcweb] TCP vs UDP for media
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On 12-05-15 4:22 PM, Young, Milan wrote:

> I'm wondering if any thought has been given to TCP as a media transport.

Where low latency transmission isn't an issue, one can generally fall
back to established TCP-based protocols, like HTTP streaming and
websockets, so we haven't really worried about that angle in the context
of webrtc.

Recording the stream is a requirement, so probably something could be
built using that, merging recordings from each endpoint to fix up any
dropped packets.

 -r

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Subject: Re: [rtcweb] TCP vs UDP for media
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On Tue, May 15, 2012 at 4:38 PM, Ralph Giles <giles@thaumas.net> wrote:
> On 12-05-15 4:22 PM, Young, Milan wrote:
>
>> I'm wondering if any thought has been given to TCP as a media transport.
>
> Where low latency transmission isn't an issue, one can generally fall
> back to established TCP-based protocols, like HTTP streaming and
> websockets, so we haven't really worried about that angle in the context
> of webrtc.
>
> Recording the stream is a requirement, so probably something could be
> built using that, merging recordings from each endpoint to fix up any
> dropped packets.

Agreed. This seems like something that could be done using just
getUserMedia() plus existing Web technologies.

-Ekr

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Cone and symmetric nat are probably not what exactly what we mean here - =
perhaps just refer to it as the various types of NATs described in RFC =
4787.=20

Might want to add to some use case that when A calls B, and B does not =
reveal their IP address to A.

Like to add Alice calls Bob with an encrypted media. Bob can tell the =
that the encrypted media is from Alice and not from an MITM

Like to add ability to make a call in a situation where even small =
sounds from microphone should be sent to the other side and not filtered =
out. Emergency calls are examples of this as are some games and "jam =
session"

Like to add case where application has one two video streams but one is =
far more important than the other. Should be a way to make sure that =
preferential treatment is given the the important stream over the less =
important streams.=20

In F22, "the IVR" -> "an DTMF based IVR"

I have a hard time deciding what level the requirements should be at. It =
seems likely that given the level they are currently at, the =
requirements are somewhat incomplete. As long as we treat these as =
partial set of requirements derived from the use cases, I'm fine with =
that.=20


New use case:

Imagine a service like webex that facilitates collaboration between two =
companies called A and B. Webex has no need to see the content of the =
communication from A to B. So webex might run the web server that helps =
set up a RTCWeb session between A and B, but webex needs to be able to =
write it's code such that it does not get the keying material used to =
encrypt the media from A to B and the companies need to be able to =
verify that webex did not get the keys. Webex has no need to see the =
media from A to B and if it can provide this sort of promise that it =
does not get the media, it makes it much easier for a customer (say =
Juniper) to use webex and know the contents of their calls are not being =
sold or used by a competitor. Given how much WebRTC will be used by =
cloud services, this is an important feature. When installing the =
"webex" application in the browser and granting permissions to the =
application, it would be good to have one of the permissions be if the =
JS got access to the media or keying material for it to enforce this =
promise.=20

Note this use case is not phrase to say this is the only way it can =
work. This is not mutually exclusive with they type of use cases =
described by Skype folks where there is an explicitly desire to make =
sure the calling service does have the keys to decrypt the media.=20








From Milan.Young@nuance.com  Tue May 15 18:37:58 2012
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From: "Young, Milan" <Milan.Young@nuance.com>
To: Eric Rescorla <ekr@rtfm.com>, Ralph Giles <giles@thaumas.net>
Thread-Topic: [rtcweb] TCP vs UDP for media
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Yes, existing web technologies for transport would work fine.  In fact I wr=
ote a demo based on WebSockets, the AudioAPI, and getUserMedia.  But it's a=
 bit cludgy and only transmits PCM audio.

Would my use case for live access to an encoded media stream be a good fit =
for a revised MediaStreamRecorder?  Would this group the right place to hos=
t such an effort?

 Thanks

-----Original Message-----
From: Eric Rescorla [mailto:ekr@rtfm.com]=20
Sent: Tuesday, May 15, 2012 4:47 PM
To: Ralph Giles
Cc: Young, Milan; rtcweb@ietf.org
Subject: Re: [rtcweb] TCP vs UDP for media

On Tue, May 15, 2012 at 4:38 PM, Ralph Giles <giles@thaumas.net> wrote:
> On 12-05-15 4:22 PM, Young, Milan wrote:
>
>> I'm wondering if any thought has been given to TCP as a media transport.
>
> Where low latency transmission isn't an issue, one can generally fall=20
> back to established TCP-based protocols, like HTTP streaming and=20
> websockets, so we haven't really worried about that angle in the=20
> context of webrtc.
>
> Recording the stream is a requirement, so probably something could be=20
> built using that, merging recordings from each endpoint to fix up any=20
> dropped packets.

Agreed. This seems like something that could be done using just
getUserMedia() plus existing Web technologies.

-Ekr

From magnus.westerlund@ericsson.com  Tue May 15 23:44:01 2012
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On 2012-05-16 03:37, Young, Milan wrote:
> Yes, existing web technologies for transport would work fine.  In
> fact I wrote a demo based on WebSockets, the AudioAPI, and
> getUserMedia.  But it's a bit cludgy and only transmits PCM audio.
> 
> Would my use case for live access to an encoded media stream be a
> good fit for a revised MediaStreamRecorder?  Would this group the
> right place to host such an effort?

I think this is something you should take to the W3C. As it appear to be
primarily an API question. Getting access to the content in the JS
application or request the browser to record to a local file which later
can be uploaded it would not be in this WG.

Cheers

Magnus Westerlund
WG chair

> 
> Thanks
> 
> -----Original Message----- From: Eric Rescorla [mailto:ekr@rtfm.com]
>  Sent: Tuesday, May 15, 2012 4:47 PM To: Ralph Giles Cc: Young,
> Milan; rtcweb@ietf.org Subject: Re: [rtcweb] TCP vs UDP for media
> 
> On Tue, May 15, 2012 at 4:38 PM, Ralph Giles <giles@thaumas.net>
> wrote:
>> On 12-05-15 4:22 PM, Young, Milan wrote:
>> 
>>> I'm wondering if any thought has been given to TCP as a media
>>> transport.
>> 
>> Where low latency transmission isn't an issue, one can generally
>> fall back to established TCP-based protocols, like HTTP streaming
>> and websockets, so we haven't really worried about that angle in
>> the context of webrtc.
>> 
>> Recording the stream is a requirement, so probably something could
>> be built using that, merging recordings from each endpoint to fix
>> up any dropped packets.
> 
> Agreed. This seems like something that could be done using just 
> getUserMedia() plus existing Web technologies.
> 
> -Ekr _______________________________________________ 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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WG,

This is an individual contribution on how to do codec control in WebRTC.

Ericsson has an current IPR statment on the referenced Codec Operation
Points RTCP extension. This statement will be updated, hopefully next
week, with new terms making it a non-asssert statement for
implementation in webbrowsers and FRAND for other usages. I will send a
pointer when the official statement is available.

Cheers

Magnus

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.

	Title           : Codec Control for WebRTC
	Author(s)       : Magnus Westerlund
                          Bo Burman
	Filename        : draft-westerlund-rtcweb-codec-control-00.txt
	Pages           : 18
	Date            : 2012-05-16

   This document proposes how WebRTC should handle media codec control
   between peers.  With media codec control we mean such parameters as
   video resolution and frame-rate.  This includes both initial
   establishment of capabilities using the SDP based JSEP signalling and
   during ongoing real-time interactive sessions in response to user and
   application events.  The solution uses SDP for initial boundary
   establishment that are rarely, if ever changed.  During the session
   the RTCP based Codec Operations Point (COP) signaling solution is
   used for dynamic control of parameters enabling timely and responsive
   controls.


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-westerlund-rtcweb-codec-control-0=
0.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-westerlund-rtcweb-codec-control-00=
.txt

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

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From martin.thomson@gmail.com  Wed May 16 11:56:17 2012
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From: Martin Thomson <martin.thomson@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Relaying for privacy
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On 15 May 2012 18:31, Cullen Jennings <fluffy@cisco.com> wrote:
> Might want to add to some use case that when A calls B, and B does not reveal their IP address to A.

I agree, but it's questionable whether this is something we
(ietf/w3c...browsers) need to solve the problem or whether we can rely
on sites respecting privacy appropriately.

Obviously, it's quite simple to allocate a relay for all new sessions
in the browser.  The hard part is knowing when it is OK to reveal
host/reflexive candidates so that the call can proceed without
relaying.

 - Is it when the user grants permission for the site to use
microphone/camera?  This seems wrong.  An individual desire for
privacy is often based on who is asking, which in this case is A, not
the site.  It's plausible that the site can be given permission to
access mic/camera prior to the call arriving.

 - Is it when the user indicates acceptance of the call?  I don't
think that the plan is to signal acceptance through chrome, so this
would have to rely on cues from the site, such as calling
createAnswer.  At that point, if the site wants the media off the
relay, they now have an incentive to fake acceptance.  The same
applies to relying on any other cues from the site, including faking
the _initiation_ of the call, which can be done from both A and B.

 - Is there special chrome that enables this?  Adding more chrome
doesn't sound like a great idea to me, but maybe this behaviour could
be tied to the various "private/incognito mode" options in the
browser.

I did have another related thought, but distractions erased it.

From juberti@google.com  Wed May 16 14:54:45 2012
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From: Justin Uberti <juberti@google.com>
Date: Wed, 16 May 2012 17:54:17 -0400
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Relaying for privacy
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--20cf3005dcc896566a04c02e5df6
Content-Type: text/plain; charset=UTF-8

My view was that we need to make it _possible_ for sites to respect privacy
(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.

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 forcing
use of relay is even the right thing to do.

On Wed, May 16, 2012 at 2:56 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 15 May 2012 18:31, Cullen Jennings <fluffy@cisco.com> wrote:
> > Might want to add to some use case that when A calls B, and B does not
> reveal their IP address to A.
>
> I agree, but it's questionable whether this is something we
> (ietf/w3c...browsers) need to solve the problem or whether we can rely
> on sites respecting privacy appropriately.


> Obviously, it's quite simple to allocate a relay for all new sessions
> in the browser.  The hard part is knowing when it is OK to reveal
> host/reflexive candidates so that the call can proceed without
> relaying.
>
>  - Is it when the user grants permission for the site to use
> microphone/camera?  This seems wrong.  An individual desire for
> privacy is often based on who is asking, which in this case is A, not
> the site.  It's plausible that the site can be given permission to
> access mic/camera prior to the call arriving.
>
>  - Is it when the user indicates acceptance of the call?  I don't
> think that the plan is to signal acceptance through chrome, so this
> would have to rely on cues from the site, such as calling
> createAnswer.  At that point, if the site wants the media off the
> relay, they now have an incentive to fake acceptance.  The same
> applies to relying on any other cues from the site, including faking
> the _initiation_ of the call, which can be done from both A and B.
>
>  - Is there special chrome that enables this?  Adding more chrome
> doesn't sound like a great idea to me, but maybe this behaviour could
> be tied to the various "private/incognito mode" options in the
> browser.
>
> I did have another related thought, but distractions erased it.
>

--20cf3005dcc896566a04c02e5df6
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

My view was that we need to make it _possible_ for sites to respect privacy=
 (via the API), and provide documentation that encourages sites to do so. A=
s you point out, I think it&#39;s difficult for the browser to automaticall=
y figure out what to do here, and when to do it.<div>

<br></div><div>The identity of the &quot;called&quot; user may not be a fac=
tor in certain applications (e.g. some p2p game), making it unclear as to w=
hether forcing use of relay is even the right thing to do.<br><br><div clas=
s=3D"gmail_quote">

On Wed, May 16, 2012 at 2:56 PM, Martin Thomson <span dir=3D"ltr">&lt;<a hr=
ef=3D"mailto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gma=
il.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"=
margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

On 15 May 2012 18:31, Cullen Jennings &lt;<a href=3D"mailto:fluffy@cisco.co=
m">fluffy@cisco.com</a>&gt; wrote:<br>
&gt; Might want to add to some use case that when A calls B, and B does not=
 reveal their IP address to A.<br>
<br>
I agree, but it&#39;s questionable whether this is something we<br>
(ietf/w3c...browsers) need to solve the problem or whether we can rely<br>
on sites respecting privacy appropriately.</blockquote><blockquote class=3D=
"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding=
-left:1ex">
<br>
Obviously, it&#39;s quite simple to allocate a relay for all new sessions<b=
r>
in the browser. =C2=A0The hard part is knowing when it is OK to reveal<br>
host/reflexive candidates so that the call can proceed without<br>
relaying.<br>
<br>
=C2=A0- Is it when the user grants permission for the site to use<br>
microphone/camera? =C2=A0This seems wrong. =C2=A0An individual desire for<b=
r>
privacy is often based on who is asking, which in this case is A, not<br>
the site. =C2=A0It&#39;s plausible that the site can be given permission to=
<br>
access mic/camera prior to the call arriving.<br>
<br>
=C2=A0- Is it when the user indicates acceptance of the call? =C2=A0I don&#=
39;t<br>
think that the plan is to signal acceptance through chrome, so this<br>
would have to rely on cues from the site, such as calling<br>
createAnswer. =C2=A0At that point, if the site wants the media off the<br>
relay, they now have an incentive to fake acceptance. =C2=A0The same<br>
applies to relying on any other cues from the site, including faking<br>
the _initiation_ of the call, which can be done from both A and B.<br>
<br>
=C2=A0- Is there special chrome that enables this? =C2=A0Adding more chrome=
<br>
doesn&#39;t sound like a great idea to me, but maybe this behaviour could<b=
r>
be tied to the various &quot;private/incognito mode&quot; options in the<br=
>
browser.<br>
<br>
I did have another related thought, but distractions erased it.<br></blockq=
uote></div></div>

--20cf3005dcc896566a04c02e5df6--

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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 16 May 2012 15:09:43 -0700
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Subject: Re: [rtcweb] Relaying for privacy
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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 privacy
> (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.
>
> 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 forcing
> use of relay is even the right thing to do.

I concur with all this.

I think it's also important to reiterate that this is about the site cooperating
to protect your privacy. If you want to protect the user's IP address from 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).

-Ekr

From ekr@rtfm.com  Wed May 16 19:07:21 2012
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From: Eric Rescorla <ekr@rtfm.com>
Date: Wed, 16 May 2012 19:07:15 -0700
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Yes. Tor is too slow to do voice effectively. I would expect torbutton or a t=
orbutton-like to route media through a turn server. This will provide worse p=
rivacy than onion routing but better than just letting the server control th=
e routing.

Ekr

On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli <denglingl=
i@chinamobile.com> wrote:

> Hi, Ekr
>=20
> I agree with you about the Martin's comments.=20
> For the proposal of Torbutton, I have a concern about the employment of
> union routing (encrypting and decrypting several times between the relayin=
g
> nodes along the way) in Tor would be not acceptable to the RTC requirement=
s
> of VoIP use-cases.
> Would this be an issue?
>=20
> BR
> Lingli
>=20
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org [mailto:rtcweb-bounce=
s@ietf.org] =E4=BB=A3=E8=A1=A8 Eric
> Rescorla
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 6=
:10
> =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti
> =E6=8A=84=E9=80=81: rtcweb@ietf.org
> =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy
>=20
> 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=20
>> privacy (via the API), and provide documentation that encourages sites=20=

>> to do so. As you point out, I think it's difficult for the browser to=20
>> 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=20
>> applications (e.g. some p2p game), making it unclear as to whether=20
>> forcing 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
> cooperating to protect your privacy. If you want to protect the user's IP
> address from 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> __________ Information from ESET NOD32 Antivirus, version of virus signatu=
re
> database 7138 (20120515) __________
>=20
> The message was checked by ESET NOD32 Antivirus.
>=20
> http://www.eset.com
>=20
>=20
>=20

From denglingli@chinamobile.com  Wed May 16 22:54:05 2012
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It makes sense to me.
However, considering the relaying burden would not be trivial, I'd =
further prefer that the enable/disable button be dedicatedly set for the =
RTCWEB instance in question, rather than the whole pack of connections =
of the browsers.=20
In other words, it is better to be triggered at the time of page-loading =
for the RTCWEB app for a global configuration (keeping current IP =
privacy for all callers) or when called by for a specific configuration =
(keeping privacy for the incoming caller).=20
In either cases, the setting can be done through getusermedia as Martin =
said.

