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Subject: [rtcweb] New Milestones for RTCWEB WG
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WG,

As raised in the WG meeting last week we plan to submit new milestones.
These is to make the time line a bit more realistic to actually be meet.
They also try to adjust to the picked document structure.

Here they are:
Oct 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements),
Overview (draft-ietf-rtcweb-overview), Security and Privacy Documents
(I-D) to W3C

Dec 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements)
document to IESG for publication as Informational

Apr 2012 Send W3C input on what requirements the current solution has on
API(s)

May 2012 Send Overview (draft-ietf-rtcweb-overview), Security and
Privacy Model documents to IESG for publication as Informational

May 2012 Send Signalling and Negotiation, and NAT Traversal to IESG for
publication as Proposed Standard

Jun 2012 Send Media Transport, Media Processing, and Codecs to IESG for
publication as Proposed Standard

Jun 2012 Send Datagram Transport for non-media data to IESG for
publication as Proposed Standard

These will soon be submitted for official publication. If you have any
issues please raise them on the list and we can discuss them.

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 a.bakker@vu.nl  Wed Aug  3 01:36:11 2011
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Subject: [rtcweb] New use case: Large multiparty session
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Hi

as proposed in the July 26th meeting I'd like to put forward a new use
case: assume a client is involved in a large multiparty session but is
unable (or unwilling) to upload his signals to all participants.

This suggests forwarding support from the network is required, e.g.
some relay server, multicast, or the future IETF peer-to-peer streaming
protocol (PPSP).

Thanks,
     Arno Bakker

From partr@cisco.com  Wed Aug  3 05:12:36 2011
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Thread-Topic: [rtcweb] New Milestones for RTCWEB WG
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References: <4E37EAAB.3020708@ericsson.com>
From: "Parthasarathi R (partr)" <partr@cisco.com>
To: "Magnus Westerlund" <magnus.westerlund@ericsson.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] New Milestones for RTCWEB WG
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Hi Magnus,=20

The current RTCWeb overview, usecase & requirement document does not =
have architecture diagrams. Please clarify which one of the RTCWeb WG =
item below will cover RTCWeb architecture.

Thanks
Partha=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
> Sent: Tuesday, August 02, 2011 5:47 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] New Milestones for RTCWEB WG
>=20
> WG,
>=20
> As raised in the WG meeting last week we plan to submit new=20
> milestones.
> These is to make the time line a bit more realistic to=20
> actually be meet.
> They also try to adjust to the picked document structure.
>=20
> Here they are:
> Oct 2011 Send Use Cases=20
> (draft-ietf-rtcweb-use-cases-and-requirements),
> Overview (draft-ietf-rtcweb-overview), Security and Privacy Documents
> (I-D) to W3C
>=20
> Dec 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements)
> document to IESG for publication as Informational
>=20
> Apr 2012 Send W3C input on what requirements the current=20
> solution has on
> API(s)
>=20
> May 2012 Send Overview (draft-ietf-rtcweb-overview), Security=20
> and Privacy Model documents to IESG for publication as Informational
>=20
> May 2012 Send Signalling and Negotiation, and NAT Traversal=20
> to IESG for publication as Proposed Standard
>=20
> Jun 2012 Send Media Transport, Media Processing, and Codecs=20
> to IESG for publication as Proposed Standard
>=20
> Jun 2012 Send Datagram Transport for non-media data to IESG=20
> for publication as Proposed Standard
>=20
> These will soon be submitted for official publication. If you=20
> have any issues please raise them on the list and we can discuss them.
>=20
> cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM
> ----------------------------------------------------------------------
> Ericsson AB                | Phone  +46 10 7148287
> F=E4r=F6gatan 6                | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

From randell-ietf@jesup.org  Wed Aug  3 05:52:20 2011
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Subject: Re: [rtcweb] New use case: Large multiparty session
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On 8/3/2011 4:36 AM, Arno Bakker wrote:
> Hi
>
> as proposed in the July 26th meeting I'd like to put forward a new use
> case: assume a client is involved in a large multiparty session but is
> unable (or unwilling) to upload his signals to all participants.
>
> This suggests forwarding support from the network is required, e.g.
> some relay server, multicast, or the future IETF peer-to-peer streaming
> protocol (PPSP).

So the use case is:

User is part of a multiparty (3 or greater) session, but for one of 
several reasons (such as
available upstream bandwidth)  cannot send media to all other 
participants ("mesh" conferencing).
This requires that another node, either a central network node (such as 
a conferencing server)
or another member in the session, relay or mix the user's media to 
distribute to the rest of the
members of the session.

-- 
Randell Jesup
randell-ietf@jesup.org


From a.bakker@vu.nl  Wed Aug  3 06:37:25 2011
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Subject: Re: [rtcweb] New use case: Large multiparty session
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On 03/08/2011 14:51, Randell Jesup wrote:
> On 8/3/2011 4:36 AM, Arno Bakker wrote:
 >>
>> assume a client is involved in a large multiparty session but is
>> unable (or unwilling) to upload his signals to all participants.
>>
>> This suggests forwarding support from the network is required, e.g.
>> some relay server, multicast, or the future IETF peer-to-peer streaming
>> protocol (PPSP).
>
> So the use case is:
>
> User is part of a multiparty (3 or greater) session, but for one of
> several reasons (such as
> available upstream bandwidth) cannot send media to all other
> participants ("mesh" conferencing).
> This requires that another node, either a central network node (such as
> a conferencing server)
> or another member in the session, relay or mix the user's media to
> distribute to the rest of the
> members of the session.
>


Yes. Although your wording suggests a single relay node (central or 
member) is used. Please leave the option of true peer-to-peer 
distribution (or IP multicast) of the signal open.

CU,
      Arno

From randell-ietf@jesup.org  Wed Aug  3 07:12:27 2011
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On 7/18/2011 2:30 PM, Cullen Jennings wrote:
> I'd like to add an use case along the following lines...
>
> Rohan works for international aid agency and deploys wireless networks =
in places where disasters have wiped out all the existing infrastructure.=
 There's just been an earthquake in Haiti. Betty works for Canada Red Cro=
ss and need to get the Red Cross location in Haiti connected to other aid=
 groups in Haiti to help arrange complex logistics of things like what or=
der various aid agencies are even going to use the limited resources of t=
he runway at the airport. Rohan and Betty both use the AidNet web applica=
tion for communicating and have cached copies using the HTML5 offline sup=
port on their computers or, if they can't get their computers into the co=
untry, can at least get copy in via USB key and find a local computer wit=
h a browser. Rohan sets up a wireless access point on the tallest thing h=
e can find and managed to provide wireless coverage for many square miles=
=2E Betty connects to that and manages to call Rohan to coordinate and le=
t Rohan know the location of RedCross so that better wireless can be arra=
nged there.
>
> This one more or less one of the use cases for P2PSIP work to be able t=
o set up an ad hoc voice communications system before one even had access=
 to DNS server or any global internet infrastructure. I think that the wh=
ole P2PSIP system could be done inside browser (or a system similar to it=
). The underling transport of P2PSIP is very close to the proposal here i=
ncluding the use of ICE. One would probably need to define a new transpor=
t that an Reload overlay could use but other than that, I think one could=
 meet this use case using something along the lines of Reload running in =
browser.

This is an interesting use-case - how would this really work in terms of =

discovery (does it require some external application to handle=20
discovery?), what support would it likely need in the browser, and what=20
are the security implications?

--=20
Randell Jesup
randell-ietf@jesup.org



From randell-ietf@jesup.org  Wed Aug  3 07:23:35 2011
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>  A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirements-01.txt

>4.2.6.  Multiparty video communication
>
>4.2.6.1.  Description
>
>    In this use case the simple video communication service is extended
>    by allowing multiparty sessions.  No central server is involved - the
>    browser of each participant sends and receives streams to and from
>    all other session participants.  The web application in the browser
>    of each user is responsible for setting up streams to all receivers.
>
>    The audio sent by each participant is a mono stream.  However, in
>    order to enhance intelligibility, the web application pans the audio
>    from different participants differently when rendering the audio.
>    This is done automatically, but users can change how the different
>    participants are placed in the (virtual) room.


Note that this covers the request from Arno for peer-to-peer distribution.

Also, I would remove the audio is mono reference.  I see no need to add 
that restriction.

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Rohan Red Cross Use Case
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On 7/18/2011 11:30 AM, Cullen Jennings wrote:
> I'd like to add an use case along the following lines...
>
> Rohan works for international aid agency and deploys wireless networks =
in places where disasters have wiped out all the existing infrastructure.=
 There's just been an earthquake in Haiti. Betty works for Canada Red Cro=
ss and need to get the Red Cross location in Haiti connected to other aid=
 groups in Haiti to help arrange complex logistics of things like what or=
der various aid agencies are even going to use the limited resources of t=
he runway at the airport. Rohan and Betty both use the AidNet web applica=
tion for communicating and have cached copies using the HTML5 offline sup=
port on their computers or, if they can't get their computers into the co=
untry, can at least get copy in via USB key and find a local computer wit=
h a browser. Rohan sets up a wireless access point on the tallest thing h=
e can find and managed to provide wireless coverage for many square miles=
=2E Betty connects to that and manages to call Rohan to coordinate and le=
t Rohan know the location of RedCross so that
>    better wireless can be arranged there.
>
> This one more or less one of the use cases for P2PSIP work to be able t=
o set up an ad hoc voice communications system before one even had access=
 to DNS server...

If Rohan can't figure out how to include a DNS resolver that is=20
advertised via DHCP with some locally-resolvable names (and, if needed,=20
a web landing page that has some basic info about what to configure and=20
where to get help) then Rohan shouldn't be in charge of setting up the=20
wireless access point.

This is of course the same argument we had years ago on the original=20
p2psip list.

Matthew Kaufman



From igor.faynberg@alcatel-lucent.com  Wed Aug  3 10:29:23 2011
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Subject: Re: [rtcweb] New use case: Large multiparty session
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I strongly support the use case, but I was under the impression that 
there has been a use case thar required a similar architecture: a 
"central" server (of course, it can be replicated) for accumulating 
input and distributing output.

For one thing, the music collaboration use case, I believe, calls for 
this architecture.

Perhaps one thing to watch for in the use case write-ups is separation 
between a use case and architecture. These are two different things 
because one architecture can fit many (and maybe all) use cases.

Igor

On 8/3/2011 8:51 AM, Randell Jesup wrote:
> On 8/3/2011 4:36 AM, Arno Bakker wrote:
>> Hi
>>
>> as proposed in the July 26th meeting I'd like to put forward a new use
>> case: assume a client is involved in a large multiparty session but is
>> unable (or unwilling) to upload his signals to all participants.
>>
>> This suggests forwarding support from the network is required, e.g.
>> some relay server, multicast, or the future IETF peer-to-peer streaming
>> protocol (PPSP).
>
> So the use case is:
>
> User is part of a multiparty (3 or greater) session, but for one of 
> several reasons (such as
> available upstream bandwidth)  cannot send media to all other 
> participants ("mesh" conferencing).
> This requires that another node, either a central network node (such 
> as a conferencing server)
> or another member in the session, relay or mix the user's media to 
> distribute to the rest of the
> members of the session.
>

From ted.ietf@gmail.com  Wed Aug  3 13:22:39 2011
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There is a poll at http://doodle.com/w9q95yfdunddn9r7 .  Please fill
it out with the dates you are available and times you prefer.  The
chairs realize that folks are on vacation, but in order to nail this
down somewhat, we plan on closing the poll August 10th.  We assume
we'll get a representative sample by then, even if we don't catch all
the potential attendees.

We're still working on the agenda, but we assume that we'll talk about
security, rtp muxing, and the non-real time data flows.

regards,

Ted, Cullen, Magnus

From partr@cisco.com  Wed Aug  3 16:59:46 2011
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Thread-Topic: RTCWeb architecture text requirement [Was RE: [rtcweb] New use case: Large multiparty session]
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From: "Parthasarathi R (partr)" <partr@cisco.com>
To: <igor.faynberg@alcatel-lucent.com>, <rtcweb@ietf.org>
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Subject: [rtcweb] RTCWeb architecture text requirement [Was RE: New use case: Large multiparty session]
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Hi all,

<snip>
>=20
> Perhaps one thing to watch for in the use case write-ups is=20
> separation between a use case and architecture. These are two=20
> different things because one architecture can fit many (and=20
> maybe all) use cases.
>=20
</snip>

As mentioned by Igor, I'm also looking for the architecture document
which provide the insight into RTCWeb deployment possibilities.

draft-rosenberg-rtcweb-framework-00 is one of the RTCWeb document which
is in the similar line.  I'm interested in hearing others opinion on
this.=20

Thanks
Partha=20


> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Igor Faynberg
> Sent: Wednesday, August 03, 2011 11:00 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] New use case: Large multiparty session
>=20
> I strongly support the use case, but I was under the=20
> impression that there has been a use case thar required a=20
> similar architecture: a "central" server (of course, it can=20
> be replicated) for accumulating input and distributing output.
>=20
> For one thing, the music collaboration use case, I believe,=20
> calls for this architecture.
>=20
> Perhaps one thing to watch for in the use case write-ups is=20
> separation between a use case and architecture. These are two=20
> different things because one architecture can fit many (and=20
> maybe all) use cases.
>=20
> Igor
>=20
> On 8/3/2011 8:51 AM, Randell Jesup wrote:
> > On 8/3/2011 4:36 AM, Arno Bakker wrote:
> >> Hi
> >>
> >> as proposed in the July 26th meeting I'd like to put forward a new=20
> >> use
> >> case: assume a client is involved in a large multiparty=20
> session but=20
> >> is unable (or unwilling) to upload his signals to all participants.
> >>
> >> This suggests forwarding support from the network is required, e.g.
> >> some relay server, multicast, or the future IETF peer-to-peer=20
> >> streaming protocol (PPSP).
> >
> > So the use case is:
> >
> > User is part of a multiparty (3 or greater) session, but for one of=20
> > several reasons (such as available upstream bandwidth)  cannot send=20
> > media to all other participants ("mesh" conferencing).
> > This requires that another node, either a central network=20
> node (such=20
> > as a conferencing server) or another member in the session,=20
> relay or=20
> > mix the user's media to distribute to the rest of the=20
> members of the=20
> > session.
> >
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

From a.bakker@vu.nl  Thu Aug  4 00:32:09 2011
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-01.txt
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On 03/08/2011 16:22, Randell Jesup wrote:
>> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirements-01.txt
>
>
>> 4.2.6. Multiparty video communication
>>
>
> Note that this covers the request from Arno for peer-to-peer distribution.
>

Yes. But perhaps we should rephrase requirement F11 to

"The browser MUST be able to transmit streams to
at least one peer"

(peer could be other browser or relay server) to more properly highlight 
the problem of insufficient upload in larger
sessions.