BR
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Eric Rescorla [mailto:ekr@rtfm.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 =
10:07
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: Justin Uberti; <rtcweb@ietf.org>
=E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Relaying for =
privacy

Yes. Tor is too slow to do voice effectively. I would expect torbutton =
or a torbutton-like to route media through a turn server. This will =
provide worse privacy than onion routing but better than just letting =
the server control the routing.

Ekr

On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli =
<denglingli@chinamobile.com> wrote:

> Hi, Ekr
>=20
> I agree with you about the Martin's comments.=20
> For the proposal of Torbutton, I have a concern about the employment=20
> of union routing (encrypting and decrypting several times between the=20
> relaying nodes along the way) in Tor would be not acceptable to the=20
> RTC requirements of VoIP use-cases.
> Would this be an issue?
>=20
> BR
> Lingli
>=20
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 Eric=20
> Rescorla
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2012=E5=B9=B45=E6=9C=8817=E6=97=A5 6:10
> =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti
> =E6=8A=84=E9=80=81: rtcweb@ietf.org
> =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy
>=20
> 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=20
>> privacy (via the API), and provide documentation that encourages=20
>> sites to do so. As you point out, I think it's difficult for the=20
>> 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=20
>> applications (e.g. some p2p game), making it unclear as to whether=20
>> forcing 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=20
> cooperating to protect your privacy. If you want to protect the user's =

> IP address from 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=20
> (though likely that can't go through Tor for performance reasons).
>=20
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> __________ Information from ESET NOD32 Antivirus, version of virus=20
> signature database 7138 (20120515) __________
>=20
> The message was checked by ESET NOD32 Antivirus.
>=20
> http://www.eset.com
>=20
>=20
>=20

__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




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From: =?UTF-8?B?6YKT54G16I6JL2RlbmdsaW5nbGk=?= <denglingli@chinamobile.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>	<003101cd33c9$593793b0$0ba6bb10$@chinamobile.com>	<B7AB7EC0-35B0-46BA-8786-6AD92A9AD168@rtfm.com> <007b01cd33f1$03ccc9e0$0b665da0$@chinamobile.com>
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Date: Thu, 17 May 2012 13:56:38 +0800
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Cc: rtcweb@ietf.org
Subject: [rtcweb] =?utf-8?b?562U5aSNOiAg562U5aSNOiDnrZTlpI06ICBSZWxheWlu?= =?utf-8?q?g_for_privacy?=
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Sorry, by " In either cases, the setting can be done through =
getusermedia as Martin said." I mean " In either cases, the setting =
could be done through getusermedia as Martin said."
Since according to our experience with chrome so far, media relaying =
through turn is not yet supported.

Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 =
=E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 =
13:51
=E6=94=B6=E4=BB=B6=E4=BA=BA: 'Eric Rescorla'
=E6=8A=84=E9=80=81: rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: [rtcweb] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: =
Relaying for privacy

It makes sense to me.
However, considering the relaying burden would not be trivial, I'd =
further prefer that the enable/disable button be dedicatedly set for the =
RTCWEB instance in question, rather than the whole pack of connections =
of the browsers.=20
In other words, it is better to be triggered at the time of page-loading =
for the RTCWEB app for a global configuration (keeping current IP =
privacy for all callers) or when called by for a specific configuration =
(keeping privacy for the incoming caller).=20
In either cases, the setting can be done through getusermedia as Martin =
said.

BR
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Eric Rescorla [mailto:ekr@rtfm.com]
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 =
10:07
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: Justin Uberti; <rtcweb@ietf.org>
=E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Relaying for =
privacy

Yes. Tor is too slow to do voice effectively. I would expect torbutton =
or a torbutton-like to route media through a turn server. This will =
provide worse privacy than onion routing but better than just letting =
the server control the routing.

Ekr

On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli =
<denglingli@chinamobile.com> wrote:

> Hi, Ekr
>=20
> I agree with you about the Martin's comments.=20
> For the proposal of Torbutton, I have a concern about the employment=20
> of union routing (encrypting and decrypting several times between the=20
> relaying nodes along the way) in Tor would be not acceptable to the=20
> RTC requirements of VoIP use-cases.
> Would this be an issue?
>=20
> BR
> Lingli
>=20
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 Eric=20
> Rescorla
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2012=E5=B9=B45=E6=9C=8817=E6=97=A5 6:10
> =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti
> =E6=8A=84=E9=80=81: rtcweb@ietf.org
> =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy
>=20
> 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=20
>> privacy (via the API), and provide documentation that encourages=20
>> sites to do so. As you point out, I think it's difficult for the=20
>> 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=20
>> applications (e.g. some p2p game), making it unclear as to whether=20
>> forcing 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=20
> cooperating to protect your privacy. If you want to protect the user's =

> IP address from 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=20
> (though likely that can't go through Tor for performance reasons).
>=20
> -Ekr
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> __________ Information from ESET NOD32 Antivirus, version of virus=20
> signature database 7138 (20120515) __________
>=20
> The message was checked by ESET NOD32 Antivirus.
>=20
> http://www.eset.com
>=20
>=20
>=20

__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com



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


From martin.thomson@gmail.com  Thu May 17 18:05:26 2012
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References: <CABkgnnWEqqAD0rPrFK3mj5jBCoqmyPTY_PFgVyg8t84egN0Baw@mail.gmail.com> <CAOJ7v-2K9VS0YKo=RJSPLcmT=X-kTQ3ynyp9APin92eW=LceTg@mail.gmail.com> <CABcZeBPDJ6MQgQFBW4+Gnq9V-Oo-Gb9d_7nuCnySzbFBzDoXVg@mail.gmail.com> <003101cd33c9$593793b0$0ba6bb10$@chinamobile.com> <B7AB7EC0-35B0-46BA-8786-6AD92A9AD168@rtfm.com> <007b01cd33f1$03ccc9e0$0b665da0$@chinamobile.com> <007c01cd33f1$d2dc4bc0$7894e340$@chinamobile.com>
Date: Thu, 17 May 2012 18:05:23 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
To: =?UTF-8?B?6YKT54G16I6JL2RlbmdsaW5nbGk=?= <denglingli@chinamobile.com>
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Subject: Re: [rtcweb] =?utf-8?b?562U5aSNOiDnrZTlpI06IOetlOWkjTogUmVsYXlpbmcg?= =?utf-8?q?for_privacy?=
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--e89a8f83a703a9b22004c04525ce
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I do not think that any option other than what Eric described is truly
feasible. That is, of the options provided, only the one where the site has
to be trusted bit to reveal your IP is likely to succeed.

getUserMedia provides a cue of a fundamentally different nature to that you
are looking for in this use case. getUserMedia signals that the user is ok
with the site gaining access to their camera/microphone.  That is a
contract between the user and the site, not the two users.

If you don't trust the site, then you are in Torbutton land. That is,
specific UI for this.

I've concluded that we don't need this requirement. There is little that a
browser can sensibly do beyond something like Torbutton.

And I'm not convinced that trying to replicate, or require the replication
of Torbutton, is a great idea either. Tor users understand the value it
provides, most other users won't care for the added complication.

On May 16, 2012 10:59 PM, "=E9=82=93=E7=81=B5=E8=8E=89/denglingli" <denglin=
gli@chinamobile.com>
wrote:
>
> Sorry, by " In either cases, the setting can be done through getusermedia
as Martin said." I mean " In either cases, the setting could be done
through getusermedia as Martin said."
> Since according to our experience with chrome so far, media relaying
through turn is not yet supported.
>
> Lingli
>
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org [mailto:rtcweb-bounc=
es@ietf.org] =E4=BB=A3=E8=A1=A8
=E9=82=93=E7=81=B5=E8=8E=89/denglingli
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 =
13:51
> =E6=94=B6=E4=BB=B6=E4=BA=BA: 'Eric Rescorla'
> =E6=8A=84=E9=80=81: rtcweb@ietf.org
> =E4=B8=BB=E9=A2=98: [rtcweb] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: Rela=
ying for privacy
>
> It makes sense to me.
> However, considering the relaying burden would not be trivial, I'd
further prefer that the enable/disable button be dedicatedly set for the
RTCWEB instance in question, rather than the whole pack of connections of
the browsers.
> In other words, it is better to be triggered at the time of page-loading
for the RTCWEB app for a global configuration (keeping current IP privacy
for all callers) or when called by for a specific configuration (keeping
privacy for the incoming caller).
> In either cases, the setting can be done through getusermedia as Martin
said.
>
> BR
> Lingli
>
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Eric Rescorla [mailto:ekr@rtfm.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=A5 =
10:07
> =E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
> =E6=8A=84=E9=80=81: Justin Uberti; <rtcweb@ietf.org>
> =E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Relaying for privacy
>
> Yes. Tor is too slow to do voice effectively. I would expect torbutton or
a torbutton-like to route media through a turn server. This will provide
worse privacy than onion routing but better than just letting the server
control the routing.
>
> Ekr
>
> On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli <dengli=
ngli@chinamobile.com>
wrote:
>
> > Hi, Ekr
> >
> > I agree with you about the Martin's comments.
> > For the proposal of Torbutton, I have a concern about the employment
> > of union routing (encrypting and decrypting several times between the
> > relaying nodes along the way) in Tor would be not acceptable to the
> > RTC requirements of VoIP use-cases.
> > Would this be an issue?
> >
> > BR
> > Lingli
> >
> > -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> > =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org [mailto:rtcweb-bou=
nces@ietf.org] =E4=BB=A3=E8=A1=A8 Eric
> > Rescorla
> > =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=
=A5 6:10
> > =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti
> > =E6=8A=84=E9=80=81: rtcweb@ietf.org
> > =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy
> >
> > 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
> >> privacy (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=
.
> >>
> >> 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
> >> forcing use of relay is even the right thing to do.
> >
> > I concur with all this.
> >
> > I think it's also important to reiterate that this is about the site
> > cooperating to protect your privacy. If you want to protect the user's
> > IP address from 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).
> >
> > -Ekr
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> > signature database 7138 (20120515) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> >
> >
>
> __________ Information from ESET NOD32 Antivirus, version of virus
signature database 7138 (20120515) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.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

--e89a8f83a703a9b22004c04525ce
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<p>I do not think that any option other than what Eric described is truly f=
easible. That is, of the options provided, only the one where the site has =
to be trusted bit to reveal your IP is likely to succeed.</p>
<p>getUserMedia provides a cue of a fundamentally different nature to that =
you are looking for in this use case. getUserMedia signals that the user is=
 ok with the site gaining access to their camera/microphone.=C2=A0 That is =
a contract between the user and the site, not the two users.</p>

<p>If you don&#39;t trust the site, then you are in Torbutton land. That is=
, specific UI for this.</p>
<p>I&#39;ve concluded that we don&#39;t need this requirement. There is lit=
tle that a browser can sensibly do beyond something like Torbutton.</p>
<p>And I&#39;m not convinced that trying to replicate, or require the repli=
cation of Torbutton, is a great idea either. Tor users understand the value=
 it provides, most other users won&#39;t care for the added complication.</=
p>

<p>On May 16, 2012 10:59 PM, &quot;=E9=82=93=E7=81=B5=E8=8E=89/denglingli&q=
uot; &lt;<a href=3D"mailto:denglingli@chinamobile.com">denglingli@chinamobi=
le.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Sorry, by &quot; In either cases, the setting can be done through getu=
sermedia as Martin said.&quot; I mean &quot; In either cases, the setting c=
ould be done through getusermedia as Martin said.&quot;<br>
&gt; Since according to our experience with chrome so far, media relaying t=
hrough turn is not yet supported.<br>
&gt;<br>
&gt; Lingli<br>
&gt;<br>
&gt; -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----<br>
&gt; =E5=8F=91=E4=BB=B6=E4=BA=BA: <a href=3D"mailto:rtcweb-bounces@ietf.org=
">rtcweb-bounces@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces@ietf=
.org">rtcweb-bounces@ietf.org</a>] =E4=BB=A3=E8=A1=A8 =E9=82=93=E7=81=B5=E8=
=8E=89/denglingli<br>
&gt; =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=
=A5 13:51<br>
&gt; =E6=94=B6=E4=BB=B6=E4=BA=BA: &#39;Eric Rescorla&#39;<br>
&gt; =E6=8A=84=E9=80=81: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org=
</a><br>
&gt; =E4=B8=BB=E9=A2=98: [rtcweb] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: R=
elaying for privacy<br>
&gt;<br>
&gt; It makes sense to me.<br>
&gt; However, considering the relaying burden would not be trivial, I&#39;d=
 further prefer that the enable/disable button be dedicatedly set for the R=
TCWEB instance in question, rather than the whole pack of connections of th=
e browsers.<br>

&gt; In other words, it is better to be triggered at the time of page-loadi=
ng for the RTCWEB app for a global configuration (keeping current IP privac=
y for all callers) or when called by for a specific configuration (keeping =
privacy for the incoming caller).<br>

&gt; In either cases, the setting can be done through getusermedia as Marti=
n said.<br>
&gt;<br>
&gt; BR<br>
&gt; Lingli<br>
&gt;<br>
&gt; -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----<br>
&gt; =E5=8F=91=E4=BB=B6=E4=BA=BA: Eric Rescorla [mailto:<a href=3D"mailto:e=
kr@rtfm.com">ekr@rtfm.com</a>]<br>
&gt; =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=E6=97=
=A5 10:07<br>
&gt; =E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli<br=
>
&gt; =E6=8A=84=E9=80=81: Justin Uberti; &lt;<a href=3D"mailto:rtcweb@ietf.o=
rg">rtcweb@ietf.org</a>&gt;<br>
&gt; =E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Relaying for priv=
acy<br>
&gt;<br>
&gt; Yes. Tor is too slow to do voice effectively. I would expect torbutton=
 or a torbutton-like to route media through a turn server. This will provid=
e worse privacy than onion routing but better than just letting the server =
control the routing.<br>

&gt;<br>
&gt; Ekr<br>
&gt;<br>
&gt; On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli &lt;=
<a href=3D"mailto:denglingli@chinamobile.com">denglingli@chinamobile.com</a=
>&gt; wrote:<br>
&gt;<br>
&gt; &gt; Hi, Ekr<br>
&gt; &gt;<br>
&gt; &gt; I agree with you about the Martin&#39;s comments.<br>
&gt; &gt; For the proposal of Torbutton, I have a concern about the employm=
ent<br>
&gt; &gt; of union routing (encrypting and decrypting several times between=
 the<br>
&gt; &gt; relaying nodes along the way) in Tor would be not acceptable to t=
he<br>
&gt; &gt; RTC requirements of VoIP use-cases.<br>
&gt; &gt; Would this be an issue?<br>
&gt; &gt;<br>
&gt; &gt; BR<br>
&gt; &gt; Lingli<br>
&gt; &gt;<br>
&gt; &gt; -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----<br>
&gt; &gt; =E5=8F=91=E4=BB=B6=E4=BA=BA: <a href=3D"mailto:rtcweb-bounces@iet=
f.org">rtcweb-bounces@ietf.org</a> [mailto:<a href=3D"mailto:rtcweb-bounces=
@ietf.org">rtcweb-bounces@ietf.org</a>] =E4=BB=A3=E8=A1=A8 Eric<br>
&gt; &gt; Rescorla<br>
&gt; &gt; =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8817=
=E6=97=A5 6:10<br>
&gt; &gt; =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti<br>
&gt; &gt; =E6=8A=84=E9=80=81: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@iet=
f.org</a><br>
&gt; &gt; =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy<br>
&gt; &gt;<br>
&gt; &gt; On Wed, May 16, 2012 at 2:54 PM, Justin Uberti &lt;<a href=3D"mai=
lto:juberti@google.com">juberti@google.com</a>&gt; wrote:<br>
&gt; &gt;&gt; My view was that we need to make it _possible_ for sites to r=
espect<br>
&gt; &gt;&gt; privacy (via the API), and provide documentation that encoura=
ges<br>
&gt; &gt;&gt; sites to do so. As you point out, I think it&#39;s difficult =
for the<br>
&gt; &gt;&gt; browser to automatically figure out what to do here, and when=
 to do it.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; The identity of the &quot;called&quot; user may not be a fact=
or in certain<br>
&gt; &gt;&gt; applications (e.g. some p2p game), making it unclear as to wh=
ether<br>
&gt; &gt;&gt; forcing use of relay is even the right thing to do.<br>
&gt; &gt;<br>
&gt; &gt; I concur with all this.<br>
&gt; &gt;<br>
&gt; &gt; I think it&#39;s also important to reiterate that this is about t=
he site<br>
&gt; &gt; cooperating to protect your privacy. If you want to protect the u=
ser&#39;s<br>
&gt; &gt; IP address from the site, then the user needs to use Torbutton or=
 some such.<br>
&gt; &gt; And I would want Torbutton to arrange relay my RTCWEB traffic as =
well<br>
&gt; &gt; (though likely that can&#39;t go through Tor for performance reas=
ons).<br>
&gt; &gt;<br>
&gt; &gt; -Ekr<br>
&gt; &gt; _______________________________________________<br>
&gt; &gt; rtcweb mailing list<br>
&gt; &gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://=
www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt;<br>
&gt; &gt; __________ Information from ESET NOD32 Antivirus, version of viru=
s<br>
&gt; &gt; signature database 7138 (20120515) __________<br>
&gt; &gt;<br>
&gt; &gt; The message was checked by ESET NOD32 Antivirus.<br>
&gt; &gt;<br>
&gt; &gt; <a href=3D"http://www.eset.com">http://www.eset.com</a><br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt; &gt;<br>
&gt;<br>
&gt; __________ Information from ESET NOD32 Antivirus, version of virus sig=
nature database 7138 (20120515) __________<br>
&gt;<br>
&gt; The message was checked by ESET NOD32 Antivirus.<br>
&gt;<br>
&gt; <a href=3D"http://www.eset.com">http://www.eset.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>
</p>