CU,
     Arno

From stefan.lk.hakansson@ericsson.com  Thu Aug  4 00:57:21 2011
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From: =?Windows-1252?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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All,
=20
sorry for cross-posting.
=20
As one of the editors for the use case doc, I would like to summarize the u=
se case discussion a bit. A and B below is my understanding of the status r=
egarding use cases after attending the webrtc meeting Sat July 23rd and the=
 first rtcweb session of IETF 81, and after trying to digest the mail lists=
. C discusses how we could move forward, D where we should discuss and fina=
lly E my opinion as an individual.

Input/feedback is appreciated.
=20
A. Existing (old) use cases=20
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
The use cases listed in http://datatracker.ietf.org/doc/draft-ietf-rtcweb-u=
se-cases-and-requirements/ where not questioned, all survived. However, the=
 document type should be changed.


B. New use cases
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
I noted (as not immediately dismissed) the following proposals:
=20
    1. Distributed music band, discussed at the webrtc meeting
    2. Use cases regarding different situations when being invited to a =93=
session=94, e.g. browser open, browser open but another tab active, browser=
 open but active in session, browser closed, =85. (Matthew Kaufman); discus=
sed at webrtc meeting
    3. Different TURN provider scenarios (Cullen Jennings); discussed at th=
e webrtc meeting
    4. More challenging NAT case (Matthew Kaufman); UDP blocked http://www.=
ietf.org/mail-archive/web/rtcweb/current/msg00468.html
    5. E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/rtcweb/cur=
rent/msg00525.html
    6. Local Recording (John Ewell) at webrtc meeting
    7. Remote recording (John)  http://www.ietf.org/mail-archive/web/rtcweb=
/current/msg00472.html
    8. Emergency access for disabled (Bernard Aboba) http://www.ietf.org/ma=
il-archive/web/rtcweb/current/msg00478.html
    9. Clue use cases (Roni Even) http://tools.ietf.org/html/draft-ietf-clu=
e-telepresence-use-cases-01
    10. Rohan red cross (Cullen Jennings); proposed a bit earlier http://ww=
w.ietf.org/mail-archive/web/rtcweb/current/msg00323.html

Probably I have missed a few.
=20
C. Way forward
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
The proper way forward would probably be to have people draft text describi=
ng these use cases and derive requirements they add, and then have a discus=
sion on whether they should be included for version one of the standard or =
not.
=20
In some cases the description exists already (Rohan red cross, emergency ac=
cess for disabled, clue use cases) but not requirements.
=20
Are there volunteers (maybe the ones proposing them)?

D. Where to discuss
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
This mail is sent to both public-webrtc at w3.org and to rtcweb at ietf.org=
. We have discussed that there should be only one use case document that is=
 common for the webrtc and the rtcweb WGs, possibly the current ietf use ca=
se doc (http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-req=
uirements/). Does this mean that it would be OK to discuss use cases on the=
 rtcweb list only, or should we cross post?

Some of the proposed new use cases (2, 6) have relatively little bearing on=
 IETF IMO; they are more W3C related, perhaps they should be documented and=
 discussed in a webrtc context only.


E. Opinion as individual
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
My personal opinion is that we at this stage should focus on the basic use =
cases, and solve those in a good way. To me that would mean that we should =
add 3 (to allow different providers) and 4 (if you can=92t connect no use c=
ase can be supported). I think that these use cases can be added as twists =
of/extensions to the first use case (Simple Video Communication Service).
=20
In my view it also means that we should add text and requirements for 2 to =
make sure that the communication capabilites we are adding to the browser i=
s usable in everyday scenarios. Presumable this would lead requirements (e.=
g. background execution capability) that are transferred to other W3C WGs -=
 not to requirements in the webrtc/rtcweb space.
=20
>From a W3C perspective it could also make sense to add a recording use case=
 (as it seems that other W3C WGs rely on webrtc to provide an API for recor=
ding).

The rest of the new proposed use cases, IMO, could be looked into in a seco=
nd stage when we've established the basics.

=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
=20
Comments?
=20
Stefan=

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As one of the webrtc chairs I have the generic comment that the sooner info=
rmation is available the better. In particular, webrtc is planning a f2f me=
eting at the W3C TPAC (Oct 31 and Nov 1). To get the most out of that meeti=
ng it would be good if the Use Cases and Sec/Priv documents are available, =
and if there can be some advance info on what reqs the "current solution" h=
as on the API(s) by then.

Cheers,
Stefan
________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Magnus=
 Westerlund [magnus.westerlund@ericsson.com]
Sent: Tuesday, August 02, 2011 2:16 PM
To: rtcweb@ietf.org
Subject: [rtcweb] New Milestones for RTCWEB WG

WG,

As raised in the WG meeting last week we plan to submit new milestones.
These is to make the time line a bit more realistic to actually be meet.
They also try to adjust to the picked document structure.

Here they are:
Oct 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements),
Overview (draft-ietf-rtcweb-overview), Security and Privacy Documents
(I-D) to W3C

Dec 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements)
document to IESG for publication as Informational

Apr 2012 Send W3C input on what requirements the current solution has on
API(s)

May 2012 Send Overview (draft-ietf-rtcweb-overview), Security and
Privacy Model documents to IESG for publication as Informational

May 2012 Send Signalling and Negotiation, and NAT Traversal to IESG for
publication as Proposed Standard

Jun 2012 Send Media Transport, Media Processing, and Codecs to IESG for
publication as Proposed Standard

Jun 2012 Send Datagram Transport for non-media data to IESG for
publication as Proposed Standard

These will soon be submitted for official publication. If you have any
issues please raise them on the list and we can discuss them.

cheers

Magnus Westerlund

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

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

From randell-ietf@jesup.org  Thu Aug  4 08:05:09 2011
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On 8/4/2011 3:57 AM, Stefan Håkansson LK wrote:

>
> B. New use cases
> ===========
> I noted (as not immediately dismissed) the following proposals:
>
>      1. Distributed music band, discussed at the webrtc meeting
>      2. Use cases regarding different situations when being invited to a “session”, e.g. browser open, browser open but another tab active, browser open but active in session, browser closed, …. (Matthew Kaufman); discussed at webrtc meeting
>      3. Different TURN provider scenarios (Cullen Jennings); discussed at the webrtc meeting
>      4. More challenging NAT case (Matthew Kaufman); UDP blocked http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
>      5. E911 (Paul Beaumont) http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
>      6. Local Recording (John Ewell) at webrtc meeting

Would this cover all "voicemail" and "videomail" cases?  I.e. having a client
accept connections in the background if the call is not answered, optionally
play a message, and record incoming audio/video, and allow the remote user to
interact with it.  Also remote playback of messages.  If not (and if it's not
covered by the current document, we need to add this usecase.

Note that there are two variants: one local (local client handles it, which
has more security issues around camera access and pre-approval), and one for
remote recording (after some time with no answer or if not "registered" call
is forwarded to a message server that answers it).  Note that there are
security identity issues here (key chains, etc).


>
>      7. Remote recording (John)  http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
>      8. Emergency access for disabled (Bernard Aboba) http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
>      9. Clue use cases (Roni Even) http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
>      10. Rohan red cross (Cullen Jennings); proposed a bit earlier http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
>
> Probably I have missed a few.

I'd add:


11. Remote assistance (ala VNC or RDP) - User is helping another user on
their computer with either view-only or view-with-control, either of just
the browser of the the entire screen.  Computer audio is optionally sent
as well.  They are optionally talking and/or viewing each other while this
is occurring.  They may be transferring files over a reliable data
connection during this session.


Commentary: How often have you talked to your father/mother/brother/etc
and tried to debug their computer problems remotely?  And getting them to
install a specific remote-assistance package, and to use it, and get it to
work through firewalls, can be very painful.  (There are other uses of this
ability to copilot as well, including classroom instruction, distance learning,
etc.)  This use-case enables someone to build a JS app to provide this
functionality.  (Note that for the W3 and browser vendors there will be
significant security issues to resolve.)  I think this is actually a fairly
important use-case.


Additional resulting requirements:

* Need to be able to select a "video" source other than a camera (window or
screen) (W3 issue)

* Need to be able to send multiple video and audio streams from different
sources (probably already covered, mostly W3 issue)

* Need to be able to use a reliable data connection (for mouse/keyboard/etc
control, plus file transfer)


12. Security camera/baby monitor usage - use a persistent or temporary
connection to provide basic security camera operation, including switching
cameras or mics, or with the ability to remotely request review of recent
data recorded.  Should be able to switch to 2-way audio and/or video remotely.


Additional requirements:

* Need to be able to select a "video" source other than a camera (file)
(W3 issue)

* Remote control of camera/mic selection and enabling (W3 issue) - may
require reliable data connection

* May need control from JS over codecs used, at least voice vs audio, etc,
and max video framerate/resolution/bandwidth (W3 issue largely?)


>
>
> E. Opinion as individual
> ===============
> My personal opinion is that we at this stage should focus on the basic use cases, and solve those in a good way. To me that would mean that we should add 3 (to allow different providers) and 4 (if you can’t connect no use case can be supported). I think that these use cases can be added as twists of/extensions to the first use case (Simple Video Communication Service).
>
> In my view it also means that we should add text and requirements for 2 to make sure that the communication capabilites we are adding to the browser is usable in everyday scenarios. Presumable this would lead requirements (e.g. background execution capability) that are transferred to other W3C WGs - not to requirements in the webrtc/rtcweb space.
>
> > From a W3C perspective it could also make sense to add a recording use case (as it seems that other W3C WGs rely on webrtc to provide an API for recording).
>
> The rest of the new proposed use cases, IMO, could be looked into in a second stage when we've established the basics.

I'd very much want to include the "remote assistance" case, and I think the
voicemail cases could be very important in overall acceptance and utility, especially
given the issues over whether someone's machine is running and accepting calls.

Security cam/baby-monitor is less important, but highlights some controls that
we may want to expose to the JS over maximum bitrate/framerate/resolution/etc,
and of course a slew of security issues for W3 to think about.

-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Thu Aug  4 08:38:30 2011
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Responding to myself:


> 11. Remote assistance (ala VNC or RDP) - User is helping another user on
> their computer with either view-only or view-with-control, either of just
> the browser of the the entire screen.  Computer audio is optionally sent
> as well.  They are optionally talking and/or viewing each other while this
> is occurring.  They may be transferring files over a reliable data
> connection during this session.
>
>
> Commentary: How often have you talked to your father/mother/brother/etc
> and tried to debug their computer problems remotely?  And getting them to
> install a specific remote-assistance package, and to use it, and get it to
> work through firewalls, can be very painful.  (There are other uses of this
> ability to copilot as well, including classroom instruction, distance learning,
> etc.)  This use-case enables someone to build a JS app to provide this
> functionality.  (Note that for the W3 and browser vendors there will be
> significant security issues to resolve.)  I think this is actually a fairly
> important use-case.

Other uses this use-case covers:

Meetecho - ability to use as a video source a window on your computer (i.e.
presentation), while also optionally exchanging or sending audio and/or video,
and optionally having a (reliable) data channel for side-chat ala the meetecho
chat window.  Note that this is most useful if the window can be any window, not
just the browser.

Very useful for business/sales presentations as well (could help avoid a whole
bunch of travel by sales engineers and the like).

I don't believe this would add any new requirements.


-- 
Randell Jesup
randell-ietf@jesup.org


From ekr@rtfm.com  Thu Aug  4 09:46:56 2011
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I wanted to follow up to something I said at the microphone last week
with regard to Cullen's presentation
(http://www.ietf.org/proceedings/81/slides/rtcweb-13.pdf) and in
particular slide 9, which argues that using a first-class signaling
protocol for the actual signaling has better security properties.  I
got on Cullen pretty hard about this, but in retrospect I think I may
have been a bit hasty.