--e89a8f83a703a9b22004c04525ce--

From denglingli@chinamobile.com  Thu May 17 18:20:23 2012
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From: =?UTF-8?B?6YKT54G16I6JL2RlbmdsaW5nbGk=?= <denglingli@chinamobile.com>
To: "'Martin Thomson'" <martin.thomson@gmail.com>
References: <CABkgnnWEqqAD0rPrFK3mj5jBCoqmyPTY_PFgVyg8t84egN0Baw@mail.gmail.com>	<CAOJ7v-2K9VS0YKo=RJSPLcmT=X-kTQ3ynyp9APin92eW=LceTg@mail.gmail.com>	<CABcZeBPDJ6MQgQFBW4+Gnq9V-Oo-Gb9d_7nuCnySzbFBzDoXVg@mail.gmail.com>	<003101cd33c9$593793b0$0ba6bb10$@chinamobile.com>	<B7AB7EC0-35B0-46BA-8786-6AD92A9AD168@rtfm.com>	<007b01cd33f1$03ccc9e0$0b665da0$@chinamobile.com>	<007c01cd33f1$d2dc4bc0$7894e340$@chinamobile.com> <CABkgnnX=JBafTtML45NtpTu1ixDuJeM3MT3Xsre9tOn6dOy5Eg@mail.gmail.com>
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Date: Fri, 18 May 2012 09:17:09 +0800
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My mistake. It should be at =E2=80=9Cpeerconnection=E2=80=9D rather than =
=E2=80=9Cgetusermedia=E2=80=9D in order to get what we meant.

=20

=E5=8F=91=E4=BB=B6=E4=BA=BA: Martin Thomson =
[mailto:martin.thomson@gmail.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8818=E6=97=A5 =
9:05
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: rtcweb@ietf.org; Eric Rescorla
=E4=B8=BB=E9=A2=98: Re: [rtcweb] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: =
=E7=AD=94=E5=A4=8D: Relaying for privacy

=20

I do not think that any option other than what Eric described is truly =
feasible. That is, of the options provided, only the one where the site =
has to be trusted bit to reveal your IP is likely to succeed.

getUserMedia provides a cue of a fundamentally different nature to that =
you are looking for in this use case. getUserMedia signals that the user =
is ok with the site gaining access to their camera/microphone.  That is =
a contract between the user and the site, not the two users.

If you don't trust the site, then you are in Torbutton land. That is, =
specific UI for this.

I've concluded that we don't need this requirement. There is little that =
a browser can sensibly do beyond something like Torbutton.

And I'm not convinced that trying to replicate, or require the =
replication of Torbutton, is a great idea either. Tor users understand =
the value it provides, most other users won't care for the added =
complication.

On May 16, 2012 10:59 PM, "=E9=82=93=E7=81=B5=E8=8E=89/denglingli" =
<denglingli@chinamobile.com> wrote:
>
> Sorry, by " In either cases, the setting can be done through =
getusermedia as Martin said." I mean " In either cases, the setting =
could be done through getusermedia as Martin said."
> Since according to our experience with chrome so far, media relaying =
through turn is not yet supported.
>
> Lingli
>
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 =
=E9=82=93=E7=81=B5=E8=8E=89/denglingli
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2012=E5=B9=B45=E6=9C=8817=E6=97=A5 13:51
> =E6=94=B6=E4=BB=B6=E4=BA=BA: 'Eric Rescorla'
> =E6=8A=84=E9=80=81: rtcweb@ietf.org
> =E4=B8=BB=E9=A2=98: [rtcweb] =E7=AD=94=E5=A4=8D: =E7=AD=94=E5=A4=8D: =
Relaying for privacy
>
> It makes sense to me.
> However, considering the relaying burden would not be trivial, I'd =
further prefer that the enable/disable button be dedicatedly set for the =
RTCWEB instance in question, rather than the whole pack of connections =
of the browsers.
> In other words, it is better to be triggered at the time of =
page-loading for the RTCWEB app for a global configuration (keeping =
current IP privacy for all callers) or when called by for a specific =
configuration (keeping privacy for the incoming caller).
> In either cases, the setting can be done through getusermedia as =
Martin said.
>
> BR
> Lingli
>
> -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> =E5=8F=91=E4=BB=B6=E4=BA=BA: Eric Rescorla [mailto:ekr@rtfm.com]
> =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2012=E5=B9=B45=E6=9C=8817=E6=97=A5 10:07
> =E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
> =E6=8A=84=E9=80=81: Justin Uberti; <rtcweb@ietf.org>
> =E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Relaying for =
privacy
>
> Yes. Tor is too slow to do voice effectively. I would expect torbutton =
or a torbutton-like to route media through a turn server. This will =
provide worse privacy than onion routing but better than just letting =
the server control the routing.
>
> Ekr
>
> On May 16, 2012, at 18:06, =E9=82=93=E7=81=B5=E8=8E=89/denglingli =
<denglingli@chinamobile.com> wrote:
>
> > Hi, Ekr
> >
> > I agree with you about the Martin's comments.
> > For the proposal of Torbutton, I have a concern about the employment
> > of union routing (encrypting and decrypting several times between =
the
> > relaying nodes along the way) in Tor would be not acceptable to the
> > RTC requirements of VoIP use-cases.
> > Would this be an issue?
> >
> > BR
> > Lingli
> >
> > -----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
> > =E5=8F=91=E4=BB=B6=E4=BA=BA: rtcweb-bounces@ietf.org =
[mailto:rtcweb-bounces@ietf.org] =E4=BB=A3=E8=A1=A8 Eric
> > Rescorla
> > =E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: =
2012=E5=B9=B45=E6=9C=8817=E6=97=A5 6:10
> > =E6=94=B6=E4=BB=B6=E4=BA=BA: Justin Uberti
> > =E6=8A=84=E9=80=81: rtcweb@ietf.org
> > =E4=B8=BB=E9=A2=98: Re: [rtcweb] Relaying for privacy
> >
> > 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
> >> privacy (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.
> >>
> >> 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
> >> forcing use of relay is even the right thing to do.
> >
> > I concur with all this.
> >
> > I think it's also important to reiterate that this is about the site
> > cooperating to protect your privacy. If you want to protect the =
user's
> > IP address from 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).
> >
> > -Ekr
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
> > __________ Information from ESET NOD32 Antivirus, version of virus
> > signature database 7138 (20120515) __________
> >
> > The message was checked by ESET NOD32 Antivirus.
> >
> > http://www.eset.com
> >
> >
> >
>
> __________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.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


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</o:shapelayout></xml><![endif]--></head><body lang=3DZH-CN link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>My mistake. It should be at =E2=80=9Cpeerconnection=E2=80=9D rather =
than =E2=80=9Cgetusermedia=E2=80=9D in order to get what we =
meant.<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt'>=E5=8F=91=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>:</span></span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt'> Martin Thomson =
[mailto:martin.thomson@gmail.com] <br></span><b><span =
style=3D'font-size:10.0pt'>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span =
lang=3DEN-US>:</span></span></b><span lang=3DEN-US =
style=3D'font-size:10.0pt'> 2012</span><span =
style=3D'font-size:10.0pt'>=E5=B9=B4<span =
lang=3DEN-US>5</span>=E6=9C=88<span lang=3DEN-US>18</span>=E6=97=A5<span =
lang=3DEN-US> 9:05<br></span><b>=E6=94=B6=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>:</span></b><span lang=3DEN-US> =
</span>=E9=82=93=E7=81=B5=E8=8E=89<span =
lang=3DEN-US>/denglingli<br></span><b>=E6=8A=84=E9=80=81<span =
lang=3DEN-US>:</span></b><span lang=3DEN-US> rtcweb@ietf.org; Eric =
Rescorla<br></span><b>=E4=B8=BB=E9=A2=98<span =
lang=3DEN-US>:</span></b><span lang=3DEN-US> Re: [rtcweb] =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: Relaying for =
privacy<o:p></o:p></span></span></p><p class=3DMsoNormal><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p><span lang=3DEN-US>I do not =
think that any option other than what Eric described is truly feasible. =
That is, of the options provided, only the one where the site has to be =
trusted bit to reveal your IP is likely to =
succeed.<o:p></o:p></span></p><p><span lang=3DEN-US>getUserMedia =
provides a cue of a fundamentally different nature to that you are =
looking for in this use case. getUserMedia signals that the user is ok =
with the site gaining access to their camera/microphone.&nbsp; That is a =
contract between the user and the site, not the two =
users.<o:p></o:p></span></p><p><span lang=3DEN-US>If you don't trust the =
site, then you are in Torbutton land. That is, specific UI for =
this.<o:p></o:p></span></p><p><span lang=3DEN-US>I've concluded that we =
don't need this requirement. There is little that a browser can sensibly =
do beyond something like Torbutton.<o:p></o:p></span></p><p><span =
lang=3DEN-US>And I'm not convinced that trying to replicate, or require =
the replication of Torbutton, is a great idea either. Tor users =
understand the value it provides, most other users won't care for the =
added complication.<o:p></o:p></span></p><p><span lang=3DEN-US>On May =
16, 2012 10:59 PM, &quot;</span>=E9=82=93=E7=81=B5=E8=8E=89<span =
lang=3DEN-US>/denglingli&quot; &lt;<a =
href=3D"mailto:denglingli@chinamobile.com">denglingli@chinamobile.com</a>=
&gt; wrote:<br>&gt;<br>&gt; Sorry, by &quot; In either cases, the =
setting can be done through getusermedia as Martin said.&quot; I mean =
&quot; In either cases, the setting could be done through getusermedia =
as Martin said.&quot;<br>&gt; Since according to our experience with =
chrome so far, media relaying through turn is not yet =
supported.<br>&gt;<br>&gt; Lingli<br>&gt;<br>&gt; =
-----</span>=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6<span =
lang=3DEN-US>-----<br>&gt; </span>=E5=8F=91=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
</span>=E4=BB=A3=E8=A1=A8 =E9=82=93=E7=81=B5=E8=8E=89<span =
lang=3DEN-US>/denglingli<br>&gt; =
</span>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span lang=3DEN-US>: =
2012</span>=E5=B9=B4<span lang=3DEN-US>5</span>=E6=9C=88<span =
lang=3DEN-US>17</span>=E6=97=A5<span lang=3DEN-US> 13:51<br>&gt; =
</span>=E6=94=B6=E4=BB=B6=E4=BA=BA<span lang=3DEN-US>: 'Eric =
Rescorla'<br>&gt; </span>=E6=8A=84=E9=80=81<span lang=3DEN-US>: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; =
</span>=E4=B8=BB=E9=A2=98<span lang=3DEN-US>: [rtcweb] =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: Relaying for =
privacy<br>&gt;<br>&gt; It makes sense to me.<br>&gt; However, =
considering the relaying burden would not be trivial, I'd further prefer =
that the enable/disable button be dedicatedly set for the RTCWEB =
instance in question, rather than the whole pack of connections of the =
browsers.<br>&gt; In other words, it is better to be triggered at the =
time of page-loading for the RTCWEB app for a global configuration =
(keeping current IP privacy for all callers) or when called by for a =
specific configuration (keeping privacy for the incoming =
caller).<br>&gt; In either cases, the setting can be done through =
getusermedia as Martin said.<br>&gt;<br>&gt; BR<br>&gt; =
Lingli<br>&gt;<br>&gt; =
-----</span>=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6<span =
lang=3DEN-US>-----<br>&gt; </span>=E5=8F=91=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>: Eric Rescorla [mailto:<a =
href=3D"mailto:ekr@rtfm.com">ekr@rtfm.com</a>]<br>&gt; =
</span>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span lang=3DEN-US>: =
2012</span>=E5=B9=B4<span lang=3DEN-US>5</span>=E6=9C=88<span =
lang=3DEN-US>17</span>=E6=97=A5<span lang=3DEN-US> 10:07<br>&gt; =
</span>=E6=94=B6=E4=BB=B6=E4=BA=BA<span lang=3DEN-US>: =
</span>=E9=82=93=E7=81=B5=E8=8E=89<span lang=3DEN-US>/denglingli<br>&gt; =
</span>=E6=8A=84=E9=80=81<span lang=3DEN-US>: Justin Uberti; &lt;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br>&gt; =
</span>=E4=B8=BB=E9=A2=98<span lang=3DEN-US>: Re: =
</span>=E7=AD=94=E5=A4=8D<span lang=3DEN-US>: [rtcweb] Relaying for =
privacy<br>&gt;<br>&gt; Yes. Tor is too slow to do voice effectively. I =
would expect torbutton or a torbutton-like to route media through a turn =
server. This will provide worse privacy than onion routing but better =
than just letting the server control the routing.<br>&gt;<br>&gt; =
Ekr<br>&gt;<br>&gt; On May 16, 2012, at 18:06, =
</span>=E9=82=93=E7=81=B5=E8=8E=89<span lang=3DEN-US>/denglingli &lt;<a =
href=3D"mailto:denglingli@chinamobile.com">denglingli@chinamobile.com</a>=
&gt; wrote:<br>&gt;<br>&gt; &gt; Hi, Ekr<br>&gt; &gt;<br>&gt; &gt; I =
agree with you about the Martin's comments.<br>&gt; &gt; For the =
proposal of Torbutton, I have a concern about the employment<br>&gt; =
&gt; of union routing (encrypting and decrypting several times between =
the<br>&gt; &gt; relaying nodes along the way) in Tor would be not =
acceptable to the<br>&gt; &gt; RTC requirements of VoIP =
use-cases.<br>&gt; &gt; Would this be an issue?<br>&gt; &gt;<br>&gt; =
&gt; BR<br>&gt; &gt; Lingli<br>&gt; &gt;<br>&gt; &gt; =
-----</span>=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6<span =
lang=3DEN-US>-----<br>&gt; &gt; </span>=E5=8F=91=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>: <a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> =
[mailto:<a =
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>] =
</span>=E4=BB=A3=E8=A1=A8<span lang=3DEN-US> Eric<br>&gt; &gt; =
Rescorla<br>&gt; &gt; </span>=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4<span =
lang=3DEN-US>: 2012</span>=E5=B9=B4<span =
lang=3DEN-US>5</span>=E6=9C=88<span lang=3DEN-US>17</span>=E6=97=A5<span =
lang=3DEN-US> 6:10<br>&gt; &gt; </span>=E6=94=B6=E4=BB=B6=E4=BA=BA<span =
lang=3DEN-US>: Justin Uberti<br>&gt; &gt; </span>=E6=8A=84=E9=80=81<span =
lang=3DEN-US>: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; &gt; =
</span>=E4=B8=BB=E9=A2=98<span lang=3DEN-US>: Re: [rtcweb] Relaying for =
privacy<br>&gt; &gt;<br>&gt; &gt; On Wed, May 16, 2012 at 2:54 PM, =
Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com">juberti@google.com</a>&gt; =
wrote:<br>&gt; &gt;&gt; My view was that we need to make it _possible_ =
for sites to respect<br>&gt; &gt;&gt; privacy (via the API), and provide =
documentation that encourages<br>&gt; &gt;&gt; sites to do so. As you =
point out, I think it's difficult for the<br>&gt; &gt;&gt; browser to =
automatically figure out what to do here, and when to do it.<br>&gt; =
&gt;&gt;<br>&gt; &gt;&gt; The identity of the &quot;called&quot; user =
may not be a factor in certain<br>&gt; &gt;&gt; applications (e.g. some =
p2p game), making it unclear as to whether<br>&gt; &gt;&gt; forcing use =
of relay is even the right thing to do.<br>&gt; &gt;<br>&gt; &gt; I =
concur with all this.<br>&gt; &gt;<br>&gt; &gt; I think it's also =
important to reiterate that this is about the site<br>&gt; &gt; =
cooperating to protect your privacy. If you want to protect the =
user's<br>&gt; &gt; IP address from the site, then the user needs to use =
Torbutton or some such.<br>&gt; &gt; And I would want Torbutton to =
arrange relay my RTCWEB traffic as well<br>&gt; &gt; (though likely that =
can't go through Tor for performance reasons).<br>&gt; &gt;<br>&gt; &gt; =
-Ekr<br>&gt; &gt; =
_______________________________________________<br>&gt; &gt; rtcweb =
mailing list<br>&gt; &gt; <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>&gt; &gt; <a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><br>&gt; &gt;<br>&gt; &gt; __________ =
Information from ESET NOD32 Antivirus, version of virus<br>&gt; &gt; =
signature database 7138 (20120515) __________<br>&gt; &gt;<br>&gt; &gt; =
The message was checked by ESET NOD32 Antivirus.<br>&gt; &gt;<br>&gt; =
&gt; <a href=3D"http://www.eset.com">http://www.eset.com</a><br>&gt; =
&gt;<br>&gt; &gt;<br>&gt; &gt;<br>&gt;<br>&gt; __________ Information =
from ESET NOD32 Antivirus, version of virus signature database 7138 =
(20120515) __________<br>&gt;<br>&gt; The message was checked by ESET =
NOD32 Antivirus.<br>&gt;<br>&gt; <a =
href=3D"http://www.eset.com">http://www.eset.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.ietf.or=
g/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.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></span></p></div></body></html>
------=_NextPart_000_0041_01CD34D7.0098F690--


From stefan.lk.hakansson@ericsson.com  Fri May 18 05:13:51 2012
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 05/16/2012 03:31 AM, Cullen Jennings wrote:
...
>
> New use case:
>
> Imagine a service like webex that facilitates collaboration between
> two companies called A and B. Webex has no need to see the content of
> the communication from A to B. So webex might run the web server that
> helps set up a RTCWeb session between A and B, but webex needs to be
> able to write it's code such that it does not get the keying material
> used to encrypt the media from A to B and the companies need to be
> able to verify that webex did not get the keys. Webex has no need to
> see the media from A to B and if it can provide this sort of promise
> that it does not get the media, it makes it much easier for a
> customer (say Juniper) to use webex and know the contents of their
> calls are not being sold or used by a competitor. Given how much
> WebRTC will be used by cloud services, this is an important feature.

At least on the surface, this seems to have a lot in common with what 
Oscar proposed the other day ([1]), at least if the WebEx session has 
more than one concurrent user.

[1] http://www.ietf.org/mail-archive/web/rtcweb/current/msg04457.html

> When installing the "webex" application in the browser and granting
> permissions to the application, it would be good to have one of the
> permissions be if the JS got access to the media or keying material
> for it to enforce this promise.
>
> Note this use case is not phrase to say this is the only way it can
> work. This is not mutually exclusive with they type of use cases
> described by Skype folks where there is an explicitly desire to make
> sure the calling service does have the keys to decrypt the media.
>
>
>
>
>
>
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Sat May 19 06:46:42 2012
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Subject: [rtcweb] Hold the date: Trying to get a workshop on congestion control for real time interactive media together....
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FYI ... .the IAB is trying to get together a workshop (of the typical 
IAB format; people get invited if they contribute ideas beforehand) 
together before the Vancouver IETF, covering the same subject area as 
the RTP congestion work.

This would have a goal, among other things, of getting people to think 
about the subject matter for our BOF/WG, and get people better informed 
(and interested!) before the BOF.

The likely date is Saturday before the IETF (July 28), but this is not 
settled yet, so don't book tickets specifically for this until you hear 
the official announcement....

              Harald


From magnus.westerlund@ericsson.com  Mon May 21 00:01:27 2012
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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On 2012-05-21 03:44, éçµè/denglingli wrote:
> Hi, Magnus
> 
> It seems to me that there may be another security threat in multi-party
> applications of COP, where an entity needs to combine multiple sets of
> requested parameters, than the one discussed in the draft. 
> That the initial downgrading of the combined potential ceiling for collected
> parameters for media quality (codec capabilities plus COP parameters as
> stated in Section 5) through SDP transaction by a malicious participant.
> Unlike the one stated in Section 8, the latter behavior only happens once
> and could neither been distinguished afterwards as "actively harmful" nor to
> be ignored in order to serve actually poorly-equipped users.
> Would that be an issue?
> 

Yes, this is clearly a security threat to the complete solution. Not
that it is specific to codec control. It is a threat to all things
expressed in the SDP, like which codecs being used, security mechanism
is negotiated etc.

In the WebRTC security architecture my undestanding is that it so far
are a deliberate choice of allowing the JavaScript and the web browser
to be allowed to modify the SDP if desired by them. Thus a security
model based on hop by hop security for the JSEP/SDP messages has been
selected. For example the usage of HTTPS / Websocket over TLS can
provide the security to prevent third parties not directly addressed
from seeing and affecting the JSEP/SDP messages.

Cheers

Magnus Westerlund

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


From denglingli@chinamobile.com  Mon May 21 02:55:13 2012
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Subject: [rtcweb] =?utf-8?b?562U5aSNOiDnrZTlpI06ICBGd2Q6IEktRCBBY3Rpb246?= =?utf-8?q?_draft-westerlund-rtcweb-codec-control-00=2Etxt?=
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Agreed.

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Magnus Westerlund =
[mailto:magnus.westerlund@ericsson.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8821=E6=97=A5 =
14:59
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] Fwd: I-D Action: =
draft-westerlund-rtcweb-codec-control-00.txt

On 2012-05-21 03:44, =E9=82=93=E7=81=B5=E8=8E=89/denglingli wrote:
> Hi, Magnus
>=20
> It seems to me that there may be another security threat in=20
> multi-party applications of COP, where an entity needs to combine=20
> multiple sets of requested parameters, than the one discussed in the =
draft.
> That the initial downgrading of the combined potential ceiling for=20
> collected parameters for media quality (codec capabilities plus COP=20
> parameters as stated in Section 5) through SDP transaction by a =
malicious participant.
> Unlike the one stated in Section 8, the latter behavior only happens=20
> once and could neither been distinguished afterwards as "actively=20
> harmful" nor to be ignored in order to serve actually poorly-equipped =
users.
> Would that be an issue?
>=20

Yes, this is clearly a security threat to the complete solution. Not =
that it is specific to codec control. It is a threat to all things =
expressed in the SDP, like which codecs being used, security mechanism =
is negotiated etc.

In the WebRTC security architecture my undestanding is that it so far =
are a deliberate choice of allowing the JavaScript and the web browser =
to be allowed to modify the SDP if desired by them. Thus a security =
model based on hop by hop security for the JSEP/SDP messages has been =
selected. For example the usage of HTTPS / Websocket over TLS can =
provide the security to prevent third parties not directly addressed =
from seeing and affecting the JSEP/SDP messages.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From magnus.westerlund@ericsson.com  Mon May 21 06:31:09 2012
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Subject: [rtcweb] Interim participation registration
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Hi,

I would like to have a firm understanding of how many that really will
show up for either day of the interim meeting. The main purpose is to
ensure that break snacks and lunch are provided in the right amount.

So please fill in this doodle to register your participation for each of
the days that you will be present.

http://www.doodle.com/x3z6uhsy4q4s6fc4

Thanks

Magnus Westerlund
Interim meeting host



----------------------------------------------------------------------
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 martin.thomson@gmail.com  Mon May 21 08:42:51 2012
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> On 2012-05-21 03:44, =E9=82=93=E7=81=B5=E8=8E=89/denglingli wrote:
>> That the initial downgrading of the combined potential ceiling for colle=
cted
>> parameters for media quality (codec capabilities plus COP parameters as
>> stated in Section 5) through SDP transaction by a malicious participant.

As Magnus pointed out, this is true of anything that can be signaled.
The application can do anything up to (and including) a complete
denial of service.  The fact that the application is effectively the
victim of this attack means that we probably shouldn't concern
ourselves overmuch.

From denglingli@chinamobile.com  Mon May 21 18:29:18 2012
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To: "'Martin Thomson'" <martin.thomson@gmail.com>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Date: Tue, 22 May 2012 09:25:56 +0800
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Cc: rtcweb@ietf.org
Subject: [rtcweb] =?utf-8?b?562U5aSNOiAg562U5aSNOiBGd2Q6IEktRCBBY3Rpb246?= =?utf-8?q?_draft-westerlund-rtcweb-codec-control-00=2Etxt?=
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Hi, Martin

You are right. According to the current discussion, I would suggest two =
options about the security threat statement:=20
1, remove the item c from Section 8; or
2, add a few words to notify what is left out and may be of concern in =
the field and people would know what to expect in reality.
Would you agree?

BR=20
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Martin Thomson =
[mailto:martin.thomson@gmail.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8821=E6=97=A5 =
23:43
=E6=94=B6=E4=BB=B6=E4=BA=BA: Magnus Westerlund
=E6=8A=84=E9=80=81: =E9=82=93=E7=81=B5=E8=8E=89/denglingli; =
rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: Re: [rtcweb] =E7=AD=94=E5=A4=8D: Fwd: I-D Action: =
draft-westerlund-rtcweb-codec-control-00.txt

> On 2012-05-21 03:44, =E9=82=93=E7=81=B5=E8=8E=89/denglingli wrote:
>> That the initial downgrading of the combined potential ceiling for=20
>> collected parameters for media quality (codec capabilities plus COP=20
>> parameters as stated in Section 5) through SDP transaction by a =
malicious participant.

As Magnus pointed out, this is true of anything that can be signaled.
The application can do anything up to (and including) a complete denial =
of service.  The fact that the application is effectively the victim of =
this attack means that we probably shouldn't concern ourselves overmuch.

__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From martin.thomson@gmail.com  Mon May 21 19:51:10 2012
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Date: Mon, 21 May 2012 19:51:09 -0700
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From: Martin Thomson <martin.thomson@gmail.com>
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On 21 May 2012 18:25, =E9=82=93=E7=81=B5=E8=8E=89/denglingli <denglingli@ch=
inamobile.com> wrote:
> [...] I would suggest two options about the security threat statement:
> 1, remove the item c from Section 8; or
> 2, add a few words to notify what is left out and may be of concern in th=
e field and people would know what to expect in reality.
> Would you agree?

I'm not sure about the full implications of these sorts of mechanisms
yet.  I was speaking more generally about rtcweb.  In general however,
it is true that a site (or any entity on the signaling path) can
degrade the experience.

Adding a mechanism like what Magnus proposes lets any participant in a
call perform the same sorts of degradations.  As a site operator, I
want to ensure that this doesn't happen, so that my users always get
the best experience from my site.  So that's not necessarily an
exposure that I like.  Of course, I haven't really analyzed the upside
of this particular trade-off yet...

--Martin

p.s. it might be possible for a mechanism like this to "fix"
downgrades chosen by the site, though that wouldn't be advisable.  The
proposal indicates otherwise: that the constraints set during
negotiation are firm.

From denglingli@chinamobile.com  Mon May 21 20:25:57 2012
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To: "'Martin Thomson'" <martin.thomson@gmail.com>
References: <20120516140228.4049.34228.idtracker@ietfa.amsl.com>	<4FB3B55F.3080607@ericsson.com>	<003f01cd36f3$5302aed0$f9080c70$@chinamobile.com>	<4FB9E79C.1050300@ericsson.com>	<CABkgnnUs4K3aP7Ge4+sQ7e6UDEwx-hGJi50Tn6hG4rEwiz98HQ@mail.gmail.com>	<001901cd37b9$d66c9490$8345bdb0$@chinamobile.com> <CABkgnnUanv7oBzhOC1Eg69u7SznZUKORQi_ZBaa0LUEi64mf6g@mail.gmail.com>
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Date: Tue, 22 May 2012 11:22:43 +0800
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I don't think it is a good idea to loosen the firm constraints that the =
site/a third party would set to the media negotiation.
Looking at the other side, the browser may not know everything about the =
session as well as the other side of communication (as JSEP suggested).
Besides, there may be policies that a service provider would like to =
configure in order to regulate media behaviors of its users, especially =
when it comes to interworking with legacy networks.

Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Martin Thomson =
[mailto:martin.thomson@gmail.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8822=E6=97=A5 =
10:51
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: Magnus Westerlund; rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] =E7=AD=94=E5=A4=8D: =
Fwd: I-D Action: draft-westerlund-rtcweb-codec-control-00.txt

On 21 May 2012 18:25, =E9=82=93=E7=81=B5=E8=8E=89/denglingli =
<denglingli@chinamobile.com> wrote:
> [...] I would suggest two options about the security threat statement:
> 1, remove the item c from Section 8; or 2, add a few words to notify=20
> what is left out and may be of concern in the field and people would =
know what to expect in reality.
> Would you agree?

I'm not sure about the full implications of these sorts of mechanisms =
yet.  I was speaking more generally about rtcweb.  In general however, =
it is true that a site (or any entity on the signaling path) can degrade =
the experience.

Adding a mechanism like what Magnus proposes lets any participant in a =
call perform the same sorts of degradations.  As a site operator, I want =
to ensure that this doesn't happen, so that my users always get the best =
experience from my site.  So that's not necessarily an exposure that I =
like.  Of course, I haven't really analyzed the upside of this =
particular trade-off yet...

--Martin

p.s. it might be possible for a mechanism like this to "fix"
downgrades chosen by the site, though that wouldn't be advisable.  The =
proposal indicates otherwise: that the constraints set during =
negotiation are firm.

__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From magnus.westerlund@ericsson.com  Mon May 21 23:43:21 2012
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On 2012-05-22 03:25, éçµè/denglingli wrote:
> Hi, Martin
> 
> You are right. According to the current discussion, I would suggest two options about the security threat statement: 
> 1, remove the item c from Section 8; or

I don't think this is an appropriate choice. The reason is that it
applies to multi-party cases where an given end-point targets the other
participants. Mitigation in the media plane central node for this attack
is something the implementation should have.

> 2, add a few words to notify what is left out and may be of concern in the field and people would know what to expect in reality.
> Would you agree?

I would like to make it clear that this documents security consideration
is a short summary of the most important attacks. The COP drafts
security consideration is not existing and that will be addressed in the
next version. That will clearly discuss the SDP angle and at least point
out the need for protection between the nodes. But also that in
multi-party there exist a potential for down-grading the general
constraints.

I will consider adding the SDP based downgrade into this document also.

Cheers

Magnus Westerlund

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


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References: <20120516140228.4049.34228.idtracker@ietfa.amsl.com> <4FB3B55F.3080607@ericsson.com> <003f01cd36f3$5302aed0$f9080c70$@chinamobile.com> <4FB9E79C.1050300@ericsson.com> <CABkgnnUs4K3aP7Ge4+sQ7e6UDEwx-hGJi50Tn6hG4rEwiz98HQ@mail.gmail.com> <001901cd37b9$d66c9490$8345bdb0$@chinamobile.com> <4FBB3586.4050902@ericsson.com>
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Agreed.=20

Thanks,
Lingli

-----=E9=82=AE=E4=BB=B6=E5=8E=9F=E4=BB=B6-----
=E5=8F=91=E4=BB=B6=E4=BA=BA: Magnus Westerlund =
[mailto:magnus.westerlund@ericsson.com]=20
=E5=8F=91=E9=80=81=E6=97=B6=E9=97=B4: 2012=E5=B9=B45=E6=9C=8822=E6=97=A5 =
14:43
=E6=94=B6=E4=BB=B6=E4=BA=BA: =E9=82=93=E7=81=B5=E8=8E=89/denglingli
=E6=8A=84=E9=80=81: 'Martin Thomson'; rtcweb@ietf.org
=E4=B8=BB=E9=A2=98: Re: =E7=AD=94=E5=A4=8D: [rtcweb] =E7=AD=94=E5=A4=8D: =
Fwd: I-D Action: draft-westerlund-rtcweb-codec-control-00.txt

On 2012-05-22 03:25, =E9=82=93=E7=81=B5=E8=8E=89/denglingli wrote:
> Hi, Martin
>=20
> You are right. According to the current discussion, I would suggest =
two options about the security threat statement:=20
> 1, remove the item c from Section 8; or

I don't think this is an appropriate choice. The reason is that it =
applies to multi-party cases where an given end-point targets the other =
participants. Mitigation in the media plane central node for this attack =
is something the implementation should have.