The basic imperative for securing the camera and microphone in RTCWeb
is that you be able to determine who the media will go to (e.g., you
are talking to Ford.) What I had in my head in my slides was that we
would do this by restricting access to the JavaScript APIs. I.e., that
you would grant www.ford.com the right to use the WebRTC APIs. I.e.,
the policy would be:

  P1: Allow any script coming from www.ford.com to call anywhere.

The way (or at least a way) you use this to get a call is that
DoubleClick brings up an IFRAME on Ford's site that loads up JS
calling the right location at Ford. And since the browser has
direct access to the origin, it can evaluate this policy
directly. If you're not willing to have a hosted IFRAME like
this, then the permissions reduces to letting DC call alyone.


The model in slide 9, however, is different: instead of restricting
access to the APIs to a given site, it instead allows anyone to
invoke the APIs, but *calls* are restricted to a given site. So,
in this case, the access control policy would be:

  P2: Allow anyone to place a call provided that it goes to
  *@ford.com.

The browser evaluates this (in the simplest case) by connecting to
ford.com's SIP server and verifying that it is indeed ford.com.



My (rather loud) argument at the microphone was that P1 and P2 have
similar security properties and I was trying to push back on Cullen's
claim that P2 was better. In either case, ford.com gets to capture
your audio and video and send it wherever it wants. And obviously, if
it just wants to turn on your camera and send it to your ex-wife it
can. However, thinking about it more, I think that claim is
too strong; there are two differences, one technical and one social.

The technical difference is simply that in one case Ford has to stand
up a bunch of new technical infrastructure to serve/service the WebRTC
code that establishes its origin and initiates the call [0]. By contrast,
the solution Cullen proposes allows reuse of all the existing calling
technical infrastructure (whether SIP or Jingle) that Ford already
has.

The social difference is about the assertion that Ford is making.
In the former case, Ford is saying "I'll do something with this call,
and you'll like it." In the second they are saying "you are making
a call to someone at Ford." So, while from a technical security
guy perspective, they are equally able to screw you over in either
case by lying, if they *aren't* lying then the policy your browser
is enforcing on your behalf is a lot clearer: "only call people
who have Ford addresses."


At a higher level, I think we've been failing to draw the distinction
between two cases which are socially (and arguably technically) very
different:

1. The calling service is permitted to place a call but all the
   information about who you are actually calling is accessible
   only to the JS provided by the calling service.

2. The calling service is permitted to place a call but the
   browser has independent information about who you are
   actually talking to. [and could at least in principle render
   it to the user.]

Matthew's UI document and Cullen's presentation both point towards
the notion that #2 is valuable as well, with Cullen's presentation
being more oriented towards signaling and Matthew's more oriented
towards the media-level security, but in both cases suggesting
that we may not just want to authorize the calling service and
let it call wherever with no other independent security mechanisms.

I'm not sure whether this concept covers all the relevant cases, since
at least conceptually it depends on the notion that everyone has an
actual name that is somehow qualified by a universally meaningful name
and there are probably cases where I just want to phone home to the
Web site. However, it seems like it's at least plausibly a useful
piece of functionality in addition to what I had been thinking of as
an origin-based API restriction.

-Ekr


[0] It can of course outsource that with <script src="..."> or
Akamai, but then it loses control of the interaction.

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Eric Rescorla said:=20

>   P2: Allow anyone to place a call provided that it goes to
>   *@ford.com.
>=20
> The browser evaluates this (in the simplest case) by connecting to
> ford.com's SIP server and verifying that it is indeed ford.com

These two statements are *not* equivalent.  The first statement relates to =
the To: field=3B=20
the second relates to authentication of the identity of the server to which=
 the browser connects.
These elements are orthogonal and should not be conflated.=20

As an example=2C if a BOSH connection manager is configured to route stanza=
s to domains other than
its own=2C a browser can use a BOSH connection manager to send XMPP stanzas=
 to any
XMPP user.   Authenticating the BOSH connection manager (e.g. via TLS) impo=
ses no
restriction whatsoever on who can be "called" via Jingle using that BOSH co=
nnection manger.  =20
That restriction would only be implemented within the BOSH connection manag=
er=2C *not* within the=20
browser.=20
 		 	   		  =

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<html>
<head>
<style><!--
.hmmessage P
{
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body.hmmessage
{
font-size: 10pt=3B
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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
Eric Rescorla said: <br><br><div>&gt=3B   P2: Allow anyone to place a call =
provided that it goes to<br>&gt=3B   *@ford.com.<br>&gt=3B <br>&gt=3B The b=
rowser evaluates this (in the simplest case) by connecting to<br>&gt=3B for=
d.com's SIP server and verifying that it is indeed ford.com<br><br>These tw=
o statements are *not* equivalent.&nbsp=3B The first statement relates to t=
he To: field=3B <br>the second relates to authentication of the identity of=
 the server to which the browser connects.<br>These elements are orthogonal=
 and should not be conflated. <br><br>As an example=2C if a BOSH connection=
 manager is configured to route stanzas to domains other than<br>its own=2C=
 a browser can use a BOSH connection manager to send XMPP stanzas to any<br=
>XMPP user.&nbsp=3B&nbsp=3B Authenticating the BOSH connection manager (e.g=
. via TLS) imposes no<br>restriction whatsoever on who can be "called" via =
Jingle using that BOSH connection manger.&nbsp=3B&nbsp=3B <br>That restrict=
ion would only be implemented within the BOSH connection manager=2C *not* w=
ithin the <br>browser. <br></div> 		 	   		  </div></body>
</html>=

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From harald@alvestrand.no  Thu Aug  4 23:40:58 2011
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Subject: Re: [rtcweb] More on authorization and endpoint authentication
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On 08/04/11 18:46, Eric Rescorla wrote:
> At a higher level, I think we've been failing to draw the distinction
> between two cases which are socially (and arguably technically) very
> different:
>
> 1. The calling service is permitted to place a call but all the
>     information about who you are actually calling is accessible
>     only to the JS provided by the calling service.
>
> 2. The calling service is permitted to place a call but the
>     browser has independent information about who you are
>     actually talking to. [and could at least in principle render
>     it to the user.]
I do not like the thought of going down the case #2 road.

The reason is that in case 2, the concept of remote identity gets 
hardwired into the browser, and the API becomes useful only for 
applications that subscribe to the particular identity model that is 
hardwired into the browser.

In case #1, the API can be used with any (or no) concept of "remote 
identity", and the JS provider (who has a kind of "identity" by virtue 
of the Same Origin Policy definition, so it's already wired into the 
browser) gets to hold the responsibility.


From ekr@rtfm.com  Fri Aug  5 06:41:59 2011
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From: Eric Rescorla <ekr@rtfm.com>
Date: Fri, 5 Aug 2011 06:41:52 -0700
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On Thu, Aug 4, 2011 at 5:13 PM, Bernard Aboba <bernard_aboba@hotmail.com> w=
rote:
> Eric Rescorla said:
>
>> P2: Allow anyone to place a call provided that it goes to
>> *@ford.com.
>>
>> The browser evaluates this (in the simplest case) by connecting to
>> ford.com's SIP server and verifying that it is indeed ford.com
>
> These two statements are *not* equivalent.=A0 The first statement relates=
 to
> the To: field;
> the second relates to authentication of the identity of the server to whi=
ch
> the browser connects.
> These elements are orthogonal and should not be conflated.

Yes, you're right that I've elided some steps here.

In order for the browser to meaningful enforce a restriction on who you can
call, it needs to either (a) directly verify the endpoint, at least as
far as the
domain or (b) trust that the adjacency it is connected to will route messag=
es
correctly. So, what I had in mind here (again, in the simplest case) was
that the browser would connect to ford.com *and* use To: foo@ford.com,
thus having confidence that the message is routed correctly. Another altern=
ative
would be to connect to some trusted SIP proxy and use To: foo@ford.com.
Yet another alternative would of course be RFC 4474 Identity.

However, the point I was trying to make is that this is distinct from letti=
ng
the Web server that provided the page route the messages itself. You
might imagine not trusting the page but nevertheless be willing to allow
it to provide you with an address which you can independently verify
and derference.

Best,
-Ekr

From ted.ietf@gmail.com  Fri Aug  5 10:37:43 2011
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On Fri, Aug 5, 2011 at 6:41 AM, Eric Rescorla <ekr@rtfm.com> wrote:
> However, the point I was trying to make is that this is distinct from letting
> the Web server that provided the page route the messages itself. You
> might imagine not trusting the page but nevertheless be willing to allow
> it to provide you with an address which you can independently verify
> and derference.
>
> Best,
> -Ekr


Do you really mean "an address" which you can independently verify and
dereference, or an identity you can verify?

Let's take the poker site example.  The web site owner wants to embed
video streams in the poker application, so that the players are
reasonably assured that they are not playing a bot (or, at least, that
the player using a bot must be physically present during the game).
The poker players are likely put into groups based on arrival time,
maximum bid, etc.  in one mode; in another mode, a group of friends
may come and play together.  In the first mode, the players likely do
not want to share any identifying information with each other.  In the
second mode, the first player might invite the other players to a
room, by sharing a URI provided by the web server, or he might pass
their identities to the web server, so it can limit access to those
signing in with those identities.

In neither case, do we really want the web server to pass the video.
It serves as the initial signaling path for the messages among
players, but that's really it for the audio/video streaming part.

I've been assuming that the website would provide an identity in this
case, if one were needed (e.g. player5579@pokersite.example), but that
the primary reason for that identity would be outside the connectivity
model (so you know if you're playing the same player again, for
example, and can react if the picture looks different, or you don't
want to play with someone who is better than you).  Anyone in model
one (using a URI to join a specific game) would be assigned an
identity.  Anyone in model two (where the website locks down the game
to specific players) has to know what the player identities are to
invite.

In neither case does this seem to me to map well to SIP identity, and
I'm concerned that if we start with SIP identity as a mental model,
we're going to have softphones in browsers that can't do any of the
more web-based things that we actually want to enable.

regards,

Ted

From igor.faynberg@alcatel-lucent.com  Fri Aug  5 11:05:32 2011
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There are many interesting points here (as there are elsewhere on this 
thread).

Just to pick one, I wonder if it is an appropriate time to start zeroing 
in on acceptable identities as well as on the way of mapping them to a 
specific "person" (including bots).
There are two aspects to it. One is authentcation (i.e., a sort of a 
transitive closure reliance: if I know that A and B are different 
identities of P, and I have authenticated P based on A, I may , in 
certain circumstances, trust what comes from B), and the other is 
actually the choice of identifier.

Of course, I realize that this question sort of strays off the thread...

On the identifer choice, in direct response to Ted's last point, I would 
think that using SIP identities has much merit in several deployments. 
One is mobile telecom, where incidentally an end-point device is always 
sufficiently authenticated, with a  straight-forward extension to the 
person's authentication as the next step.

(By the way, when I read the quoted paragraph from Eric, I did assume 
that he meant "identity." )


Igor

On 8/5/2011 1:38 PM, Ted Hardie wrote:
> On Fri, Aug 5, 2011 at 6:41 AM, Eric Rescorla<ekr@rtfm.com>  wrote:
>> However, the point I was trying to make is that this is distinct from letting
>> the Web server that provided the page route the messages itself. You
>> might imagine not trusting the page but nevertheless be willing to allow
>> it to provide you with an address which you can independently verify
>> and derference.
>>
>> Best,
>> -Ekr
>
> Do you really mean "an address" which you can independently verify and
> dereference, or an identity you can verify?
>
> Let's take the poker site example.  The web site owner wants to embed
> video streams in the poker application, so that the players are
> reasonably assured that they are not playing a bot (or, at least, that
> the player using a bot must be physically present during the game).
> The poker players are likely put into groups based on arrival time,
> maximum bid, etc.  in one mode; in another mode, a group of friends
> may come and play together.  In the first mode, the players likely do
> not want to share any identifying information with each other.  In the
> second mode, the first player might invite the other players to a
> room, by sharing a URI provided by the web server, or he might pass
> their identities to the web server, so it can limit access to those
> signing in with those identities.
>
> In neither case, do we really want the web server to pass the video.
> It serves as the initial signaling path for the messages among
> players, but that's really it for the audio/video streaming part.
>
> I've been assuming that the website would provide an identity in this
> case, if one were needed (e.g. player5579@pokersite.example), but that
> the primary reason for that identity would be outside the connectivity
> model (so you know if you're playing the same player again, for
> example, and can react if the picture looks different, or you don't
> want to play with someone who is better than you).  Anyone in model
> one (using a URI to join a specific game) would be assigned an
> identity.  Anyone in model two (where the website locks down the game
> to specific players) has to know what the player identities are to
> invite.
>
> In neither case does this seem to me to map well to SIP identity, and
> I'm concerned that if we start with SIP identity as a mental model,
> we're going to have softphones in browsers that can't do any of the
> more web-based things that we actually want to enable.
>
> regards,
>
> Ted
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

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Date: Fri, 05 Aug 2011 11:33:30 -0700
From: Matthew Kaufman <matthew.kaufman@skype.net>
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Subject: [rtcweb] codec and connection negotiation
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I put this together to try to help myself, and perhaps others, 
understand the various ways in which codec and connection negotiation 
might be decomposed and then put together for RTCWEB applications. It is 
a bit of a stream of consciousness, but hopefully we can get some good 
discussion provoked.