> 2, add a few words to notify what is left out and may be of concern in =
the field and people would know what to expect in reality.
> Would you agree?

I would like to make it clear that this documents security consideration =
is a short summary of the most important attacks. The COP drafts =
security consideration is not existing and that will be addressed in the =
next version. That will clearly discuss the SDP angle and at least point =
out the need for protection between the nodes. But also that in =
multi-party there exist a potential for down-grading the general =
constraints.

I will consider adding the SDP based downgrade into this document also.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F=C3=A4r=C3=B6gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


__________ Information from ESET NOD32 Antivirus, version of virus =
signature database 7138 (20120515) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From dromasca@avaya.com  Tue May 22 04:18:12 2012
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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <rtcweb@ietf.org>
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Do you need a count only for people who plan to attend on-site?=20

Dan




> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Magnus Westerlund
> Sent: Monday, May 21, 2012 4:31 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Interim participation registration
>=20
> Hi,
>=20
> I would like to have a firm understanding of how many that really will
> show up for either day of the interim meeting. The main purpose is to
> ensure that break snacks and lunch are provided in the right amount.
>=20
> So please fill in this doodle to register your participation for each
> of
> the days that you will be present.
>=20
> http://www.doodle.com/x3z6uhsy4q4s6fc4
>=20
> Thanks
>=20
> Magnus Westerlund
> Interim meeting host
>=20
>=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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From magnus.westerlund@ericsson.com  Tue May 22 05:07:14 2012
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On 2012-05-22 13:18, Romascanu, Dan (Dan) wrote:
> Do you need a count only for people who plan to attend on-site? 

Good question. Yes, the this was for physical participation. We know
that we will have some number of remote participation. I don't know how
important it is to get a head count for them.

Cheers

Magnus

> 
> Dan
> 
> 
> 
> 
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Magnus Westerlund
>> Sent: Monday, May 21, 2012 4:31 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] Interim participation registration
>>
>> Hi,
>>
>> I would like to have a firm understanding of how many that really will
>> show up for either day of the interim meeting. The main purpose is to
>> ensure that break snacks and lunch are provided in the right amount.
>>
>> So please fill in this doodle to register your participation for each
>> of
>> the days that you will be present.
>>
>> http://www.doodle.com/x3z6uhsy4q4s6fc4
>>
>> Thanks
>>
>> Magnus Westerlund
>> Interim meeting host
>>
>>
>>
>> ----------------------------------------------------------------------
>> 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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From: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Maybe it is important for Webex, if you plan to use Webex for remote =
participation. Or other tool. I do not know. In any case, you would need =
a separate doodle or a different option on the current doodle.=20

Regards,

Dan




> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Tuesday, May 22, 2012 3:07 PM
> To: Romascanu, Dan (Dan)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Interim participation registration
>=20
> On 2012-05-22 13:18, Romascanu, Dan (Dan) wrote:
> > Do you need a count only for people who plan to attend on-site?
>=20
> Good question. Yes, the this was for physical participation. We know
> that we will have some number of remote participation. I don't know =
how
> important it is to get a head count for them.
>=20
> Cheers
>=20
> Magnus
>=20
> >
> > Dan
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >> Behalf Of Magnus Westerlund
> >> Sent: Monday, May 21, 2012 4:31 PM
> >> To: rtcweb@ietf.org
> >> Subject: [rtcweb] Interim participation registration
> >>
> >> Hi,
> >>
> >> I would like to have a firm understanding of how many that really
> will
> >> show up for either day of the interim meeting. The main purpose is
> to
> >> ensure that break snacks and lunch are provided in the right =
amount.
> >>
> >> So please fill in this doodle to register your participation for
> each
> >> of
> >> the days that you will be present.
> >>
> >> http://www.doodle.com/x3z6uhsy4q4s6fc4
> >>
> >> Thanks
> >>
> >> Magnus Westerlund
> >> Interim meeting host
> >>
> >>
> >>
> >> =
--------------------------------------------------------------------
> --
> >> 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
>=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
> ----------------------------------------------------------------------


From harald@alvestrand.no  Tue May 22 06:10:14 2012
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This is a multi-part message in MIME format.
--------------060308010508000706060201
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Content-Transfer-Encoding: 8bit

Note that the registration for the WEBRTC meeting on Monday is separate, 
and that the participants are listed here:

http://www.w3.org/2011/04/webrtc/wiki/June_11_2012

Formalities...

             Harald

On 05/21/2012 03:31 PM, Magnus Westerlund wrote:
> Hi,
>
> I would like to have a firm understanding of how many that really will
> show up for either day of the interim meeting. The main purpose is to
> ensure that break snacks and lunch are provided in the right amount.
>
> So please fill in this doodle to register your participation for each of
> the days that you will be present.
>
> http://www.doodle.com/x3z6uhsy4q4s6fc4
>
> Thanks
>
> Magnus Westerlund
> Interim meeting host
>
>
>
> ----------------------------------------------------------------------
> 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
>


--------------060308010508000706060201
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">
    Note that the registration for the WEBRTC meeting on Monday is
    separate, and that the participants are listed here:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=ISO-8859-1">
    <a href="http://www.w3.org/2011/04/webrtc/wiki/June_11_2012">http://www.w3.org/2011/04/webrtc/wiki/June_11_2012</a><br>
    <br>
    Formalities...<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Harald<br>
    <br>
    On 05/21/2012 03:31 PM, Magnus Westerlund wrote:
    <blockquote cite="mid:4FBA4399.4050909@ericsson.com" type="cite">
      <pre wrap="">Hi,

I would like to have a firm understanding of how many that really will
show up for either day of the interim meeting. The main purpose is to
ensure that break snacks and lunch are provided in the right amount.

So please fill in this doodle to register your participation for each of
the days that you will be present.

<a class="moz-txt-link-freetext" href="http://www.doodle.com/x3z6uhsy4q4s6fc4">http://www.doodle.com/x3z6uhsy4q4s6fc4</a>

Thanks

Magnus Westerlund
Interim meeting host



----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM
----------------------------------------------------------------------
Ericsson AB                | Phone  +46 10 7148287
F&auml;r&ouml;gatan 6                | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden| mailto: <a class="moz-txt-link-abbreviated" href="mailto:magnus.westerlund@ericsson.com">magnus.westerlund@ericsson.com</a>
----------------------------------------------------------------------

_______________________________________________
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>

--------------060308010508000706060201--

From mary.ietf.barnes@gmail.com  Tue May 22 11:54:05 2012
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Subject: [rtcweb] CLUE WG interim meeting before RTCWEB meeting
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Hi,

This is a reminder that we did schedule the CLUE WG interim meeting to
facilitate attendance of folks that are also active in RTCWEB as there
is overlap when it comes to multi-stream.  The best we could do was to
schedule the CLUE meeting on June 7th and 8th (the Thursday and Friday
the week before the W3C WebRTC and IETF RTCWEB meetings).  This
results in folks that want to attend both to spend a weekend in
Stockholm - whether you consider that a good thing or not, a break
between two fairly intense meetings is not a bad thing, otherwise, we
would end up more fried than we do after a week of IETF meetings.

Anyways, as a co-chair of CLUE, I strongly encourage folks to try to
attend the CLUE meeting.  We have a doodle setup so that the logistics
can be properly arranged:
http://doodle.com/wwrdycma9ymrdn3r
Ericsson is also hosting this meeting so it will be at the same
location as the WebRTC & RTCWEB meetings.

Regards,
Mary Barnes
CLUE WG co-chair

From martin.thomson@gmail.com  Tue May 22 15:13:07 2012
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Pedantic review of draft-ietf-rtcweb-security-02
... with reference to draft-ietf-rtcweb-security-arch-01

Both documents are in pretty good shape.  They a largely coherent and
comprehensive.  That's not to say that they are finished.

To some extent, a lot of these issues are simply due to entropy.  We've
made quite a few decisions in the time since these were last updated.
We've also made no progress on other issues.


HTTP/HTTPS

S3, last paragraph before S3.1:

   [...], but realistically many sites do not run
   HTTPS [RFC2818] and so our ability to defend against network
   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.  The 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.  Otherwise, this is a bit of a scary statement
to just drop in without support:
   Note:  this issue is not restricted to PAGES
   which contain mixed content.  If a page from a given origin ever
   loads mixed content then it is possible for a network attacker to
   infect the browser's notion of that origin semi-permanently.))

NETWORK ATTACKERS

Same paragraph:

   [...], with the assumption that
   protection against network attackers is provided by running HTTPS.

Thankfully, the draft does not make this assumption.  Attacks on HTTP
are out of scope, but we still need to deal with network attackers.

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).  The implication from
S4.1 is that this is sufficient as well as necessary.  There is one
crucial piece of the argument that is absent:

   A site with access to camera or microphone could send media either to
   itself or any site that indicates consent (see CORS).  Sending media
   over HTTP or thewebsocketprotocol is likely to perform less well than
   is ideal, but it would work.

Therefore, it's easy to draw the implicit conclusions of the draft,
namely:
  1. a receipt consent mechanism like CORS is necessary, and
  2. user consent for access to input devices is necessary _and_
     sufficient...for this mode.

PRIVATE COMMUNICATIONS AND CONSENT

The above assumes that the site has access to media.  That is,
permission for input devices is being granted to the site.  However, 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.

This changes the assumptions - and the nature of the consent -
dramatically.  S4.1.2 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.  And UI.  From S4.1.2:

   Naturally, it is somewhat challenging to design UI
   primitives which express this sort of policy.

Complications include group calling.  How does the site ask for
permissions to talk to "a@b.com" and "x@y.net"?  How does such a
privilege persist?  Does it even make sense to persist at all?
Obviously, a conference of any significant size tends toward having a
bridge.  At that point, input devices are most likely granted to
"host.example.com".

TRULY PRIVATE COMMUNICATIONS

Keeping the site out of the loop requires that the browser lock down
access to media recording.  The trick is to ensure that the media is
unmodified all the way from source to sink.  It'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.  That requires a verifiable assertion
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.  My initial thought is that it was not
especially useful in this context.  I can imagine work-arounds that
would enable features that depend on visibility such as recording where
authenticity is also desirable.

IDENTIFICATION

S4.1.1.1.1.1.1.1.1 asks the obvious question:

   [...] there is a question about whether the
   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.  This is
exactly what you get when you use SRTP security descriptions (and also
EKT, if you allow the site to insert keys).

USER INTERACTION CONSIDERATIONS

S4.1.1.2 hides two important UI considerations:

   [...] great care must be taken in the design of this interface
   to avoid the users just clicking through

and

   [...] the user
   interface chrome must clearly display elements showing that the call
   is continuing in order to avoid attacks where the calling site just
   leaves it up indefinitely

These are both massively important.  If 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:
 a) the limitations of consent mechanisms
 b) providing appropriate feedback

For the latter, this can get complicated.  I recall a discussion of the
geolocation API that lead to the conclusion that no UI feedback would be
provided.  I still believe that this was a poor choice.  This case bears
a certain amount of similarity to that discussion.  I should really join
that media capture group...

In any case, this probably needs at least basic treatment here, even if
that is just by reference.

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.  That is, there is
something like an IdP in use.  Calling a site (as opposed to a person)
is very much a case where the usual domain trust anchors are perfectly
adequate.  The 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.  Any site that deployed TLSA (c.f. DANE WG) would require extra
checking.

I'm surprised to see nothing on this particular use case.  It's a
particularly useful one.

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.  Site
and ad networks work very hard to ensure that advertisements are
appropriate to the context in which they are shown.  Realtime 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.  In part, this is the same problem we already
have.  In addition, sites need to take some responsibility for making
any necessary distinction sufficiently clear on their own interface.  In
this context, the burden lies with the ad network.

((Aside.  I'm not sure that this is always true:
   [...] sites which host
   advertisements often learn very little about whether individual users
   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.  If the ads are served inline, then I would have to
assume that obfuscation of ad network content is the only protection
against the site learning about ad content and user interactions.))

COMMUNICATIONS CONSENT

The idea that a target for a media flow consents is not binary in the
sense that ICE establishes.  Even 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.  Because this
information comes from a web attacker, the browser cannot trust this
information.  The browser has to rate-limit Binding requests based on
its own information.  In 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.  The 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.  Plus, if we are going to offer a
data channel that doesn't use RTCP, then that option isn't available.

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.

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...

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

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

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.

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

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.  Initially, I thought that it might be
sufficient to prevent modifications to the session once a page goes
rogue.  At 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.  However, 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.  For
instance, EKT might allow the attacker to update keys.  Given that it
should also be possible to insert a relay (in one direction at least) by
providing faked "candidates", this cracks it wide open.  The more
renegotiation capabilities exist on the media path, the worse this is.

ICE TRANSACTION ID

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

ICE KEEPALIVES

Are not sufficient, so don't require them.  Instead, require a repeat of
the connectivity check.  That 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.

UNLINKABILITY

security-arch@S5.5

Nothing on this in the security doc.  What is the goal here?

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

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Subject: Re: [rtcweb] Strawman agenda and documents deadline for interim
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We (I speak for the webrtc chairs!) would like to see a session on the 
data channel. In addition to the protocol drafts that should be 
discussed (draft-ietf-rtcweb-data-channel and 
draft-jesup-rtcweb-data-protocol), the webrtc WG has a session on the 
data API on Monday, and most likely there will be things discovered in 
that session that must be brought forward to the rtcweb WG.

Stefan for the webrtc chairs.

On 05/09/2012 05:52 PM, Ted Hardie wrote:
> Howdy,
>
> We would like document updates for the interim to be complete by June
> 4th, so that folks have time to review them before the interim
> meeting.  Slides associated with the documents should be in by June
> 9th.  Below is the current strawman agenda; please comment.
>
> regards,
>
> Ted, Cullen, Magnus
>
> Day 1
>
> - Use Cases (2 H)
> * New use cases needing additional discussion if not resolved on
> mailing list.
> * Use cases that require cloning (yes or no?)
>
>
> - WebRTC's RTP Usage (3 H)
> * Topologies that need to be supported
> - what can be at one end of peer connection and what can be at other end
> * Walk through of all the pieces suggested
> - how to map media streams to rtp session.
>   -- example: describe how retransmission SSRCs relate to main SSRCs
> - mapping between RTP and what happens in JS
>
> - JSEP
> * Discussion of State Machine
> * Interactions allowed
> * SDP specification
>   -- mapping between SDP and what happens in JS
>
> - Security
> * Continued Consent
> * Identity and IdP
> * Details of Security Architecture
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Strawman agenda and documents deadline for interim
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On Thu, May 24, 2012 at 5:47 AM, Stefan Hakansson LK
<stefan.lk.hakansson@ericsson.com> wrote:
> We (I speak for the webrtc chairs!) would like to see a session on the da=
ta
> channel. In addition to the protocol drafts that should be discussed
> (draft-ietf-rtcweb-data-channel and draft-jesup-rtcweb-data-protocol), th=
e
> webrtc WG has a session on the data API on Monday, and most likely there
> will be things discovered in that session that must be brought forward to
> the rtcweb WG.
>
> Stefan for the webrtc chairs.
>

Thanks for the input, Stefan; do you have a sense of how much time you
think is needed?

regards,

Ted Hardie

>
> On 05/09/2012 05:52 PM, Ted Hardie wrote:
>>
>> Howdy,
>>
>> We would like document updates for the interim to be complete by June
>> 4th, so that folks have time to review them before the interim
>> meeting. =A0Slides associated with the documents should be in by June
>> 9th. =A0Below is the current strawman agenda; please comment.
>>
>> regards,
>>
>> Ted, Cullen, Magnus
>>
>> Day 1
>>
>> - Use Cases (2 H)
>> * New use cases needing additional discussion if not resolved on
>> mailing list.
>> * Use cases that require cloning (yes or no?)
>>
>>
>> - WebRTC's RTP Usage (3 H)
>> * Topologies that need to be supported
>> - what can be at one end of peer connection and what can be at other end
>> * Walk through of all the pieces suggested
>> - how to map media streams to rtp session.
>> =A0-- example: describe how retransmission SSRCs relate to main SSRCs
>> - mapping between RTP and what happens in JS
>>
>> - JSEP
>> * Discussion of State Machine
>> * Interactions allowed
>> * SDP specification
>> =A0-- mapping between SDP and what happens in JS
>>
>> - Security
>> * Continued Consent
>> * Identity and IdP
>> * Details of Security Architecture
>> _______________________________________________
>> 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 ted.ietf@gmail.com  Thu May 24 09:03:28 2012
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Subject: [rtcweb] Updated draft agenda for the 83.5 interim
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Here's an updated draft agenda for the 83.5 interim meeting.  After
each document is discussed, the chairs will be looking for reviewers
who can commit to review the updated document prior to Vancouver.  We
will also be looking to update the milestones on each document
discussed; we'll ask for each document as we go, then reconfirm or
reconcile at the end of the meeting.