A. How to encode codec capabilities or choices

A1 - Don't provide a canonical way of encoding codec settings. The 
choice of which codec and what parameters is encoded in an ad-hoc way 
that requires no standardization.

A2 - Encode codec settings using SDP. The choice of which codec and what 
parameters is encoded using the syntax and semantics of SDP and reuses 
the SDP standardization work.

A3 - Encode codec settings using a JSON encoding of SDP. The choice of 
which codec and what parameters is using the semantics of SDP but a JSON 
syntax. Reuses the SDP standardization plus new standards around how to 
encode SDP in JSON.

B. How to expose codec capabilities and controls

B1. Expose codec capabilities and settings via individual Javascript 
APIs. For example one might say "camera.encodeMode = "H.264"; 
camera.encodeWidth = 640;" or "if (camera.canEncode("H.264")) ..."

B2. Expose codec capabilities and settings via a combined Javascript API 
that concatenates all the capabilities and settings into a single 
object. Calling "camera.encodeCapabilities()" would return a large 
object that is a full enumeration of what it can do.

B3. Expose codec capabilities and settings via a combined Javascript API 
that produces and consumes SDP strings.

B4. Don't expose codec capbilities and settings via Javascript, rather 
handle this outside of Javascript.

C. How to agree on which codec and settings to use

C1. The server receives information about the capabilities, decides what 
settings should be used, and communicates them to the endpoints

C2. The endpoints agree using offer-answer, using the server as a 
communication channel

C3. The endpoints agree using offer-answer, directly between the two 
endpoints


(There's obviously a few additional nuances, like whether to use 
something like RFC 5939 SDP capability negotiation, that aren't 
well-captured above.)

Now, with the above, we find that some of the models that have been 
discussed are combinations of the above choices.

X1 - Use A2, B3, C2 - The endpoints generate SDP for offer-answer by 
firing an SDP event, having it sent via a server to the far end, where 
it is injected as SDP into the Javascript.

X2 - Use A1, B1, C1 - The endpoints run Javascript that inspects their 
capabilities, sends that to the server which decides what is best, and 
sends back information that the Javascript uses to set up the encoders.

X3 - Use A2, B4, C3 - The endpoints establish a communication channel 
using ICE and an out-of-band channel, they then use SDP inside of SIP to 
directly negotiate using offer-answer, the server has no information 
about what was chosen and does not participate.

We also find that almost all the models can map, though with varying 
degrees of pain, to others.

For instance:

Y1 - Use A2, B1, C2 - The endpoints run Javascript that inspects their 
capabilities and encodes it into an SDP string. This is sent via the 
server to the far end, where the SDP is parsed in Javascript and the 
individual APIs called to implement SDP offer-answer.

Y2 - Use A1, B3, C1 - The endpoints generate SDP for offer-answer by 
firing an SDP event. The endpoint then runs Javascript that extracts 
from the SDP all the individual parameters and re-encodes them using an 
ad-hoc scheme. The server determines what each end should do and sends 
ad-hoc messages to the endpoints which then generate the SDP answer that 
causes the desired outcome.

It should be noted that extracting a full set of capabilities when only 
offer-answer is available via the API is particularly painful, as it 
might be necessary to explore a very wide space of fake offers to get 
answers that clarify things like "can do G.722 audio but only if not 
doing H.264 video".

We might be able to learn from some previous experiences:

1. Web-based email. Web browsers run Javascript that uses ad-hoc 
signaling to/from the web site in order to send and receive email. The 
browser does not have SMTP, POP, or IMAP implementations, or even know 
how to parse RFC822 headers on its own. This argues for A1, B1 (or maybe 
B2), C1.

2. Web-based image display. Web browsers send an "Accept" header via 
HTTP indicating which MIME types they can handle. Knowing whether a 
browser can do JPEG is as simple as looking for "image/jpeg" in the 
header. This is probably at the wrong layer for sophisticated web 
applications (as this isn't about what comes back over HTTP, but what 
can be supported over a separate channel), but might be appropriate if 
we could agree on a small number of supported codecs and simply have the 
browser tell the server in a similar manner whether or not it can do RTCWEB.

3. SIP. SIP endpoints use SDP directly between the endpoints (though 
possibly with intermediaries that rarely change the SDP) to do 
offer-answer. This is most like A2, B3 (or B4), C3 (or C2).

4. MGCP. MGCP uses SDP for both capabilities and settings, but the 
server is in control. This is most like A2, B3, C1.

5. H.323 (H.245) uses a protocol other than SDP for both capabilities 
and settings, but the server is in control. This is like A3 (or some 
other encoding), B2, C1

As for connection negotiation, most of the separation previously also 
applies... again whether to put the ICE candidates into SDP or not, 
whether to have APIs that take blobs or individual calls, etc.

But the other issue that arises when we start talking about connection 
negotiation is that if you choose SDP offer-answer for codec 
negotiation, and SDP for ICE candidates, there is the temptation to 
combine the SDP for the two, thus conflating the process of establishing 
a logical channel over which various types of media might flow, and 
establishing just which media should be sent over that channel.

Matthew Kaufman


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Subject: Re: [rtcweb] codec and connection negotiation
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Now, to follow up on my previous message with a bit of opinion...

I think we should not use SDP as the API, and we should not have the 
browser implementing SDP offer-answer.

1. If SDP offer-answer is all that is available to the Javascript 
programmer (and server programmer), it becomes very difficult to know 
what the full capability set is without complex probing. This 
significantly complicates the building of future innovative applications 
on top of the browser as platform. Many of the current applications that 
show off browser capabilities do so by using capabilities that were 
already present for other reasons, and we can expect the same innovation 
to occur here, if we provide the tools.

2. Using SDP offer-answer has the advantage of reusing all the IETF 
standards work that occurs to define SDP when a new codec comes along. 
But it also has the *disadvantage* of having to wait for the IETF to 
produce these standards, which may be incomplete, or unable to control 
parameters which are necessary for web applications.

3. Anything that is missing in SDP (example: forcing the Opus codec to 
"music" mode for encoding) will still need to be exposed as a Javascript 
API on the encoder. Thus we end up with a mix of possibly-conflicting 
settings that are adjusted via the explicit API and the opaque SDP API.

4. Other work in browsers (like recording camera and microphone to disk) 
will require the ability to directly control the encoders. If these APIs 
exist then there will definitely be conflicting settings. What happens 
if you feed in an SDP answer that requires VP8 encoding and then set the 
encoder to H.264 mode via the Javascript?


Matthew Kaufman

From matthew.kaufman@skype.net  Fri Aug  5 18:11:41 2011
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While doing the research for the previous thread, I saw that MGCP 
actually has some things we might wish to borrow for RTCWEB. 
Specifically, some of the notification events... would sure be nice to 
get Javascript events when we saw "packet loss exceeded" or "RTP/RTCP 
timeout" for instance so that the web application could take an 
appropriate action.

Matthew Kaufman

From partr@cisco.com  Sat Aug  6 02:42:56 2011
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From: "Parthasarathi R (partr)" <partr@cisco.com>
To: "Matthew Kaufman" <matthew.kaufman@skype.net>, <rtcweb@ietf.org>
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Matthew,

I'm seeing two aspects here

1) API: Level of control has to be given to the Javascript programmer.
Here, the discussion has to focus on the abstraction by which
application shall be developed easily without much understanding the
protocol intricacies. Even API shall be provided to the extent that
username & destination host (xyz@abc.com) as a input shall trigger the
session creation with default codec.

2) Protocol: The protocol selection has to be based on better interop
with other browsers and other VoIP entity also. I'm not favoring for
creating different island of realtime protocol which has to interop with
"n" of gateways.

Please read inline for more comments.

Thanks
Partha
=20

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Matthew Kaufman
Sent: Saturday, August 06, 2011 12:24 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] codec and connection negotiation

Now, to follow up on my previous message with a bit of opinion...

I think we should not use SDP as the API,

<partha> It is fine to discuss which level of control has to be
provided. </partha>

 and we should not have the browser implementing SDP offer-answer.
<partha> I could not see the strong reason for not doing so. </partha>

1. If SDP offer-answer is all that is available to the Javascript
programmer (and server programmer), it becomes very difficult to know
what the full capability set is without complex probing. This
significantly complicates the building of future innovative applications
on top of the browser as platform. Many of the current applications that
show off browser capabilities do so by using capabilities that were
already present for other reasons, and we can expect the same innovation
to occur here, if we provide the tools.

2. Using SDP offer-answer has the advantage of reusing all the IETF
standards work that occurs to define SDP when a new codec comes along.=20
But it also has the *disadvantage* of having to wait for the IETF to
produce these standards, which may be incomplete, or unable to control
parameters which are necessary for web applications.
<partha> I prefer standard over proprietary mechanism because standard
may delay but proprietary is normal uncontrolled and impossible to
interop with other webservers </partha>

3. Anything that is missing in SDP (example: forcing the Opus codec to
"music" mode for encoding) will still need to be exposed as a Javascript
API on the encoder. Thus we end up with a mix of possibly-conflicting
settings that are adjusted via the explicit API and the opaque SDP API.
<partha> It is possible to integrate unknown SDP attribute & known SDP
attribute using the proper API design. It is the basic requirement in
any SDP stack API today. </partha>

4. Other work in browsers (like recording camera and microphone to disk)
will require the ability to directly control the encoders. If these APIs
exist then there will definitely be conflicting settings. What happens
if you feed in an SDP answer that requires VP8 encoding and then set the
encoder to H.264 mode via the Javascript?
<partha> I'm reading your query as how to priortize the value in case
extra environment change the value. It has to be worked out </partha>


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

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Hi all,
=20
browser application should have the mechanism by which it interacts with
voipserver directly instead of depend on the webserver. This usecase
provides the flexibility for the webdeveloper to focus on the
webdevelopment and use the existing voipservers for voip services by
just invoking the API.
=20
I'm not very sure whether this usecase is same as sec 4.3.1 as there is
no protocol architecture shown:
=20
         browser--------webservers---web services
         (Javascript)
             |
             ----------------voipservers-----VoIP entity (browser)
=20
This architecture facilites for webdeveloper to choose the different
vendor for webservers & voipservers. It is possible for webserver &
voipserver co-located but not mandatory. This architecture is slightly
different from draft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC
Trapezoid).  Please let me know your opinion on the same.
=20
Thanks
Partha
          =20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content=3D"text/html; charset=3Dus-ascii" =
http-equiv=3DContent-Type>
<META name=3DGENERATOR content=3D"MSHTML 8.00.6001.18928"></HEAD>
<BODY>
<DIV><FONT size=3D2 face=3DArial><SPAN class=3D645371910-06082011>Hi=20
all,</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN =
class=3D645371910-06082011>browser application=20
should have the mechanism by which it interacts with voipserver directly =
instead=20
of depend on the webserver. This usecase&nbsp;provides the flexibility =
for the=20
webdeveloper to focus on the webdevelopment and use the existing =
voipservers for=20
voip services&nbsp;by just invoking the API.</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN class=3D645371910-06082011>I'm =
not very sure=20
whether&nbsp;this usecase&nbsp;is same as&nbsp;sec 4.3.1 as there is no =
protocol=20
architecture shown:</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
browser--------webservers---web services</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;=20
(Javascript)</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
|</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;=20
----------------voipservers-----VoIP entity =
(browser)</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN class=3D645371910-06082011>This =
architecture=20
facilites for webdeveloper to choose the different vendor =
for&nbsp;webservers=20
&amp; voipservers. It is possible for webserver &amp; voipserver =
co-located but=20
not mandatory. This&nbsp;architecture is slightly different from=20
draft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC=20
Trapezoid).&nbsp;&nbsp;Please let me know your opinion on the=20
same.</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>Thanks</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>Partha</SPAN></FONT></DIV>
<DIV><FONT size=3D2 face=3DArial><SPAN=20
class=3D645371910-06082011>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;=20
</SPAN></FONT></DIV></BODY></HTML>

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This can be one of the architectures/use cases.=0ANote that it will be nice=
 to have different architectures/use cases to incubate innovation.=0A=A0=0A=
Regards,=0ASohel Khan=0A=0A=0A________________________________=0AFrom: Part=
hasarathi R (partr) <partr@cisco.com>=0ATo: rtcweb@ietf.org=0ASent: Saturda=
y, August 6, 2011 9:19 AM=0ASubject: [rtcweb] Usecase & architecture: Brows=
er application with separate webserver & voipserver=0A=0A=0AHi =0Aall,=0A=
=A0=0Abrowser application =0Ashould have the mechanism by which it interact=
s with voipserver directly instead =0Aof depend on the webserver. This usec=
ase=A0provides the flexibility for the =0Awebdeveloper to focus on the webd=
evelopment and use the existing voipservers for =0Avoip services=A0by just =
invoking the API.=0A=A0=0AI'm not very sure =0Awhether=A0this usecase=A0is =
same as=A0sec 4.3.1 as there is no protocol =0Aarchitecture shown:=0A=A0=0A=
=A0=A0=A0=A0=A0=A0=A0=A0 =0Abrowser--------webservers---web services=0A=A0=
=A0=A0=A0=A0=A0=A0=A0 =0A(Javascript)=0A=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=
=A0 =0A|=0A=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 =0A----------------voipserv=
ers-----VoIP entity (browser)=0A=A0=0AThis architecture =0Afacilites for we=
bdeveloper to choose the different vendor for=A0webservers =0A& voipservers=
. It is possible for webserver & voipserver co-located but =0Anot mandatory=
. This=A0architecture is slightly different from =0Adraft-rosenberg-rtcweb-=
framework-00 fig 2 (Browser RTC =0ATrapezoid).=A0=A0Please let me know your=
 opinion on the =0Asame.=0A=A0=0AThanks=0APartha=0A=A0=A0=A0=A0=A0=A0=A0=A0=
=A0=A0 =0A_______________________________________________=0Artcweb mailing =
list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/mailman/listinfo/rtcweb
--0-209000674-1312647300=:25384
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div><span>This can b=
e one of the architectures/use cases.</span></div><div><span>Note that it w=
ill be nice to have different architectures/use cases to incubate innovatio=
n.</span></div><div>&nbsp;</div><div>Regards,<br>Sohel Khan<br></div><div s=
tyle=3D"font-size: 12pt; font-family: 'times new roman', 'new york', times,=
 serif; "><div style=3D"font-size: 12pt; font-family: 'times new roman', 'n=
ew york', times, serif; "><font size=3D"2" face=3D"Arial"><hr size=3D"1"><b=
><span style=3D"font-weight:bold;">From:</span></b> Parthasarathi R (partr)=
 &lt;partr@cisco.com&gt;<br><b><span style=3D"font-weight: bold;">To:</span=
></b> rtcweb@ietf.org<br><b><span style=3D"font-weight: bold;">Sent:</span>=
</b> Saturday, August 6, 2011 9:19 AM<br><b><span style=3D"font-weight: bol=
d;">Subject:</span></b> [rtcweb] Usecase &amp; architecture: Browser applic=
ation with
 separate webserver &amp; voipserver<br></font><br><div id=3D"yiv698706282"=
>=0A=0A =0A =0A =0A<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv6=
98706282645371910-06082011">Hi =0Aall,</span></font></div>=0A<div><font siz=
e=3D"2" face=3D"Arial"><span class=3D"yiv698706282645371910-06082011"></spa=
n></font>&nbsp;</div>=0A<div><font size=3D"2" face=3D"Arial"><span class=3D=
"yiv698706282645371910-06082011">browser application =0Ashould have the mec=
hanism by which it interacts with voipserver directly instead =0Aof depend =
on the webserver. This usecase&nbsp;provides the flexibility for the =0Aweb=
developer to focus on the webdevelopment and use the existing voipservers f=
or =0Avoip services&nbsp;by just invoking the API.</span></font></div>=0A<d=
iv><font size=3D"2" face=3D"Arial"><span class=3D"yiv698706282645371910-060=
82011"></span></font>&nbsp;</div>=0A<div><font size=3D"2" face=3D"Arial"><s=
pan class=3D"yiv698706282645371910-06082011">I'm not very sure =0Awhether&n=
bsp;this usecase&nbsp;is same as&nbsp;sec 4.3.1 as there is no protocol =0A=
architecture shown:</span></font></div>=0A<div><font size=3D"2" face=3D"Ari=
al"><span class=3D"yiv698706282645371910-06082011"></span></font>&nbsp;</di=
v>=0A<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv698706282645371=
910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =0Abrowser--=
------webservers---web services</span></font></div>=0A<div><font size=3D"2"=
 face=3D"Arial"><span class=3D"yiv698706282645371910-06082011">&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =0A(Javascript)</span></font></div>=0A=
<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv698706282645371910-0=
6082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; =0A|</span></font></div>=0A<div><font size=3D"2" face=3D"Arial"><spa=
n class=3D"yiv698706282645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =0A----------------voipservers----=
-VoIP entity (browser)</span></font></div>=0A<div><font size=3D"2" face=3D"=
Arial"><span class=3D"yiv698706282645371910-06082011"></span></font>&nbsp;<=
/div>=0A<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv698706282645=
371910-06082011">This architecture =0Afacilites for webdeveloper to choose =
the different vendor for&nbsp;webservers =0A&amp; voipservers. It is possib=
le for webserver &amp; voipserver co-located but =0Anot mandatory. This&nbs=
p;architecture is slightly different from =0Adraft-rosenberg-rtcweb-framewo=
rk-00 fig 2 (Browser RTC =0ATrapezoid).&nbsp;&nbsp;Please let me know your =
opinion on the =0Asame.</span></font></div>=0A<div><font size=3D"2" face=3D=
"Arial"><span class=3D"yiv698706282645371910-06082011"></span></font>&nbsp;=
</div>=0A<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv69870628264=
5371910-06082011">Thanks</span></font></div>=0A<div><font size=3D"2" face=
=3D"Arial"><span class=3D"yiv698706282645371910-06082011">Partha</span></fo=
nt></div>=0A<div><font size=3D"2" face=3D"Arial"><span class=3D"yiv69870628=
2645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; =0A</span></font></div> =0A</div><br>_______________________________=
________________<br>rtcweb mailing list<br><a ymailto=3D"mailto:rtcweb@ietf=
.org" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><a href=3D"htt=
ps://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">https://www.ie=
tf.org/mailman/listinfo/rtcweb</a><br><br><br></div></div></div></body></ht=
ml>
--0-209000674-1312647300=:25384--

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Date: Sat, 06 Aug 2011 09:36:28 -0700
From: Matthew Kaufman <matthew.kaufman@skype.net>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Usecase & architecture: Browser application with separate webserver & voipserver
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On 8/6/2011 6:19 AM, Parthasarathi R (partr) wrote:
> Hi all,
> browser application should have the mechanism by which it interacts 
> with voipserver directly instead of depend on the webserver. This 
> usecase provides the flexibility for the webdeveloper to focus on the 
> webdevelopment and use the existing voipservers for voip services by 
> just invoking the API.
> I'm not very sure whether this usecase is same as sec 4.3.1 as there 
> is no protocol architecture shown:
>          browser--------webservers---web services
>          (Javascript)
>              |
>              ----------------voipservers-----VoIP entity (browser)
> This architecture facilites for webdeveloper to choose the different 
> vendor for webservers & voipservers. It is possible for webserver & 
> voipserver co-located but not mandatory. This architecture is slightly 
> different from draft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC 
> Trapezoid).  Please let me know your opinion on the same.
>

My opinion is that as long as the "voipservers" use HTTP-based signaling 
and RTCWEB-compatible VoIP, then that's fine, and indistinguishable from 
other use cases. But if the "voipservers" are existing SIP servers then 
no, I don't want additional inflexible code in the browser that turns it 
into a SIP phone.

Matthew Kaufman

--------------040306040707010506030808
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    On 8/6/2011 6:19 AM, Parthasarathi R (partr) wrote:
    <blockquote
cite="mid:A11921905DA1564D9BCF64A6430A62390608EC13@XMB-BGL-411.cisco.com"
      type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <meta name="GENERATOR" content="MSHTML 8.00.6001.18928">
      <div><font face="Arial" size="2"><span class="645371910-06082011">Hi
            all,</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011"></span></font>&nbsp;</div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">browser
            application should have the mechanism by which it interacts
            with voipserver directly instead of depend on the webserver.
            This usecase&nbsp;provides the flexibility for the webdeveloper
            to focus on the webdevelopment and use the existing
            voipservers for voip services&nbsp;by just invoking the API.</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011"></span></font>&nbsp;</div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">I'm
            not very sure whether&nbsp;this usecase&nbsp;is same as&nbsp;sec 4.3.1 as
            there is no protocol architecture shown:</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011"></span></font>&nbsp;</div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            browser--------webservers---web services</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            (Javascript)</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            |</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            ----------------voipservers-----VoIP entity (browser)</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011"></span></font>&nbsp;</div>
      <div><font face="Arial" size="2"><span class="645371910-06082011">This
            architecture facilites for webdeveloper to choose the
            different vendor for&nbsp;webservers &amp; voipservers. It is
            possible for webserver &amp; voipserver co-located but not
            mandatory. This&nbsp;architecture is slightly different from
            draft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC
            Trapezoid).&nbsp;&nbsp;Please let me know your opinion on the same.</span></font></div>
      <div><font face="Arial" size="2"><span class="645371910-06082011"></span></font><br>
      </div>
    </blockquote>
    <br>
    My opinion is that as long as the "voipservers" use HTTP-based
    signaling and RTCWEB-compatible VoIP, then that's fine, and
    indistinguishable from other use cases. But if the "voipservers" are
    existing SIP servers then no, I don't want additional inflexible
    code in the browser that turns it into a SIP phone.<br>
    <br>
    Matthew Kaufman<br>
  </body>
</html>