If you can volunteer to be a reviewer now, that would be much
appreciated.  Note-takers and jabber-room volunteers are also needed,
and early volunteers for any of the above are promised an extra cookie
at every break.  If you can volunteer but will not be physically
present at the meeting, a suitable non-cookie alternative will be
found.

Please send agenda comments to the list,

thanks,

Ted, Cullen, and Magnus

June 12, 2012

Administrivia 30 minutes
Use Cases      2 hours
RTP                 3 hours
JSEP             90 minutes

Admin            30 minutes
JSEP                1 hour
Data Channel 90 minutes
ICE DTLS          1 hour
Security              3 hours

From stefan.lk.hakansson@ericsson.com  Fri May 25 02:07:22 2012
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Subject: Re: [rtcweb] Comments on draft-ietf-rtcweb-use-cases-and-requirements-07
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On 05/16/2012 03:31 AM, Cullen Jennings wrote:
>
> Cone and symmetric nat are probably not what exactly what we mean
> here - perhaps just refer to it as the various types of NATs
> described in RFC 4787.

The text was just lifted in (I have close to zero knowledge); referring 
to RFC 4787 sounds like a good idea to me.

>
> Might want to add to some use case that when A calls B, and B does
> not reveal their IP address to A.

Based on the discussion so far, I agree that this should be added somehow.

>
> Like to add Alice calls Bob with an encrypted media. Bob can tell the
> that the encrypted media is from Alice and not from an MITM

Sounds non controversial to add to me.

>
> Like to add ability to make a call in a situation where even small
> sounds from microphone should be sent to the other side and not
> filtered out. Emergency calls are examples of this as are some games
> and "jam session"

I think this is already covered in the "Distr. music band" , with req A17.

>
> Like to add case where application has one two video streams but one
> is far more important than the other. Should be a way to make sure
> that preferential treatment is given the the important stream over
> the less important streams.

What is there now is F24: The browser MUST be able to take advantage of
                    capabilities to prioritize voice and video
                    appropriately.
This talks only about priority of audio and video relative to other 
traffic, perhaps we should add something about prioritizing between 
different audio/video flows as well.

>
> In F22, "the IVR" ->  "an DTMF based IVR"

Agree.

>
> I have a hard time deciding what level the requirements should be at.
> It seems likely that given the level they are currently at, the
> requirements are somewhat incomplete. As long as we treat these as
> partial set of requirements derived from the use cases, I'm fine with
> that.
>
>
> New use case:
>
> Imagine a service like webex that facilitates collaboration between
> two companies called A and B. Webex has no need to see the content of
> the communication from A to B. So webex might run the web server that
> helps set up a RTCWeb session between A and B, but webex needs to be
> able to write it's code such that it does not get the keying material
> used to encrypt the media from A to B and the companies need to be
> able to verify that webex did not get the keys. Webex has no need to
> see the media from A to B and if it can provide this sort of promise
> that it does not get the media, it makes it much easier for a
> customer (say Juniper) to use webex and know the contents of their
> calls are not being sold or used by a competitor. Given how much
> WebRTC will be used by cloud services, this is an important feature.
> When installing the "webex" application in the browser and granting
> permissions to the application, it would be good to have one of the
> permissions be if the JS got access to the media or keying material
> for it to enforce this promise.
>
> Note this use case is not phrase to say this is the only way it can
> work. This is not mutually exclusive with they type of use cases
> described by Skype folks where there is an explicitly desire to make
> sure the calling service does have the keys to decrypt the media.
>
>
>
>
>
>
>
> _______________________________________________ rtcweb mailing list
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb


From fluffy@iii.ca  Fri May 25 09:02:45 2012
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From: Cullen Jennings <fluffy@iii.ca>
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To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
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Subject: Re: [rtcweb] Interim participation registration
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It does not matter for webex.  I think it is slightly nice for the =
chairs and others to have an who will be remote and who will be there in =
person just so they know if the people that have been involved in a =
given mailing list discussion will be "present" in some form. I will be =
remote. I guess next time we can make the google poll have three =
options. Cullen

On May 22, 2012, at 6:09 AM, Romascanu, Dan (Dan) wrote:

> Maybe it is important for Webex, if you plan to use Webex for remote =
participation. Or other tool. I do not know. In any case, you would need =
a separate doodle or a different option on the current doodle.=20
>=20
> Regards,
>=20
> Dan
>=20
>=20
>=20
>=20
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Tuesday, May 22, 2012 3:07 PM
>> To: Romascanu, Dan (Dan)
>> Cc: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Interim participation registration
>>=20
>> On 2012-05-22 13:18, Romascanu, Dan (Dan) wrote:
>>> Do you need a count only for people who plan to attend on-site?
>>=20
>> Good question. Yes, the this was for physical participation. We know
>> that we will have some number of remote participation. I don't know =
how
>> important it is to get a head count for them.
>>=20
>> Cheers
>>=20
>> Magnus
>>=20
>>>=20
>>> Dan
>>>=20
>>>=20
>>>=20
>>>=20
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>> Behalf Of Magnus Westerlund
>>>> Sent: Monday, May 21, 2012 4:31 PM
>>>> To: rtcweb@ietf.org
>>>> Subject: [rtcweb] Interim participation registration
>>>>=20
>>>> Hi,
>>>>=20
>>>> I would like to have a firm understanding of how many that really
>> will
>>>> show up for either day of the interim meeting. The main purpose is
>> to
>>>> ensure that break snacks and lunch are provided in the right =
amount.
>>>>=20
>>>> So please fill in this doodle to register your participation for
>> each
>>>> of
>>>> the days that you will be present.
>>>>=20
>>>> http://www.doodle.com/x3z6uhsy4q4s6fc4
>>>>=20
>>>> Thanks
>>>>=20
>>>> Magnus Westerlund
>>>> Interim meeting host
>>>>=20
>>>>=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
>>>> _______________________________________________
>>>> 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Cullen Jennings <fluffy@iii.ca>
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Subject: [rtcweb] Call for Papers: IAB/IRTF Workshop on Congestion Control for	Interactive Real-Time Communication, July 28, 2012 Vancouver, Canada
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>=20
>=20
> IAB / IRTF Workshop on
> Congestion Control for Interactive
> Real-Time Communication
> July 28, 2012
> Vancouver, Canada
> =20
> 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 (see =
http://www.ietf.org/meeting/84/index.html).  Participation at the =
workshop is free of charge. There is no requirement to either register =
with or attend the IETF-84 meeting that follows the workshop.
> =20
> The workshop organizers would like to foster a discussion on:
> What are appropriate congestion signals to use for interactive media =
and data?
> What existing congestion control algorithms are appropriate for =
interactive media and data? What properties would be desirable in new =
congestion control algorithms?
> Measurement and/or simulations of new congestion signals (e.g., =
delay-based) and their interaction with existing congestion control =
mechanisms.
> What are good available techniques for adjusting sending rates for =
interactive media and data? What are the limits of those techniques? =
What properties would be desirable in new techniques?
> What application-specific considerations have to be taken into =
account?
> How can we ensure that real-time communications are well-behaved with =
respect to other Internet applications while still providing good =
quality?
> What should the IETF and/or IRTF do?
> The organizers seek position papers on any or all of these topics, as =
well as other topics related to congestion control for interactive =
realtime media.
> =20
> Every prospective workshop participant must submit a position paper =
containing a name and an email address. 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.  Additional details =
about the meeting venue will be provided to authors of accepted papers.
> Important Dates
>=20
> Position paper submission deadline:          June 23, 2012
> Notification to paper authors:                       June 30, 2012
> Workshop date:                                             July 28, =
2012
> Additional Details
>=20
> Additional details on the workshop as well as the submission process =
is available at http://www.iab.org/cc-workshop/
> Contact
>=20
> To sponsors: 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!
>=20
> In case of questions please send email to mary.ietf.barnes at =
gmail.com.
> =20


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<html><head></head><body style=3D"word-wrap: break-word; =
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"><div><br></div><div><blockquote type=3D"cite"><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px;"><span =
style=3D"font-family:'Helvetica'; =
font-size:medium;"><br></span></div><br><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; font-family: Helvetica; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: -webkit-auto; =
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word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
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auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
lang=3D"EN-US" link=3D"blue" vlink=3D"purple" style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div class=3D"WordSection1" style=3D"page: =
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margin-left: 0in; font-size: 24pt; font-family: 'Times New Roman', =
serif; margin-top: 0in; margin-bottom: 0.0001pt; text-align: center; =
"><span style=3D"font-size: 18pt; font-family: Arial, sans-serif; color: =
black; ">IAB / IRTF Workshop on<br>Congestion Control for =
Interactive<br>Real-Time Communication</span><o:p></o:p></h1><h2 =
align=3D"center" style=3D"margin-right: 0in; margin-left: 0in; =
font-size: 18pt; font-family: 'Times New Roman', serif; margin-top: 0in; =
margin-bottom: 0.0001pt; text-align: center; "><span style=3D"font-size: =
14.5pt; font-family: Arial, sans-serif; color: black; ">July 28, =
2012<br>Vancouver, Canada</span><o:p></o:p></h2><div style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; color: =
black; "><o:p>&nbsp;</o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; "><span style=3D"font-size: =
11.5pt; font-family: Arial, sans-serif; color: black; ">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 (see<span =
class=3D"Apple-converted-space">&nbsp;</span></span><a =
href=3D"http://www.ietf.org/meeting/83/index.html" style=3D"color: blue; =
text-decoration: underline; "><span style=3D"font-size: 11.5pt; =
font-family: Arial, sans-serif; color: rgb(17, 85, 204); =
">http://www.ietf.org/meeting/84/index.html</span></a><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; color: =
black; ">). &nbsp;Participation at the workshop is free of charge. There =
is no requirement to either register with or attend the IETF-84 meeting =
that follows the workshop.<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; ">The workshop organizers would like to foster =
a discussion on:</span><span style=3D"font-size: 12pt; font-family: =
'Times New Roman', serif; "><o:p></o:p></span></div><ol start=3D"1" =
type=3D"1" style=3D"margin-bottom: 0in; "><li class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; ">What are =
appropriate congestion signals to use for interactive media and =
data?<o:p></o:p></span></li><li class=3D"MsoNormal" style=3D"margin-top: =
0in; margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; =
font-size: 11pt; font-family: Calibri, sans-serif; color: black; =
vertical-align: baseline; "><span style=3D"font-size: 11.5pt; =
font-family: Arial, sans-serif; ">What existing congestion control =
algorithms are appropriate for interactive media and data? What =
properties would be desirable in new congestion control =
algorithms?<o:p></o:p></span></li><li class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; =
">Measurement and/or simulations of new congestion signals (e.g., =
delay-based) and their interaction with existing congestion control =
mechanisms.<o:p></o:p></span></li><li class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; ">What are =
good available techniques for adjusting sending rates for interactive =
media and data? What are the limits of those techniques? What properties =
would be desirable in new techniques?<o:p></o:p></span></li><li =
class=3D"MsoNormal" style=3D"margin-top: 0in; margin-right: 0in; =
margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; font-family: =
Calibri, sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; ">What =
application-specific considerations have to be taken into =
account?<o:p></o:p></span></li><li class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; ">How can we =
ensure that real-time communications are well-behaved with respect to =
other Internet applications while still providing good =
quality?<o:p></o:p></span></li><li class=3D"MsoNormal" =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; color: black; vertical-align: baseline; "><span =
style=3D"font-size: 11.5pt; font-family: Arial, sans-serif; ">What =
should the IETF and/or IRTF do?<o:p></o:p></span></li></ol><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; ">The organizers seek position papers on any =
or all of these topics, as well as other topics related to congestion =
control for interactive realtime media.<o:p></o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; "><o:p>&nbsp;</o:p></span></div><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; ">Every prospective workshop participant must =
submit a position paper containing a name and an email address. 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.&nbsp; Additional details about the meeting venue will be =
provided to authors of accepted papers.<o:p></o:p></span></div><h2 =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 18pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
14.5pt; font-family: Arial, sans-serif; color: black; ">Important =
Dates</span><o:p></o:p></h2><div style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; "><span style=3D"font-size: 11.5pt; =
font-family: Arial, sans-serif; color: black; ">Position paper =
submission deadline: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
June 23, 2012</span><br><span style=3D"font-size: 11.5pt; font-family: =
Arial, sans-serif; color: black; ">Notification to paper authors: =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; June 30, =
2012</span><br><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; ">Workshop date: =
&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;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; July 28, =
2012</span><o:p></o:p></div><h2 style=3D"margin-right: 0in; margin-left: =
0in; font-size: 18pt; font-family: 'Times New Roman', serif; "><span =
style=3D"font-size: 14.5pt; font-family: Arial, sans-serif; color: =
black; ">Additional Details<o:p></o:p></span></h2><div =
style=3D"margin-top: 0in; margin-right: 0in; margin-left: 0in; =
margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, =
sans-serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; ">Additional details on the workshop as well =
as the submission process is available at<span =
class=3D"Apple-converted-space">&nbsp;</span><a =
href=3D"http://www.iab.org/cc-workshop/" style=3D"color: blue; =
text-decoration: underline; =
">http://www.iab.org/cc-workshop/</a><o:p></o:p></span></div><h2 =
style=3D"margin-right: 0in; margin-left: 0in; font-size: 18pt; =
font-family: 'Times New Roman', serif; "><span style=3D"font-size: =
14.5pt; font-family: Arial, sans-serif; color: black; =
">Contact<o:p></o:p></span></h2><h2 style=3D"margin-right: 0in; =
margin-left: 0in; font-size: 18pt; font-family: 'Times New Roman', =
serif; "><span style=3D"font-size: 11.5pt; font-family: Arial, =
sans-serif; color: black; font-weight: normal; ">To sponsors: 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!</span><span style=3D"font-weight: normal; =
"><o:p></o:p></span></h2><div style=3D"margin-top: 0in; margin-right: =
0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: 11pt; =
font-family: Calibri, sans-serif; "><span style=3D"font-size: 11.5pt; =
font-family: Arial, sans-serif; color: black; ">In case of questions =
please send email to mary.ietf.barnes at<span =
class=3D"Apple-converted-space">&nbsp;</span><a href=3D"http://gmail.com/"=
 style=3D"color: blue; text-decoration: underline; =
">gmail.com</a>.<o:p></o:p></span></div><div style=3D"margin-top: 0in; =
margin-right: 0in; margin-left: 0in; margin-bottom: 0.0001pt; font-size: =
11pt; font-family: Calibri, sans-serif; =
"><o:p>&nbsp;</o:p></div></div></div></span></blockquote></div><br></body>=
</html>=

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From: "DRUTA, DAN" <dd5826@att.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: Review of draft-ietf-rtcweb-security-02 and draft-ietf-rtcweb-security-arch-01 documents
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Subject: [rtcweb] Review of draft-ietf-rtcweb-security-02 and draft-ietf-rtcweb-security-arch-01 documents
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I reviewed the latest version of the drafts and below are my comments/sugge=
stions/corrections (not as comprehensive as Mark's).
In general I think the documents are well written with good references and =
paint a rather comprehensive picture of the issues to be addressed for rtcw=
eb.

Draft-ietf-rtcweb-security-02

1.Introduction -  second paragraph after the diagram:=20
"Each of their browsers exposes standardized JavaScript calling APIs which =
are used by the Web server to set up a call between Alice and Bob."
 This statement can be a bit more clear in that the APIs are executed local=
ly (in the user agent) as  java script downloaded from the Web Server. The =
next paragraph eludes to the fact that logic is controlled by the calling s=
ervice but I think it will be important to distinguish between APIs impleme=
nted by the user-agent and APIs/functions implemented by the calling servic=
e.

4.1 Access to Local Devices.=20
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 determ=
ine the real reason they give the consent to a site. Take the scenario in w=
hich a site obtains consent from the user for an app that captures a clip a=
nd 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 purpo=
se of streaming without the user knowing that. This can be confusing for us=
ers even if they trust the site.

4.2 IP location Privacy
"Note that as soon as the callee sends their ICE candidates, the callee lea=
rns the callee's IP addresses."=20
This must be "the caller learns the callee's IP address"

4.3.2.3. Third Party Identity
"It is possible (see[I-D.rescorla-rtcweb-generic-idp]) to use systems of th=
is type to authenticate RTCWEB calls, linking them to existing user notions=
 of
identity (e.g., Facebook adjacencies). Calls which are authenticated in thi=
s fashion are naturally resistant even to active MITM attack by the calling=
 site."

Not to get too much into semantics, I think it is important to make the dis=
tinction that users are authenticated (using whatever IdP) and calls are au=
thorized based on the assertions provided by the Idp for the authenticated =
user.

Draft-ietf-rtcweb-security-arch-01

1.Introduction
"The Real-Time Communications on the Web (RTCWEB) working group is tasked w=
ith standardizing protocols for real-time communications between Web browse=
rs."

While there's nothing inaccurate about the statement, I would like to make =
it more generic because it is more than just communication between web brow=
sers. In my opinion the WEB part in the acronym is all about using the web =
technologies to enable real time communications.

A proposed edited definition would be "... standardization protocols for en=
abling real-time communications within user-agents using web technologies (=
e.g JavaScript)".


6.2 Privacy

I have been trying to identify the appropriate section for a rather controv=
ersial requirement:  Lawful Intercept. It is expected that service provider=
s for RTCWeb/WebRTC will be required to provide support to lawful intercept=
 and I think we need to capture these requirements so we can map them to th=
e security architecture.=20


Thanks,
Dan

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Subject: Re: [rtcweb] Review of draft-ietf-rtcweb-security-02 and draft-ietf-rtcweb-security-arch-01 documents
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On 5/29/12 4:26 PM, DRUTA, DAN wrote:
> 6.2 Privacy
> 
> I have been trying to identify the appropriate section for a rather controversial requirement:  Lawful Intercept. It is expected that service providers for RTCWeb/WebRTC will be required to provide support to lawful intercept and I think we need to capture these requirements so we can map them to the security architecture. 

IETF Policy on Wiretapping
http://tools.ietf.org/html/rfc2804

Fabio

From harald@alvestrand.no  Tue May 29 08:51:15 2012
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Subject: [rtcweb] Fwd: New Version Notification for draft-alvestrand-rtcweb-msid-02.txt
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As directed in the Paris IETF meeting, this draft now tries to be a more 
generic mechanism with a specific specified application to media streams.

It is still marked with "comments to RTCWEB, please", but it is up to 
the chairs which group (if any) should adopt it as a WG item.

                   Harald

-------- Original Message --------
Subject: 	New Version Notification for draft-alvestrand-rtcweb-msid-02.txt
Date: 	Tue, 29 May 2012 08:46:55 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no



A new version of I-D, draft-alvestrand-rtcweb-msid-02.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-alvestrand-rtcweb-msid
Revision:	 02
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2012-05-29
WG ID:		 Individual Submission
Number of pages: 10

Abstract:
    This document specifies a grouping mechanism for RTP media streams
    that can be used to specify relations betweeen media streams within
    different RTP sessions.

    This mechanism is used to signal the association between the RTP
    concept of SSRC and the WebRTC concept of&quot;media stream&quot; /&quot;media
    stream track&quot; using SDP signalling.

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





The IETF Secretariat



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    As directed in the Paris IETF meeting, this draft now tries to be a
    more generic mechanism with a specific specified application to
    media streams.<br>
    <br>
    It is still marked with "comments to RTCWEB, please", but it is up
    to the chairs which group (if any) should adopt it as a WG item.<br>
    <br>
    Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â  Harald<br>
    <br>
    -------- Original Message --------
    <table class="moz-email-headers-table" border="0" cellpadding="0"
      cellspacing="0">
      <tbody>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-alvestrand-rtcweb-msid-02.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Tue, 29 May 2012 08:46:55 -0700</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">To: </th>
          <td><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new version of I-D, draft-alvestrand-rtcweb-msid-02.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-alvestrand-rtcweb-msid
Revision:	 02
Title:		 Cross Session Stream Identification in the Session Description Protocol
Creation date:	 2012-05-29
WG ID:		 Individual Submission
Number of pages: 10

Abstract:
   This document specifies a grouping mechanism for RTP media streams
   that can be used to specify relations betweeen media streams within
   different RTP sessions.

   This mechanism is used to signal the association between the RTP
   concept of SSRC and the WebRTC concept of &amp;quot;media stream&amp;quot; / &amp;quot;media
   stream track&amp;quot; using SDP signalling.

   This document is an input document for discussion.  It should be
   discussed in the RTCWEB WG list, <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>.


                                                                                  


The IETF Secretariat

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From cb.list6@gmail.com  Tue May 29 10:02:43 2012
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Is there yet a pointer on how a SIM / IMS based identity will work
with the webRTC web server for authentication?

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.

Thoughts?  Pointers?

Thanks,

CB

From martin.thomson@gmail.com  Tue May 29 10:17:02 2012
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That's assuming that you want to authenticate as a mobile identity.
Why would it not be sufficient to authenticate as a linked identity?

That is, I am "john.smith@example.com" and some of my calls originate
or terminate at
"+1-234-555-6789".

Can you expand on your use case a little?

On 29 May 2012 10:02, Cameron Byrne <cb.list6@gmail.com> wrote:
> Is there yet a pointer on how a SIM / IMS based identity will work
> with the webRTC web server for authentication?
>
> 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.
>
> Thoughts? =C2=A0Pointers?
>
> Thanks,
>
> CB
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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On Tue, May 29, 2012 at 10:17 AM, Martin Thomson
<martin.thomson@gmail.com> wrote:
> That's assuming that you want to authenticate as a mobile identity.
> Why would it not be sufficient to authenticate as a linked identity?
>
> That is, I am "john.smith@example.com" and some of my calls originate
> or terminate at
> "+1-234-555-6789".
>
> Can you expand on your use case a little?
>

Basic use case is exposing mobile operator IMS services via webrtc.

Such that webrtc is an IMS clients of sorts and the various IMS
services are exposed via webrtc just as they are exposed to an IMS
client (voice, video, presence, IM, ...).

Using the mobile identity to register into the IMS domain seems to be
the simplest fit if authentication is possible. From a deployment and
operational ease perspective, it sure would be nice if nothing had to
change in the IMS cloud and webrtc was just a proxy / client for
accessing the IMS services.

CB

> On 29 May 2012 10:02, Cameron Byrne <cb.list6@gmail.com> wrote:
>> Is there yet a pointer on how a SIM / IMS based identity will work
>> with the webRTC web server for authentication?
>>
>> 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.
>>
>> Thoughts? =A0Pointers?
>>
>> Thanks,
>>
>> CB
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb

From martin.thomson@gmail.com  Tue May 29 10:26:32 2012
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On 29 May 2012 10:22, Cameron Byrne <cb.list6@gmail.com> wrote:
> Using the mobile identity to register into the IMS domain seems to be
> the simplest fit if authentication is possible. From a deployment and
> operational ease perspective, it sure would be nice if nothing had to
> change in the IMS cloud and webrtc was just a proxy / client for
> accessing the IMS services.

How your clients authenticate with your service is largely not a
problem for this working group...

That said, if you want to authenticate as an IMS UE, then you need to
be able to offer the credentials that a UE would offer.  So why not
proxy the authentication requests too?  Then the server doesn't need
to store credentials.

From rbarnes@bbn.com  Tue May 29 10:29:05 2012
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Subject: Re: [rtcweb] authenticating against IMS HSS/HLR
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To answer your question directly, I don't think there's a current =
document.

However, ISTM that the HSS/HLR could just act as an Identity Provider =
[1], for example by layering BrowserID [2] on top of the IMS:
-- UE generates key pair locally
-- UE authenticates to HSS/HLR
-- UE presents public key to HSS/HLR
-- HSS/HLR provides a certificate containing that public key
-- UE presents that certificate to the remote party
-- Remote party validates certificate under HSS/HLR public key

The main question would then be what identifiers a relying party would =
allow an HSS/HLR to vouch for.  For SIP URIs, there's no problem, =
because the domain of the HSS/HLR would be clear (the domain part of the =
URI).  But for telephone numbers, the HSS/HLR would have to be a "third =
party" IdP, which is problematic [3].

Hope this helps,
--Richard

[1] <http://tools.ietf.org/html/draft-rescorla-rtcweb-generic-idp-00>
[2] =
<http://tools.ietf.org/html/draft-rescorla-rtcweb-generic-idp-00#section-5=
.5.1>
[3] Slide 5 of =
<http://tools.ietf.org/agenda/83/slides/slides-83-rtcweb-2.pdf>


On May 29, 2012, at 1:02 PM, 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 igor.faynberg@alcatel-lucent.com  Tue May 29 10:36:26 2012
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Yes, but Dan's proposal surely takes this into consideration as I read it.

To this end, I think we need to ensure that the requirements that we 
develop do not interfer with Lawful Intercept requirements that 
providers must implement. (And to effect that we don't have to touch the 
LI requirements per se.)

Igor

On 5/29/2012 10:47 AM, Fabio Pietrosanti (naif) wrote:
> On 5/29/12 4:26 PM, DRUTA, DAN wrote:
>> 6.2 Privacy
>>
>> I have been trying to identify the appropriate section for a rather controversial requirement:  Lawful Intercept. It is expected that service providers for RTCWeb/WebRTC will be required to provide support to lawful intercept and I think we need to capture these requirements so we can map them to the security architecture.
> IETF Policy on Wiretapping
> http://tools.ietf.org/html/rfc2804
>
> Fabio
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Cameron,

Indeed, there is a lot of interesting work done in this field. (A few 
thousand messages back you will find some postings on the subject, 
including mine.)  A few references will follow, but first--the problem.

The problem is the access to "the SIM secret key."  The latter is 
unavailable by design: all ISIM operations are performed by the card. 
IMS itself builds on the AKA protocol, which establishes the necessary 
keys. From there, 3GPP recommends bootstrapping using the Generic 
Bootstrapping Architecture. 3GPP defines this in detail in 3GPP TS 33.220.

3GPP has a technical report, TR 33.924, which specifies an 
implementation of OpenID with GBA. One problem here is  client 
deployment.  Until GB function is deployed in the Client, it is, of 
course, impossible to use any solution that depends on it.

Hence the alternative solution: to use AKA. My colleagues and I have 
described a way to do so in a Bell Labs Technical Journal paper 
http://onlinelibrary.wiley.com/doi/10.1002/bltj.20426/abstract. 
Subsequently, a method to implement OpenID/AKA interworking has been 
specified by ITU-T in   Recommendation Y.2722 NGN Identity Management 
Mechanisms. (All ITU-T Recommendations as well as 3GPP documents are 
available on their respective sites.)

But the latter solution is still a non-solution as long as there is no 
client API for interactions with the SIM card. And this was the 
conclusion of a discussion on this list a while ago...

We need either such an API or GBA deployment to progress.

Igor




On 5/29/2012 1:02 PM, Cameron Byrne wrote:
> Is there yet a pointer on how a SIM / IMS based identity will work
> with the webRTC web server for authentication?
>
> 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.
>
> Thoughts?  Pointers?
>
> Thanks,
>
> CB
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--------------020409090702060102060306
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      http-equiv="Content-Type">
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  <body text="#000000" bgcolor="#ffffff">
    Cameron,<br>
    <br>
    Indeed, there is a lot of interesting work done in this field. (A
    few thousand messages back you will find some postings on the
    subject, including mine.)&nbsp; A few references will follow, but
    first--the problem.<br>
    <br>
    The problem is the access to "the SIM secret key."&nbsp; The latter is
    unavailable by design: all ISIM operations are performed by the
    card. IMS itself builds on the AKA protocol, which establishes the
    necessary keys. From there, 3GPP recommends bootstrapping using the
    Generic Bootstrapping Architecture. 3GPP defines this in detail in
    3GPP TS 33.220.&nbsp; <br>
    <br>
    3GPP has a technical report, <span style="color: rgb(34, 34, 34);
      font-family: arial,sans-serif; font-size: 16px; font-style:
      normal; font-variant: normal; font-weight: normal; letter-spacing:
      normal; line-height: normal; orphans: 2; text-indent: 0px;
      text-transform: none; white-space: normal; widows: 2;
      word-spacing: 0px; background-color: rgb(255, 255, 255); display:
      inline ! important; float: none;">TR 33.924, which specifies an
      implementation of OpenID with GBA. </span>One problem here is&nbsp;
    client deployment.&nbsp; Until GB function is deployed in the Client, it
    is, of course, impossible to use any solution that depends on it.&nbsp; <br>
    <span style="color: rgb(34, 34, 34); font-family: arial,sans-serif;
      font-size: 16px; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-indent: 0px; text-transform: none; white-space:
      normal; widows: 2; word-spacing: 0px; background-color: rgb(255,
      255, 255); display: inline ! important; float: none;"></span><br>
    Hence the alternative solution: to use AKA. My colleagues and I have
    described a way to do so in a Bell Labs Technical Journal paper <a
href="http://onlinelibrary.wiley.com/doi/10.1002/bltj.20426/abstract">http://onlinelibrary.wiley.com/doi/10.1002/bltj.20426/abstract</a>.
    Subsequently, a method to implement OpenID/AKA interworking has been
    specified by ITU-T in &nbsp; Recommendation&nbsp;<span style="color: rgb(34,
      34, 34); font-family: arial,sans-serif; font-size: 16px;
      font-style: normal; font-variant: normal; font-weight: normal;
      letter-spacing: normal; line-height: normal; orphans: 2;
      text-indent: 0px; text-transform: none; white-space: normal;
      widows: 2; word-spacing: 0px; background-color: rgb(255, 255,
      255); display: inline ! important; float: none;"> Y.2722 NGN
      Identity Management Mechanisms</span>. (All ITU-T Recommendations
    as well as 3GPP documents are available on their respective sites.)<br>
    <br>
    But the latter solution is still a non-solution as long as there is
    no client API for interactions with the SIM card. And this was the
    conclusion of a discussion on this list a while ago...<br>
    <br>
    We need either such an API or GBA deployment to progress.<br>
    <br>
    Igor<br>
    <br>
    <br>
    <span style="color: rgb(34, 34, 34); font-family: arial,sans-serif;
      font-size: 16px; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-indent: 0px; text-transform: none; white-space:
      normal; widows: 2; word-spacing: 0px; background-color: rgb(255,
      255, 255); display: inline ! important; float: none;"></span><br>
    <br>
    On 5/29/2012 1:02 PM, Cameron Byrne wrote:
    <blockquote
cite="mid:CAD6AjGSRw8ScGk-H08rL3bfmWMrpqGVgDA2HSdUgkG8bxKF9yQ@mail.gmail.com"
      type="cite">
      <pre wrap="">Is there yet a pointer on how a SIM / IMS based identity will work
with the webRTC web server for authentication?

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.

Thoughts?  Pointers?

Thanks,

CB
_______________________________________________
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>
  </body>
</html>

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On 29 May 2012 07:26, DRUTA, DAN <dd5826@att.com> wrote:
> I reviewed the latest version of the drafts and below are my comments/sug=
gestions/corrections (not as comprehensive as Mark's).

:)  (see From: header)

> 4.1 Access to Local Devices.
> Since this describes the threat I think is important to point out the fac=
t that even when the user provides consent it is difficult for them to dete=
rmine 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 permanen=
t consent to the site, the site can obtain access to the camera for the pur=
pose of streaming without the user knowing that. This can be confusing for =
users even if they trust the site.

Once a site has your camera, there's no fundamental difference between
still frame capture and video streaming (for example).  Streaming is
just an optimization of the recording interface already present.  A
site can acquire still images and upload them (using JSONP, HTTP form
post, whatever) one by one.  Streaming just means that they can do so
without clogging up my network connection with 30 JPEGs per second.

I had to think about this one a bit before understanding why this
isn't a problem.  Maybe that can be rectified by adding text.

The more challenging interface problem deals with the presentation of
permissions dialogs for specific remote users, not the browser.  The
security architecture draft hints that this might be necessary, but it
offers a far more challenging UI design problem than the above.

> 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"

addresses :)

> 6.2 Privacy
>
> I have been trying to identify the appropriate section for a rather contr=
oversial requirement: =C2=A0Lawful Intercept. It is expected that service p=
roviders for RTCWeb/WebRTC will be required to provide support to lawful in=
tercept and I think we need to capture these requirements so we can map the=
m to the security architecture.