--------------040306040707010506030808--

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From: Sohel Khan <sohel_khan777@yahoo.com>
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Thanks Matt and Partha for valuable comments and proposals.=0AThe main poin=
t here is to create standards specifying methods to exchange desired sessio=
n capabilities. From various discussions, it appears to me that there are t=
wo key options:=0A1. No SDP in browser. Object oriented and popular script =
language (e.g., JavaScript) implementation transfers desired session capabi=
lities in between browser and server. The server performs SDP offer/answer =
in the VoIP core network.=0A2. SDP in browser. Use IETF SDP offer/answer st=
andards to transfer desired session capabilities from browser to browser wi=
th/without core VoIP network.=0A=0AIn my opinion, we will need to create RT=
CWeb standards for both the options. We can either create one document cont=
aining both standards or two separate documents each containing an option. =
The architecture and use case documents should include both options.=0A=0A=
=A0=0ARegards,=0ASohel Khan=0A=0A=0A________________________________=0AFrom=
: Parthasarathi R (partr) <partr@cisco.com>=0ATo: Matthew Kaufman <matthew.=
kaufman@skype.net>; rtcweb@ietf.org=0ASent: Saturday, August 6, 2011 5:43 A=
M=0ASubject: Re: [rtcweb] codec and connection negotiation=0A=0AMatthew,=0A=
=0AI'm seeing two aspects here=0A=0A1) API: Level of control has to be give=
n to the Javascript programmer.=0AHere, the discussion has to focus on the =
abstraction by which=0Aapplication shall be developed easily without much u=
nderstanding the=0Aprotocol intricacies. Even API shall be provided to the =
extent that=0Ausername & destination host (xyz@abc.com) as a input shall tr=
igger the=0Asession creation with default codec.=0A=0A2) Protocol: The prot=
ocol selection has to be based on better interop=0Awith other browsers and =
other VoIP entity also. I'm not favoring for=0Acreating different island of=
 realtime protocol which has to interop with=0A"n" of gateways.=0A=0APlease=
 read inline for more comments.=0A=0AThanks=0APartha=0A=0A=0A-----Original =
Message-----=0AFrom: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.or=
g] On Behalf=0AOf Matthew Kaufman=0ASent: Saturday, August 06, 2011 12:24 A=
M=0ATo: rtcweb@ietf.org=0ASubject: Re: [rtcweb] codec and connection negoti=
ation=0A=0ANow, to follow up on my previous message with a bit of opinion..=
.=0A=0AI think we should not use SDP as the API,=0A=0A<partha> It is fine t=
o discuss which level of control has to be=0Aprovided. </partha>=0A=0Aand w=
e should not have the browser implementing SDP offer-answer.=0A<partha> I c=
ould not see the strong reason for not doing so. </partha>=0A=0A1. If SDP o=
ffer-answer is all that is available to the Javascript=0Aprogrammer (and se=
rver programmer), it becomes very difficult to know=0Awhat the full capabil=
ity set is without complex probing. This=0Asignificantly complicates the bu=
ilding of future innovative applications=0Aon top of the browser as platfor=
m. Many of the current applications that=0Ashow off browser capabilities do=
 so by using capabilities that were=0Aalready present for other reasons, an=
d we can expect the same innovation=0Ato occur here, if we provide the tool=
s.=0A=0A2. Using SDP offer-answer has the advantage of reusing all the IETF=
=0Astandards work that occurs to define SDP when a new codec comes along. =
=0ABut it also has the *disadvantage* of having to wait for the IETF to=0Ap=
roduce these standards, which may be incomplete, or unable to control=0Apar=
ameters which are necessary for web applications.=0A<partha> I prefer stand=
ard over proprietary mechanism because standard=0Amay delay but proprietary=
 is normal uncontrolled and impossible to=0Ainterop with other webservers <=
/partha>=0A=0A3. Anything that is missing in SDP (example: forcing the Opus=
 codec to=0A"music" mode for encoding) will still need to be exposed as a J=
avascript=0AAPI on the encoder. Thus we end up with a mix of possibly-confl=
icting=0Asettings that are adjusted via the explicit API and the opaque SDP=
 API.=0A<partha> It is possible to integrate unknown SDP attribute & known =
SDP=0Aattribute using the proper API design. It is the basic requirement in=
=0Aany SDP stack API today. </partha>=0A=0A4. Other work in browsers (like =
recording camera and microphone to disk)=0Awill require the ability to dire=
ctly control the encoders. If these APIs=0Aexist then there will definitely=
 be conflicting settings. What happens=0Aif you feed in an SDP answer that =
requires VP8 encoding and then set the=0Aencoder to H.264 mode via the Java=
script?=0A<partha> I'm reading your query as how to priortize the value in =
case=0Aextra environment change the value. It has to be worked out </partha=
>=0A=0A=0AMatthew Kaufman=0A_______________________________________________=
=0Artcweb mailing list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/mailman/lis=
tinfo/rtcweb=0A_______________________________________________=0Artcweb mai=
ling list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/mailman/listinfo/rtcweb
--0-370549104-1312649902=:16121
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div><span>Thanks Mat=
t and Partha for valuable comments and proposals.</span></div><div><span>Th=
e main point here is to create standards specifying methods to exchange des=
ired session capabilities. From various discussions, it appears to me that =
there are two key options:</span></div><div><span>1. No SDP in browser. Obj=
ect oriented and popular script language (e.g., JavaScript) implementation =
transfers desired session capabilities in between browser and server. The s=
erver performs SDP offer/answer in the VoIP core network.</span></div><div>=
<span>2. SDP in browser. Use IETF SDP offer/answer standards to transfer de=
sired session capabilities from browser to browser with/without core VoIP n=
etwork.</span></div><div><span><br></span></div><div><span>In my opinion, w=
e will need to create RTCWeb standards for both the options. We can
 either create one document containing both standards or two separate docum=
ents each containing an option. The architecture and use case documents sho=
uld include both options.</span></div><div><span><br></span></div><div>&nbs=
p;</div><div>Regards,<br>Sohel Khan<br></div><div style=3D"font-size: 12pt;=
 font-family: 'times new roman', 'new york', times, serif; "><div style=3D"=
font-size: 12pt; font-family: 'times new roman', 'new york', times, serif; =
"><font size=3D"2" face=3D"Arial"><hr size=3D"1"><b><span style=3D"font-wei=
ght:bold;">From:</span></b> Parthasarathi R (partr) &lt;partr@cisco.com&gt;=
<br><b><span style=3D"font-weight: bold;">To:</span></b> Matthew Kaufman &l=
t;matthew.kaufman@skype.net&gt;; rtcweb@ietf.org<br><b><span style=3D"font-=
weight: bold;">Sent:</span></b> Saturday, August 6, 2011 5:43 AM<br><b><spa=
n style=3D"font-weight: bold;">Subject:</span></b> Re: [rtcweb] codec and c=
onnection negotiation<br></font><br>Matthew,<br><br>I'm seeing two aspects
 here<br><br>1) API: Level of control has to be given to the Javascript pro=
grammer.<br>Here, the discussion has to focus on the abstraction by which<b=
r>application shall be developed easily without much understanding the<br>p=
rotocol intricacies. Even API shall be provided to the extent that<br>usern=
ame &amp; destination host (<a ymailto=3D"mailto:xyz@abc.com" href=3D"mailt=
o:xyz@abc.com">xyz@abc.com</a>) as a input shall trigger the<br>session cre=
ation with default codec.<br><br>2) Protocol: The protocol selection has to=
 be based on better interop<br>with other browsers and other VoIP entity al=
so. I'm not favoring for<br>creating different island of realtime protocol =
which has to interop with<br>"n" of gateways.<br><br>Please read inline for=
 more comments.<br><br>Thanks<br>Partha<br> <br><br>-----Original Message--=
---<br>From: <a ymailto=3D"mailto:rtcweb-bounces@ietf.org" href=3D"mailto:r=
tcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [mailto:<a
 ymailto=3D"mailto:rtcweb-bounces@ietf.org" href=3D"mailto:rtcweb-bounces@i=
etf.org">rtcweb-bounces@ietf.org</a>] On Behalf<br>Of Matthew Kaufman<br>Se=
nt: Saturday, August 06, 2011 12:24 AM<br>To: <a ymailto=3D"mailto:rtcweb@i=
etf.org" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>Subject: Re=
: [rtcweb] codec and connection negotiation<br><br>Now, to follow up on my =
previous message with a bit of opinion...<br><br>I think we should not use =
SDP as the API,<br><br>&lt;partha&gt; It is fine to discuss which level of =
control has to be<br>provided. &lt;/partha&gt;<br><br> and we should not ha=
ve the browser implementing SDP offer-answer.<br>&lt;partha&gt; I could not=
 see the strong reason for not doing so. &lt;/partha&gt;<br><br>1. If SDP o=
ffer-answer is all that is available to the Javascript<br>programmer (and s=
erver programmer), it becomes very difficult to know<br>what the full capab=
ility set is without complex probing. This<br>significantly complicates the
 building of future innovative applications<br>on top of the browser as pla=
tform. Many of the current applications that<br>show off browser capabiliti=
es do so by using capabilities that were<br>already present for other reaso=
ns, and we can expect the same innovation<br>to occur here, if we provide t=
he tools.<br><br>2. Using SDP offer-answer has the advantage of reusing all=
 the IETF<br>standards work that occurs to define SDP when a new codec come=
s along. <br>But it also has the *disadvantage* of having to wait for the I=
ETF to<br>produce these standards, which may be incomplete, or unable to co=
ntrol<br>parameters which are necessary for web applications.<br>&lt;partha=
&gt; I prefer standard over proprietary mechanism because standard<br>may d=
elay but proprietary is normal uncontrolled and impossible to<br>interop wi=
th other webservers &lt;/partha&gt;<br><br>3. Anything that is missing in S=
DP (example: forcing the Opus codec to<br>"music" mode for encoding)
 will still need to be exposed as a Javascript<br>API on the encoder. Thus =
we end up with a mix of possibly-conflicting<br>settings that are adjusted =
via the explicit API and the opaque SDP API.<br>&lt;partha&gt; It is possib=
le to integrate unknown SDP attribute &amp; known SDP<br>attribute using th=
e proper API design. It is the basic requirement in<br>any SDP stack API to=
day. &lt;/partha&gt;<br><br>4. Other work in browsers (like recording camer=
a and microphone to disk)<br>will require the ability to directly control t=
he encoders. If these APIs<br>exist then there will definitely be conflicti=
ng settings. What happens<br>if you feed in an SDP answer that requires VP8=
 encoding and then set the<br>encoder to H.264 mode via the Javascript?<br>=
&lt;partha&gt; I'm reading your query as how to priortize the value in case=
<br>extra environment change the value. It has to be worked out &lt;/partha=
&gt;<br><br><br>Matthew
 Kaufman<br>_______________________________________________<br>rtcweb maili=
ng list<br><a ymailto=3D"mailto:rtcweb@ietf.org" href=3D"mailto:rtcweb@ietf=
.org">rtcweb@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listin=
fo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</=
a><br>_______________________________________________<br>rtcweb mailing lis=
t<br><a ymailto=3D"mailto:rtcweb@ietf.org" href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a><br><a href=3D"https://www.ietf.org/mailman/listinfo/rtc=
web" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>=
<br><br></div></div></div></body></html>
--0-370549104-1312649902=:16121--

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Date: Sat, 06 Aug 2011 14:46:56 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: Sohel Khan <sohel_khan777@yahoo.com>, "Parthasarathi R (partr)" <partr@cisco.com>, Matthew Kaufman <matthew.kaufman@skype.net>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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+1

Thanks, Henry


On 8/6/11 11:58 AM, "Sohel Khan" <sohel_khan777@yahoo.com> wrote:

> Thanks Matt and Partha for valuable comments and proposals.
> The main point here is to create standards specifying methods to exchange
> desired session capabilities. From various discussions, it appears to me that
> there are two key options:
> 1. No SDP in browser. Object oriented and popular script language (e.g.,
> JavaScript) implementation transfers desired session capabilities in between
> browser and server. The server performs SDP offer/answer in the VoIP core
> network.
> 2. SDP in browser. Use IETF SDP offer/answer standards to transfer desired
> session capabilities from browser to browser with/without core VoIP network.
> 
> In my opinion, we will need to create RTCWeb standards for both the options.
> We can either create one document containing both standards or two separate
> documents each containing an option. The architecture and use case documents
> should include both options.
> 
>  
> Regards,
> Sohel Khan
> 
> From: Parthasarathi R (partr) <partr@cisco.com>
> To: Matthew Kaufman <matthew.kaufman@skype.net>; rtcweb@ietf.org
> Sent: Saturday, August 6, 2011 5:43 AM
> Subject: Re: [rtcweb] codec and connection negotiation
> 
> Matthew,
> 
> I'm seeing two aspects here
> 
> 1) API: Level of control has to be given to the Javascript programmer.
> Here, the discussion has to focus on the abstraction by which
> application shall be developed easily without much understanding the
> protocol intricacies. Even API shall be provided to the extent that
> username & destination host (xyz@abc.com) as a input shall trigger the
> session creation with default codec.
> 
> 2) Protocol: The protocol selection has to be based on better interop
> with other browsers and other VoIP entity also. I'm not favoring for
> creating different island of realtime protocol which has to interop with
> "n" of gateways.
> 
> Please read inline for more comments.
> 
> Thanks
> Partha
>  
> 
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Matthew Kaufman
> Sent: Saturday, August 06, 2011 12:24 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] codec and connection negotiation
> 
> Now, to follow up on my previous message with a bit of opinion...
> 
> I think we should not use SDP as the API,
> 
> <partha> It is fine to discuss which level of control has to be
> provided. </partha>
> 
>  and we should not have the browser implementing SDP offer-answer.
> <partha> I could not see the strong reason for not doing so. </partha>
> 
> 1. If SDP offer-answer is all that is available to the Javascript
> programmer (and server programmer), it becomes very difficult to know
> what the full capability set is without complex probing. This
> significantly complicates the building of future innovative applications
> on top of the browser as platform. Many of the current applications that
> show off browser capabilities do so by using capabilities that were
> already present for other reasons, and we can expect the same innovation
> to occur here, if we provide the tools.
> 
> 2. Using SDP offer-answer has the advantage of reusing all the IETF
> standards work that occurs to define SDP when a new codec comes along.
> But it also has the *disadvantage* of having to wait for the IETF to
> produce these standards, which may be incomplete, or unable to control
> parameters which are necessary for web applications.
> <partha> I prefer standard over proprietary mechanism because standard
> may delay but proprietary is normal uncontrolled and impossible to
> interop with other webservers </partha>
> 
> 3. Anything that is missing in SDP (example: forcing the Opus codec to
> "music" mode for encoding) will still need to be exposed as a Javascript
> API on the encoder. Thus we end up with a mix of possibly-conflicting
> settings that are adjusted via the explicit API and the opaque SDP API.
> <partha> It is possible to integrate unknown SDP attribute & known SDP
> attribute using the proper API design. It is the basic requirement in
> any SDP stack API today. </partha>
> 
> 4. Other work in browsers (like recording camera and microphone to disk)
> will require the ability to directly control the encoders. If these APIs
> exist then there will definitely be conflicting settings. What happens
> if you feed in an SDP answer that requires VP8 encoding and then set the
> encoder to H.264 mode via the Javascript?
> <partha> I'm reading your query as how to priortize the value in case
> extra environment change the value. It has to be worked out </partha>
> 
> 
> Matthew Kaufman
> _______________________________________________
> 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