Every time I think of this, I am more convinced that there is nothing
that needs to be specified.  Interestingly, the IdP proposal
potentially moves the onus for meeting these requirements to the IdP,
... impersonation being the key.

From harald@alvestrand.no  Tue May 29 13:43:53 2012
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Subject: Re: [rtcweb] authenticating against IMS HSS/HLR
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On 05/29/2012 07:02 PM, Cameron Byrne wrote:
> Is there yet a pointer on how a SIM / IMS based identity will work
> with the webRTC web server for authentication?
Is there a standard Javascript API defined for access to SIM-based 
identity services?

Once that exists, it seems to be all a Small Matter Of Javascript.
As long as that does not exist, we seem to be building without a foundation.


From igor.faynberg@alcatel-lucent.com  Tue May 29 14:16:54 2012
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Subject: Re: [rtcweb] authenticating against IMS HSS/HLR
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Right, and I would not even require that it be Javascript.  Say, it is 
device-specific so only the browser can use it.

Igor

On 5/29/2012 4:43 PM, Harald Alvestrand wrote:
> On 05/29/2012 07:02 PM, Cameron Byrne wrote:
>> Is there yet a pointer on how a SIM / IMS based identity will work
>> with the webRTC web server for authentication?
> Is there a standard Javascript API defined for access to SIM-based 
> identity services?
>
> Once that exists, it seems to be all a Small Matter Of Javascript.
> As long as that does not exist, we seem to be building without a 
> foundation.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Wed May 30 01:54:02 2012
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Subject: [rtcweb] Regarding the use-case document
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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

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Hi,

RFC 4169 defines digest-AKA, which is how SIM-card based AKA authentication=
 is used within the context of HTTP digest. It could be used for user authe=
ntication for HTTP after TLS has been setup.=20

There are some platforms that enable SIM access to at least some special ap=
plications. Presumably browser could be one of them and a Javascript API co=
uld be also exposed to apps if needed.

The web server would need a Radius interface to operator HLR/HSS.

I don't know if anything else would be needed.

Markus


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of ext Cameron Byrne
>Sent: 29 May, 2012 20:03
>To: rtcweb@ietf.org
>Subject: [rtcweb] authenticating against IMS HSS/HLR
>
>Is there yet a pointer on how a SIM / IMS based identity will work with th=
e
>webRTC web server for authentication?
>
>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 usernam=
e
>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 w=
eb
>server or someplace outside the HSS/HLR for binding with the username
>password.
>
>Thoughts?  Pointers?
>
>Thanks,
>
>CB
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From pravindran@sonusnet.com  Wed May 30 02:31:48 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: Stefan Hakansson LK <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Regarding the use-case document
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Subject: Re: [rtcweb] Regarding the use-case document
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Stefan,


1) I have mentioned in Call Center [2] usecase calls for "Anonymous" identi=
ty 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 discus=
sed  as part of http://www.ietf.org/mail-archive/web/rtcweb/current/msg0425=
4.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. =20

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

From goran.ap.eriksson@ericsson.com  Wed May 30 05:06:58 2012
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From: =?iso-8859-1?Q?G=F6ran_Eriksson_AP?= <goran.ap.eriksson@ericsson.com>
To: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 30 May 2012 14:06:54 +0200
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Hi Cullen

Inline.

G=F6ran=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
> Cone and symmetric nat are probably not what exactly what we=20
> mean here - perhaps just refer to it as the various types of=20
> NATs described in RFC 4787.=20
>=20
> Might want to add to some use case that when A calls B, and B=20
> does not reveal their IP address to A.
>=20
> Like to add Alice calls Bob with an encrypted media. Bob can=20
> tell the that the encrypted media is from Alice and not from an MITM
>=20
> Like to add ability to make a call in a situation where even=20
> small sounds from microphone should be sent to the other side=20
> and not filtered out. Emergency calls are examples of this as=20
> are some games and "jam session"
>=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

I agree that this is a relevant area to address to secure that WebRTC is co=
mpetitive compared to
apps done in native OS's, especiallin in enterprise context.

The current use case document states something like "being able to use prio=
rity functions in network nodes".
This is vague since it touches many matters such as whether to put audio an=
d video on the same IP-flow or have them in different
ones, which is essential when considering mapping on LTE radio channels; th=
e potential use of DS to identify flows, which
may be relevant in enterprise context, etc, how to cater for treament of st=
uff on the datagram channel, multihoming consequences, etc.

I am not sure however how far into such matters we should go in this WG, bu=
t given the importance of making a
good solution working across OS's and browser vendors and access technologi=
es, I am leaning towards support a discussions about such
details in this WG for instance using the use case document.

What is Your take of this?=20

>=20
> In F22, "the IVR" -> "an DTMF based IVR"
>=20
> I have a hard time deciding what level the requirements=20
> should be at. It seems likely that given the level they are=20
> currently at, the requirements are somewhat incomplete. As=20
> long as we treat these as partial set of requirements derived=20
> from the use cases, I'm fine with that.=20
>=20
>=20
> New use case:
>=20
> Imagine a service like webex that facilitates collaboration=20
> between two companies called A and B. Webex has no need to=20
> see the content of the communication from A to B. So webex=20
> might run the web server that helps set up a RTCWeb session=20
> between A and B, but webex needs to be able to write it's=20
> code such that it does not get the keying material used to=20
> encrypt the media from A to B and the companies need to be=20
> able to verify that webex did not get the keys. Webex has no=20
> need to see the media from A to B and if it can provide this=20
> sort of promise that it does not get the media, it makes it=20
> much easier for a customer (say Juniper) to use webex and=20
> know the contents of their calls are not being sold or used=20
> by a competitor. Given how much WebRTC will be used by cloud=20
> services, this is an important feature. When installing the=20
> "webex" application in the browser and granting permissions=20
> to the application, it would be good to have one of the=20
> permissions be if the JS got access to  the media or keying=20
> material for it to enforce this promise.=20
>=20
> Note this use case is not phrase to say this is the only way=20
> it can work. This is not mutually exclusive with they type of=20
> use cases described by Skype folks where there is an=20
> explicitly desire to make sure the calling service does have=20
> the keys to decrypt the media.=20
>=20
>=20
>=20
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

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On 5/30/2012 5:14 AM, Markus.Isomaki@nokia.com wrote:
> ...
> There are some platforms that enable SIM access to at least some special applications. Presumably browser could be one of them and a Javascript API could be also exposed to apps if needed.

Could you please supply a reference to any such platform?
> The web server would need a Radius interface to operator HLR/HSS.

Actually, it would have been Diameter, but in reality the web server 
cannot (and should not) talk to HSS directly; it will talk to an entity 
that maps the IMS identity to OpenID or whatever other identity is used.

Igor
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of ext Cameron Byrne
>> Sent: 29 May, 2012 20:03
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] authenticating against IMS HSS/HLR
>>
>> Is there yet a pointer on how a SIM / IMS based identity will work with the
>> webRTC web server for authentication?
>>
>> 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.
>>
>> Thoughts?  Pointers?
>>
>> Thanks,
>>
>> CB
>> _______________________________________________
>> 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] authenticating against IMS HSS/HLR
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On Tue, May 29, 2012 at 11:16 AM, Igor Faynberg
<igor.faynberg@alcatel-lucent.com> wrote:
> Cameron,
>
> Indeed, there is a lot of interesting work done in this field. (A few
> thousand messages back you will find some postings on the subject, includ=
ing
> mine.)=A0 A few references will follow, but first--the problem.
>
> The problem is the access to "the SIM secret key."=A0 The latter is
> unavailable by design: all ISIM operations are performed by the card. IMS
> itself builds on the AKA protocol, which establishes the necessary keys.
> From there, 3GPP recommends bootstrapping using the Generic Bootstrapping
> Architecture. 3GPP defines this in detail in 3GPP TS 33.220.
>
> 3GPP has a technical report, TR 33.924, which specifies an implementation=
 of
> OpenID with GBA. One problem here is=A0 client deployment.=A0 Until GB fu=
nction
> is deployed in the Client, it is, of course, impossible to use any soluti=
on
> that depends on it.
>

Thanks, this is probably the pointer i need.

GBA is already available to me and we have it deployed in the network
and on devices.  I will need to do more reading on this and how it all
really works.  The goal is that the IMS service is completely
abstracted from the UE and the IMS services and associated identities
can be fully instantiated on any devices that has a WebRTC browser
(smart tv, wifi-only tablet, refrigerator...)

Cameron

> Hence the alternative solution: to use AKA. My colleagues and I have
> described a way to do so in a Bell Labs Technical Journal paper
> http://onlinelibrary.wiley.com/doi/10.1002/bltj.20426/abstract.
> Subsequently, a method to implement OpenID/AKA interworking has been
> specified by ITU-T in =A0 Recommendation=A0 Y.2722 NGN Identity Managemen=
t
> Mechanisms. (All ITU-T Recommendations as well as 3GPP documents are
> available on their respective sites.)
>
> But the latter solution is still a non-solution as long as there is no
> client API for interactions with the SIM card. And this was the conclusio=
n
> of a discussion on this list a while ago...
>
> We need either such an API or GBA deployment to progress.
>
> Igor
>
>
>
>
>
> On 5/29/2012 1:02 PM, Cameron Byrne wrote:
>
> Is there yet a pointer on how a SIM / IMS based identity will work
> with the webRTC web server for authentication?
>
> 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.
>
> Thoughts?  Pointers?
>
> Thanks,
>
> CB
> _______________________________________________
> 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 stefan.lk.hakansson@ericsson.com  Thu May 31 00:27:59 2012
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Subject: Re: [rtcweb] Regarding the use-case document
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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


From Jim.Barnett@genesyslab.com  Thu May 31 06:05:23 2012
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From: "Jim Barnett" <Jim.Barnett@genesyslab.com>
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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. =20

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=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 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=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 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 comm =

>> service (made by Tim Terriberri) [3]
>>
>> 3. Comments related to security, SRTP and key exchange, F20 (by Dan=20
>> 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 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 streams =

>> New use-case: "Collaboration between companies with node [in media=20
>> 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=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 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 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 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 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 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 to access=20
>> 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=20
>> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb

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

From tim@phonefromhere.com  Thu May 31 06:31:43 2012
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From: Tim Panton <tim@phonefromhere.com>
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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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Much as I hate to re-open this, I think there is a significant use-case =
for Agents using webRTC,
especially in pop-up (e.g. campaign based) call centers.

This would (potentially  be the reverse of the case you described, with =
webRTC on the inside,=20
probably still talking through the ACD/gateway.

So, yes, this merits some further discussion in my view too.

T.

On 31 May 2012, at 14:04, 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. =20
>=20
> 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.
>=20
> - Jim
>=20
> -----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
>=20
> On 05/30/2012 11:31 AM, Ravindran, Parthasarathi wrote:
>> Stefan,
>>=20
>>=20
>> 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)
>>=20
>> 2) Also in another mail thread, "Anonymous" identity for customer is=20=

>> discussed  as part of=20
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg04254.html
>>=20
>> In case any more information required for clarity, please let me =
know.=20
>> Also, I'm fine with discussion about these usecase in interim =
meeting.
>=20
> Hi Partha,
>=20
> 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.
>=20
> 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.
>=20
> 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.
>=20
> Br,
> Stefan
>=20
>>=20
>> Thanks Partha
>>=20
>>> -----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
>>>=20
>>> The rtcweb chairs asked for input on the use-case document on April=20=

>>> 27th [1].
>>>=20
>>> Browsing the emails that followed, I note:
>>>=20
>>> 1. Proposal for a call center use case (made by Jim Barnett) [2]
>>>=20
>>> 2. Proposal to add peer-to-peer file sharing to the simple video =
comm=20
>>> service (made by Tim Terriberri) [3]
>>>=20
>>> 3. Comments related to security, SRTP and key exchange, F20 (by Dan=20=

>>> Wing) [4]
>>>=20
>>> 4. Request for clarifying "eavesdropping", Andrew Hutton [5]
>>>=20
>>> 5. Enterprise policies use-cases proposed, Roman Shpount [6]
>>>=20
>>> 6. Multiparty with low complex central node, Stefan H=E5kansson [7]
>>>=20
>>> 7. Multiparty with central node that can not decipher media Oscar=20
>>> Ohlsson [8]
>>>=20
>>> 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 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 =
streams=20
>>> New use-case: "Collaboration between companies with node [in media=20=

>>> path] that does/can not decipher media"
>>>=20
>>> 9. Proposal from Magnus Westerlund to clarify the language a bit =
[10]=20
>>> regarding 4. in this list
>>>=20
>>> Is anything important missing from the list above?
>>>=20
>>> Going through this list (and the discussion that followed on each
>>> item) it seems quite clear that:
>>>=20
>>> A. The language should be clarified (as Magnus proposed [10]), e.g.
>>> to clarify that also the data channel should be encrypted (this=20
>>> relates to 3. and 9. in the previous list)
>>>=20
>>> 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 mentioned many=20=

>>> times during the MV interim)
>>>=20
>>> 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
>>>=20
>>> D. For the new requirements proposed by Cullen (8a - d in the=20
>>> previous list), 8a and 8b can be incorporated in existing 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 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 reduces send=20=

>>> rate for different flows as a response to e.g. network congestion.
>>> The reqs seem  uncontroversial to me.
>>>=20
>>> Unless there are objections to this, the editors plan to update the=20=

>>> document according to A - D above.
>>>=20
>>>=20
>>>=20
>>> 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 to access=20=

>>> deciphered media [8] and [9]; I think this should be discussed at =
the=20
>>> interim meeting.
>>>=20
>>> Comments?
>>>=20
>>> Stefan
>>>=20
>>> [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=20=

>>> 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


From fluffy@iii.ca  Thu May 31 08:01:52 2012
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References: <B236B24082A4094A85003E8FFB8DDC3C1A45AE47@SOM-EXCH04.nuance.com> <4FB2E8FC.40404@thaumas.net> <CABcZeBPWiGYaD6yBtBzeZMNgnz20JnHTyVHTev71KZagGRiqHw@mail.gmail.com> <B236B24082A4094A85003E8FFB8DDC3C1A45AFA0@SOM-EXCH04.nuance.com> <4FB34CAD.80103@ericsson.com>
To: Milan Young <Milan.Young@nuance.com>, public-media-capture@w3.org
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Subject: Re: [rtcweb] TCP vs UDP for media
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You should take this as a requirement to the public-media-capture@w3.org =
list (added to thread). It seem reasonable to want to specify which =
codec the data returned from getusermedia should use.=20

On May 16, 2012, at 12:43 AM, Magnus Westerlund wrote:

> On 2012-05-16 03:37, Young, Milan wrote:
>> Yes, existing web technologies for transport would work fine.  In
>> fact I wrote a demo based on WebSockets, the AudioAPI, and
>> getUserMedia.  But it's a bit cludgy and only transmits PCM audio.
>>=20
>> Would my use case for live access to an encoded media stream be a
>> good fit for a revised MediaStreamRecorder?  Would this group the
>> right place to host such an effort?
>=20
> I think this is something you should take to the W3C. As it appear to =
be
> primarily an API question. Getting access to the content in the JS
> application or request the browser to record to a local file which =
later
> can be uploaded it would not be in this WG.
>=20
> Cheers
>=20
> Magnus Westerlund
> WG chair
>=20
>>=20
>> Thanks
>>=20
>> -----Original Message----- From: Eric Rescorla [mailto:ekr@rtfm.com]
>> Sent: Tuesday, May 15, 2012 4:47 PM To: Ralph Giles Cc: Young,
>> Milan; rtcweb@ietf.org Subject: Re: [rtcweb] TCP vs UDP for media
>>=20
>> On Tue, May 15, 2012 at 4:38 PM, Ralph Giles <giles@thaumas.net>
>> wrote:
>>> On 12-05-15 4:22 PM, Young, Milan wrote:
>>>=20
>>>> I'm wondering if any thought has been given to TCP as a media
>>>> transport.
>>>=20
>>> Where low latency transmission isn't an issue, one can generally
>>> fall back to established TCP-based protocols, like HTTP streaming
>>> and websockets, so we haven't really worried about that angle in
>>> the context of webrtc.
>>>=20
>>> Recording the stream is a requirement, so probably something could
>>> be built using that, merging recordings from each endpoint to fix
>>> up any dropped packets.
>>=20
>> Agreed. This seems like something that could be done using just=20
>> getUserMedia() plus existing Web technologies.
>>=20
>> -Ekr _______________________________________________ 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
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