--B_3395486819_3439882
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	charset="US-ASCII"
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<HTML>
<HEAD>
<TITLE>Re: [rtcweb] codec and connection negotiation</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>+1<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 8/6/11 11:58 AM, &quot;Sohel Khan&quot; &lt;<a href=3D"sohel_khan777@yahoo=
.com">sohel_khan777@yahoo.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE=3D"2"><FONT FACE=3D"Times New Roman"><SPAN=
 STYLE=3D'font-size:12pt'>Thanks Matt and Partha for valuable comments and pro=
posals.<BR>
The main point here is to create standards specifying methods to exchange d=
esired session capabilities. From various discussions, it appears to me that=
 there are two key options:<BR>
1. No SDP in browser. Object oriented and popular script language (e.g., Ja=
vaScript) implementation transfers desired session capabilities in between b=
rowser and server. The server performs SDP offer/answer in the VoIP core net=
work.<BR>
2. SDP in browser. Use IETF SDP offer/answer standards to transfer desired =
session capabilities from browser to browser with/without core VoIP network.=
<BR>
<BR>
In my opinion, we will need to create RTCWeb standards for both the options=
. We can either create one document containing both standards or two separat=
e documents each containing an option. The architecture and use case documen=
ts should include both options.<BR>
<BR>
&nbsp;<BR>
Regards,<BR>
Sohel Khan<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Arial"><SPAN STYLE=3D'font-size:13pt'><HR AL=
IGN=3DCENTER SIZE=3D"1" WIDTH=3D"100%"><B>From:</B></SPAN><FONT SIZE=3D"2"><SPAN STY=
LE=3D'font-size:12pt'> Parthasarathi R (partr) &lt;<a href=3D"partr@cisco.com">p=
artr@cisco.com</a>&gt;<BR>
<B>To:</B> Matthew Kaufman &lt;<a href=3D"matthew.kaufman@skype.net">matthew.=
kaufman@skype.net</a>&gt;; <a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<B>Sent:</B> Saturday, August 6, 2011 5:43 AM<BR>
<B>Subject:</B> Re: [rtcweb] codec and connection negotiation<BR>
</SPAN></FONT></FONT><FONT SIZE=3D"2"><SPAN STYLE=3D'font-size:12pt'><FONT FACE=
=3D"Times New Roman"><BR>
Matthew,<BR>
<BR>
I'm seeing two aspects here<BR>
<BR>
1) API: Level of control has to be given to the Javascript programmer.<BR>
Here, the discussion has to focus on the abstraction by which<BR>
application shall be developed easily without much understanding the<BR>
protocol intricacies. Even API shall be provided to the extent that<BR>
username &amp; destination host (<a href=3D"xyz@abc.com">xyz@abc.com</a>) as =
a input shall trigger the<BR>
session creation with default codec.<BR>
<BR>
2) Protocol: The protocol selection has to be based on better interop<BR>
with other browsers and other VoIP entity also. I'm not favoring for<BR>
creating different island of realtime protocol which has to interop with<BR=
>
&quot;n&quot; of gateways.<BR>
<BR>
Please read inline for more comments.<BR>
<BR>
Thanks<BR>
Partha<BR>
&nbsp;<BR>
<BR>
-----Original Message-----<BR>
From: <a href=3D"rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a hre=
f=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] On Be=
half<BR>
Of Matthew Kaufman<BR>
Sent: Saturday, August 06, 2011 12:24 AM<BR>
To: <a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
Subject: Re: [rtcweb] codec and connection negotiation<BR>
<BR>
Now, to follow up on my previous message with a bit of opinion...<BR>
<BR>
I think we should not use SDP as the API,<BR>
<BR>
&lt;partha&gt; It is fine to discuss which level of control has to be<BR>
provided. &lt;/partha&gt;<BR>
<BR>
&nbsp;and we should not have the browser implementing SDP offer-answer.<BR>
&lt;partha&gt; I could not see the strong reason for not doing so. &lt;/par=
tha&gt;<BR>
<BR>
1. If SDP offer-answer is all that is available to the Javascript<BR>
programmer (and server programmer), it becomes very difficult to know<BR>
what the full capability set is without complex probing. This<BR>
significantly complicates the building of future innovative applications<BR=
>
on top of the browser as platform. Many of the current applications that<BR=
>
show off browser capabilities do so by using capabilities that were<BR>
already present for other reasons, and we can expect the same innovation<BR=
>
to occur here, if we provide the tools.<BR>
<BR>
2. Using SDP offer-answer has the advantage of reusing all the IETF<BR>
standards work that occurs to define SDP when a new codec comes along. <BR>
But it also has the *disadvantage* of having to wait for the IETF to<BR>
produce these standards, which may be incomplete, or unable to control<BR>
parameters which are necessary for web applications.<BR>
&lt;partha&gt; I prefer standard over proprietary mechanism because standar=
d<BR>
may delay but proprietary is normal uncontrolled and impossible to<BR>
interop with other webservers &lt;/partha&gt;<BR>
<BR>
3. Anything that is missing in SDP (example: forcing the Opus codec to<BR>
&quot;music&quot; mode for encoding) will still need to be exposed as a Jav=
ascript<BR>
API on the encoder. Thus we end up with a mix of possibly-conflicting<BR>
settings that are adjusted via the explicit API and the opaque SDP API.<BR>
&lt;partha&gt; It is possible to integrate unknown SDP attribute &amp; know=
n SDP<BR>
attribute using the proper API design. It is the basic requirement in<BR>
any SDP stack API today. &lt;/partha&gt;<BR>
<BR>
4. Other work in browsers (like recording camera and microphone to disk)<BR=
>
will require the ability to directly control the encoders. If these APIs<BR=
>
exist then there will definitely be conflicting settings. What happens<BR>
if you feed in an SDP answer that requires VP8 encoding and then set the<BR=
>
encoder to H.264 mode via the Javascript?<BR>
&lt;partha&gt; I'm reading your query as how to priortize the value in case=
<BR>
extra environment change the value. It has to be worked out &lt;/partha&gt;=
<BR>
<BR>
<BR>
Matthew Kaufman<BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
<BR>
<BR>
</FONT></SPAN></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"95%"></SPAN></FONT><SPAN STYLE=3D'font-size:=
13pt'><FONT FACE=3D"Consolas, Courier New, Courier">__________________________=
_____________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
</FONT></SPAN></BLOCKQUOTE>
</BODY>
</HTML>


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Matt, Thank you for nicely presenting various sets of session capability tr=
ansfer methods.=0AI like X1 and X2, and also Y1 and Y2. =A0Although I do no=
t have any opinion on X3, it is okay to add this use case in the document. =
Diverse methods create innovation opportunities although create interoperab=
ility challenges.=A0=0A=0AAlthough in a utopian network, it is tempting to =
consider that no-middle box (e.g., network transcoder) should exist, unfort=
unately, not all end system support the same sets of codec in the today's n=
etwork. Therefore, a network transcoding system may be needed; thus, the ne=
ed of servers in the network.=0A=0A=0AApart from above method selections, i=
n details level, we will need some important parameters. I am writing here =
although this thread may not be the right forum.=0AOne important point is t=
o ensure we have a field or parameter, or arrangement order to depict media=
 capability preference. For example, in SDP, the first codec in the list is=
 preferred. This is helpful for both network (e.g., transcoding mediation) =
and end systems, in which those support multiple codecs but would like to s=
elect the default codec. In addition, we have to consider a parameter for "=
force: no <parameter>" to explicitly mention unwanted media capability. For=
 example, an end user or network may like to preclude a "bad-codec". In thi=
s case, the end system or network should be able to negotiate "force: no ba=
d-codec". =A0=0A=0AIn DTMF codec negotiation, we will need to ensure that a=
 parameter clearly indicates which DTMF transfer method is supported. Curre=
ntly, RFC 2833 =A0telephone-event expresses intention of out-of-band RTP DT=
MF digit transfer. In my opinion, we do not have a tool to deny SIP INFO-ba=
sed DTMF digit transfer. =A0Some transcoder in the network or end user code=
c may prefer not to use INFO-based DTMF digit transfer. In this case, a par=
ameter such as "force: no info-dtmf-event" will be helpful.=0A=0ARegards,=
=0ASohel Khan=0A=0A=0A________________________________=0AFrom: Matthew Kauf=
man <matthew.kaufman@skype.net>=0ATo: "rtcweb@ietf.org" <rtcweb@ietf.org>=
=0ASent: Friday, August 5, 2011 2:33 PM=0ASubject: [rtcweb] codec and conne=
ction negotiation=0A=0AI put this together to try to help myself, and perha=
ps others, understand the various ways in which codec and connection negoti=
ation might be decomposed and then put together for RTCWEB applications. It=
 is a bit of a stream of consciousness, but hopefully we can get some good =
discussion provoked.=0A=0AA. How to encode codec capabilities or choices=0A=
=0AA1 - Don't provide a canonical way of encoding codec settings. The choic=
e of which codec and what parameters is encoded in an ad-hoc way that requi=
res no standardization.=0A=0AA2 - Encode codec settings using SDP. The choi=
ce of which codec and what parameters is encoded using the syntax and seman=
tics of SDP and reuses the SDP standardization work.=0A=0AA3 - Encode codec=
 settings using a JSON encoding of SDP. The choice of which codec and what =
parameters is using the semantics of SDP but a JSON syntax. Reuses the SDP =
standardization plus new standards around how to encode SDP in JSON.=0A=0AB=
. How to expose codec capabilities and controls=0A=0AB1. Expose codec capab=
ilities and settings via individual Javascript APIs. For example one might =
say "camera.encodeMode =3D "H.264"; camera.encodeWidth =3D 640;" or "if (ca=
mera.canEncode("H.264")) ..."=0A=0AB2. Expose codec capabilities and settin=
gs via a combined Javascript API that concatenates all the capabilities and=
 settings into a single object. Calling "camera.encodeCapabilities()" would=
 return a large object that is a full enumeration of what it can do.=0A=0AB=
3. Expose codec capabilities and settings via a combined Javascript API tha=
t produces and consumes SDP strings.=0A=0AB4. Don't expose codec capbilitie=
s and settings via Javascript, rather handle this outside of Javascript.=0A=
=0AC. How to agree on which codec and settings to use=0A=0AC1. The server r=
eceives information about the capabilities, decides what settings should be=
 used, and communicates them to the endpoints=0A=0AC2. The endpoints agree =
using offer-answer, using the server as a communication channel=0A=0AC3. Th=
e endpoints agree using offer-answer, directly between the two endpoints=0A=
=0A=0A(There's obviously a few additional nuances, like whether to use some=
thing like RFC 5939 SDP capability negotiation, that aren't well-captured a=
bove.)=0A=0ANow, with the above, we find that some of the models that have =
been discussed are combinations of the above choices.=0A=0AX1 - Use A2, B3,=
 C2 - The endpoints generate SDP for offer-answer by firing an SDP event, h=
aving it sent via a server to the far end, where it is injected as SDP into=
 the Javascript.=0A=0AX2 - Use A1, B1, C1 - The endpoints run Javascript th=
at inspects their capabilities, sends that to the server which decides what=
 is best, and sends back information that the Javascript uses to set up the=
 encoders.=0A=0AX3 - Use A2, B4, C3 - The endpoints establish a communicati=
on channel using ICE and an out-of-band channel, they then use SDP inside o=
f SIP to directly negotiate using offer-answer, the server has no informati=
on about what was chosen and does not participate.=0A=0AWe also find that a=
lmost all the models can map, though with varying degrees of pain, to other=
s.=0A=0AFor instance:=0A=0AY1 - Use A2, B1, C2 - The endpoints run Javascri=
pt that inspects their capabilities and encodes it into an SDP string. This=
 is sent via the server to the far end, where the SDP is parsed in Javascri=
pt and the individual APIs called to implement SDP offer-answer.=0A=0AY2 - =
Use A1, B3, C1 - The endpoints generate SDP for offer-answer by firing an S=
DP event. The endpoint then runs Javascript that extracts from the SDP all =
the individual parameters and re-encodes them using an ad-hoc scheme. The s=
erver determines what each end should do and sends ad-hoc messages to the e=
ndpoints which then generate the SDP answer that causes the desired outcome=
.=0A=0AIt should be noted that extracting a full set of capabilities when o=
nly offer-answer is available via the API is particularly painful, as it mi=
ght be necessary to explore a very wide space of fake offers to get answers=
 that clarify things like "can do G.722 audio but only if not doing H.264 v=
ideo".=0A=0AWe might be able to learn from some previous experiences:=0A=0A=
1. Web-based email. Web browsers run Javascript that uses ad-hoc signaling =
to/from the web site in order to send and receive email. The browser does n=
ot have SMTP, POP, or IMAP implementations, or even know how to parse RFC82=
2 headers on its own. This argues for A1, B1 (or maybe B2), C1.=0A=0A2. Web=
-based image display. Web browsers send an "Accept" header via HTTP indicat=
ing which MIME types they can handle. Knowing whether a browser can do JPEG=
 is as simple as looking for "image/jpeg" in the header. This is probably a=
t the wrong layer for sophisticated web applications (as this isn't about w=
hat comes back over HTTP, but what can be supported over a separate channel=
), but might be appropriate if we could agree on a small number of supporte=
d codecs and simply have the browser tell the server in a similar manner wh=
ether or not it can do RTCWEB.=0A=0A3. SIP. SIP endpoints use SDP directly =
between the endpoints (though possibly with intermediaries that rarely chan=
ge the SDP) to do offer-answer. This is most like A2, B3 (or B4), C3 (or C2=
).=0A=0A4. MGCP. MGCP uses SDP for both capabilities and settings, but the =
server is in control. This is most like A2, B3, C1.=0A=0A5. H.323 (H.245) u=
ses a protocol other than SDP for both capabilities and settings, but the s=
erver is in control. This is like A3 (or some other encoding), B2, C1=0A=0A=
As for connection negotiation, most of the separation previously also appli=
es... again whether to put the ICE candidates into SDP or not, whether to h=
ave APIs that take blobs or individual calls, etc.=0A=0ABut the other issue=
 that arises when we start talking about connection negotiation is that if =
you choose SDP offer-answer for codec negotiation, and SDP for ICE candidat=
es, there is the temptation to combine the SDP for the two, thus conflating=
 the process of establishing a logical channel over which various types of =
media might flow, and establishing just which media should be sent over tha=
t channel.=0A=0AMatthew Kaufman=0A=0A______________________________________=
_________=0Artcweb mailing list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/ma=
ilman/listinfo/rtcweb
--0-293671360-1312666903=:37487
Content-Type: text/html; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable

<html><body><div style=3D"color:#000; background-color:#fff; font-family:ti=
mes new roman, new york, times, serif;font-size:12pt"><div>Matt, Thank you =
for nicely presenting various sets of session capability transfer methods.<=
/div><div>I like X1 and X2, and also Y1 and Y2. &nbsp;Although I do not hav=
e any opinion on X3, it is okay to add this use case in the document. Diver=
se methods create innovation opportunities although create interoperability=
 challenges.&nbsp;</div><div><br></div><div>Although in a utopian network, =
it is tempting to consider that no-middle box (e.g., network transcoder) sh=
ould exist, unfortunately, not all end system support the same sets of code=
c in the today's network. Therefore, a network transcoding system may be ne=
eded; thus, the need of servers in the network.<br></div><div><br></div><di=
v>Apart from above method selections, in details level, we will need some i=
mportant parameters. I am writing here although this thread may not be
 the right forum.</div><div>One important point is to ensure we have a fiel=
d or parameter, or arrangement order to depict media capability preference.=
 For example, in SDP, the first codec in the list is preferred. This is hel=
pful for both network (e.g., transcoding mediation) and end systems, in whi=
ch those support multiple codecs but would like to select the default codec=
. In addition, we have to consider a parameter for "force: no &lt;parameter=
&gt;" to explicitly mention unwanted media capability. For example, an end =
user or network may like to preclude a "bad-codec". In this case, the end s=
ystem or network should be able to negotiate "force: no bad-codec". &nbsp;<=
/div><div><br></div><div>In DTMF codec negotiation, we will need to ensure =
that a parameter clearly indicates which DTMF transfer method is supported.=
 Currently, RFC 2833 &nbsp;telephone-event expresses intention of out-of-ba=
nd RTP DTMF digit transfer. In my opinion, we do not have a tool to
 deny SIP INFO-based DTMF digit transfer. &nbsp;Some transcoder in the netw=
ork or end user codec may prefer not to use INFO-based DTMF digit transfer.=
 In this case, a parameter such as "force: no info-dtmf-event" will be help=
ful.</div><div><br></div><div>Regards,<br>Sohel Khan<br></div><div style=3D=
"font-size: 12pt; font-family: 'times new roman', 'new york', times, serif;=
 "><div style=3D"font-size: 12pt; font-family: 'times new roman', 'new york=
', times, serif; "><font size=3D"2" face=3D"Arial"><hr size=3D"1"><b><span =
style=3D"font-weight:bold;">From:</span></b> Matthew Kaufman &lt;matthew.ka=
ufman@skype.net&gt;<br><b><span style=3D"font-weight: bold;">To:</span></b>=
 "rtcweb@ietf.org" &lt;rtcweb@ietf.org&gt;<br><b><span style=3D"font-weight=
: bold;">Sent:</span></b> Friday, August 5, 2011 2:33 PM<br><b><span style=
=3D"font-weight: bold;">Subject:</span></b> [rtcweb] codec and connection n=
egotiation<br></font><br>I put this together to try to help myself, and per=
haps others,
 understand the various ways in which codec and connection negotiation migh=
t be decomposed and then put together for RTCWEB applications. It is a bit =
of a stream of consciousness, but hopefully we can get some good discussion=
 provoked.<br><br>A. How to encode codec capabilities or choices<br><br>A1 =
- Don't provide a canonical way of encoding codec settings. The choice of w=
hich codec and what parameters is encoded in an ad-hoc way that requires no=
 standardization.<br><br>A2 - Encode codec settings using SDP. The choice o=
f which codec and what parameters is encoded using the syntax and semantics=
 of SDP and reuses the SDP standardization work.<br><br>A3 - Encode codec s=
ettings using a JSON encoding of SDP. The choice of which codec and what pa=
rameters is using the semantics of SDP but a JSON syntax. Reuses the SDP st=
andardization plus new standards around how to encode SDP in JSON.<br><br>B=
. How to expose codec capabilities and controls<br><br>B1. Expose
 codec capabilities and settings via individual Javascript APIs. For exampl=
e one might say "camera.encodeMode =3D "H.264"; camera.encodeWidth =3D 640;=
" or "if (camera.canEncode("H.264")) ..."<br><br>B2. Expose codec capabilit=
ies and settings via a combined Javascript API that concatenates all the ca=
pabilities and settings into a single object. Calling "camera.encodeCapabil=
ities()" would return a large object that is a full enumeration of what it =
can do.<br><br>B3. Expose codec capabilities and settings via a combined Ja=
vascript API that produces and consumes SDP strings.<br><br>B4. Don't expos=
e codec capbilities and settings via Javascript, rather handle this outside=
 of Javascript.<br><br>C. How to agree on which codec and settings to use<b=
r><br>C1. The server receives information about the capabilities, decides w=
hat settings should be used, and communicates them to the endpoints<br><br>=
C2. The endpoints agree using offer-answer, using the server as a
 communication channel<br><br>C3. The endpoints agree using offer-answer, d=
irectly between the two endpoints<br><br><br>(There's obviously a few addit=
ional nuances, like whether to use something like RFC 5939 SDP capability n=
egotiation, that aren't well-captured above.)<br><br>Now, with the above, w=
e find that some of the models that have been discussed are combinations of=
 the above choices.<br><br>X1 - Use A2, B3, C2 - The endpoints generate SDP=
 for offer-answer by firing an SDP event, having it sent via a server to th=
e far end, where it is injected as SDP into the Javascript.<br><br>X2 - Use=
 A1, B1, C1 - The endpoints run Javascript that inspects their capabilities=
, sends that to the server which decides what is best, and sends back infor=
mation that the Javascript uses to set up the encoders.<br><br>X3 - Use A2,=
 B4, C3 - The endpoints establish a communication channel using ICE and an =
out-of-band channel, they then use SDP inside of SIP to directly
 negotiate using offer-answer, the server has no information about what was=
 chosen and does not participate.<br><br>We also find that almost all the m=
odels can map, though with varying degrees of pain, to others.<br><br>For i=
nstance:<br><br>Y1 - Use A2, B1, C2 - The endpoints run Javascript that ins=
pects their capabilities and encodes it into an SDP string. This is sent vi=
a the server to the far end, where the SDP is parsed in Javascript and the =
individual APIs called to implement SDP offer-answer.<br><br>Y2 - Use A1, B=
3, C1 - The endpoints generate SDP for offer-answer by firing an SDP event.=
 The endpoint then runs Javascript that extracts from the SDP all the indiv=
idual parameters and re-encodes them using an ad-hoc scheme. The server det=
ermines what each end should do and sends ad-hoc messages to the endpoints =
which then generate the SDP answer that causes the desired outcome.<br><br>=
It should be noted that extracting a full set of capabilities when
 only offer-answer is available via the API is particularly painful, as it =
might be necessary to explore a very wide space of fake offers to get answe=
rs that clarify things like "can do G.722 audio but only if not doing H.264=
 video".<br><br>We might be able to learn from some previous experiences:<b=
r><br>1. Web-based email. Web browsers run Javascript that uses ad-hoc sign=
aling to/from the web site in order to send and receive email. The browser =
does not have SMTP, POP, or IMAP implementations, or even know how to parse=
 RFC822 headers on its own. This argues for A1, B1 (or maybe B2), C1.<br><b=
r>2. Web-based image display. Web browsers send an "Accept" header via HTTP=
 indicating which MIME types they can handle. Knowing whether a browser can=
 do JPEG is as simple as looking for "image/jpeg" in the header. This is pr=
obably at the wrong layer for sophisticated web applications (as this isn't=
 about what comes back over HTTP, but what can be supported over a
 separate channel), but might be appropriate if we could agree on a small n=
umber of supported codecs and simply have the browser tell the server in a =
similar manner whether or not it can do RTCWEB.<br><br>3. SIP. SIP endpoint=
s use SDP directly between the endpoints (though possibly with intermediari=
es that rarely change the SDP) to do offer-answer. This is most like A2, B3=
 (or B4), C3 (or C2).<br><br>4. MGCP. MGCP uses SDP for both capabilities a=
nd settings, but the server is in control. This is most like A2, B3, C1.<br=
><br>5. H.323 (H.245) uses a protocol other than SDP for both capabilities =
and settings, but the server is in control. This is like A3 (or some other =
encoding), B2, C1<br><br>As for connection negotiation, most of the separat=
ion previously also applies... again whether to put the ICE candidates into=
 SDP or not, whether to have APIs that take blobs or individual calls, etc.=
<br><br>But the other issue that arises when we start talking about
 connection negotiation is that if you choose SDP offer-answer for codec ne=
gotiation, and SDP for ICE candidates, there is the temptation to combine t=
he SDP for the two, thus conflating the process of establishing a logical c=
hannel over which various types of media might flow, and establishing just =
which media should be sent over that channel.<br><br>Matthew Kaufman<br><br=
>_______________________________________________<br>rtcweb mailing list<br>=
<a ymailto=3D"mailto:rtcweb@ietf.org" href=3D"mailto:rtcweb@ietf.org">rtcwe=
b@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><=
br></div></div></div></body></html>
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From: Sohel Khan <sohel_khan777@yahoo.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
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Thanks Randell.=0AIn my opinion, both local and remote "voicemail" and vide=
omail" use cases will be helpful.=0A=A0=0ARegards,=0ASohel Khan=0A=0A=0A___=
_____________________________=0AFrom: Randell Jesup <randell-ietf@jesup.org=
>=0ATo: rtcweb@ietf.org; public-webrtc@w3.org=0ASent: Thursday, August 4, 2=
011 11:03 AM=0ASubject: Re: [rtcweb] Use cases: summary of status=0A=0A...<=
deleted>...=0A=0A=0AWould this cover all "voicemail" and "videomail" cases?=
=A0 I.e. having a client=0Aaccept connections in the background if the call=
 is not answered, optionally=0Aplay a message, and record incoming audio/vi=
deo, and allow the remote user to=0Ainteract with it.=A0 Also remote playba=
ck of messages.=A0 If not (and if it's not=0Acovered by the current documen=
t, we need to add this usecase.=0A=0ANote that there are two variants: one =
local (local client handles it, which=0Ahas more security issues around cam=
era access and pre-approval), and one for=0Aremote recording (after some ti=
me with no answer or if not "registered" call=0Ais forwarded to a message s=
erver that answers it).=A0 Note that there are=0Asecurity identity issues h=
ere (key chains, etc).=0A=0A....<deleted>....=0A-- =0ARandell Jesup=0Arande=
ll-ietf@jesup.org=0A=0A_______________________________________________=0Art=
cweb mailing list=0Artcweb@ietf.org=0Ahttps://www.ietf.org/mailman/listinfo=
/rtcweb
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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><span>Thanks Ran=
dell.</span></div><div><span>In my opinion, both local and remote "voicemai=
l" and videomail" use cases will be helpful.</span></div><div>&nbsp;</div><=
div>Regards,<br>Sohel Khan<br></div><div style=3D"font-size: 12pt; font-fam=
ily: 'times new roman', 'new york', times, serif; "><div style=3D"font-size=
: 12pt; font-family: 'times new roman', 'new york', times, serif; "><font s=
ize=3D"2" face=3D"Arial"><hr size=3D"1"><b><span style=3D"font-weight:bold;=
">From:</span></b> Randell Jesup &lt;randell-ietf@jesup.org&gt;<br><b><span=
 style=3D"font-weight: bold;">To:</span></b> rtcweb@ietf.org; public-webrtc=
@w3.org<br><b><span style=3D"font-weight: bold;">Sent:</span></b> Thursday,=
 August 4, 2011 11:03 AM<br><b><span style=3D"font-weight: bold;">Subject:<=
/span></b> Re: [rtcweb] Use cases: summary of
 status<br></font><br>...&lt;deleted&gt;...</div><div style=3D"font-family:=
 times new roman, new york, times, serif; font-size: 12pt;" class=3D"yui_3_=
2_0_4_131266795137570"><br></div><div style=3D"font-family: times new roman=
, new york, times, serif; font-size: 12pt;" class=3D"yui_3_2_0_4_1312667951=
37570"><br>Would this cover all "voicemail" and "videomail" cases?&nbsp; I.=
e. having a client<br>accept connections in the background if the call is n=
ot answered, optionally<br>play a message, and record incoming audio/video,=
 and allow the remote user to<br>interact with it.&nbsp; Also remote playba=
ck of messages.&nbsp; If not (and if it's not<br>covered by the current doc=
ument, we need to add this usecase.<br><br>Note that there are two variants=
: one local (local client handles it, which<br>has more security issues aro=
und camera access and pre-approval), and one for<br>remote recording (after=
 some time with no answer or if not "registered" call<br>is forwarded to a
 message server that answers it).&nbsp; Note that there are<br>security ide=
ntity issues here (key chains, etc).<br><br>....&lt;deleted&gt;....<br>-- <=
br>Randell Jesup<br><a ymailto=3D"mailto:randell-ietf@jesup.org" href=3D"ma=
ilto:randell-ietf@jesup.org">randell-ietf@jesup.org</a><br><br>____________=
___________________________________<br>rtcweb mailing list<br><a ymailto=3D=
"mailto:rtcweb@ietf.org" href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a=
><br><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_bl=
ank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br><br><br></div></di=
v></div></body></html>
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From christer.holmberg@ericsson.com  Sat Aug  6 15:22:54 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>, "Parthasarathi R (partr)" <partr@cisco.com>
Date: Sun, 7 Aug 2011 00:19:50 +0200
Thread-Topic: [rtcweb] Usecase & architecture: Browser application with separate webserver & voipserver
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Subject: Re: [rtcweb] Usecase & architecture: Browser application with separate webserver & voipserver
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Hi,

If I understand Matthew correctly, he is saing that, from the browser's per=
spective, the VoIP server is acting as a web server.

I agree, and I don't see any new browser requirements derived from that.

(There might be requirements on the VoIP server, but that is outside the sc=
ope of RTCWEB).

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Matthe=
w Kaufman [matthew.kaufman@skype.net]
Sent: Saturday, August 06, 2011 7:36 PM
To: Parthasarathi R (partr)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Usecase & architecture: Browser application with sepa=
rate webserver & voipserver

On 8/6/2011 6:19 AM, Parthasarathi R (partr) wrote:
Hi all,

browser application should have the mechanism by which it interacts with vo=
ipserver directly instead of depend on the webserver. This usecase provides=
 the flexibility for the webdeveloper to focus on the webdevelopment and us=
e the existing voipservers for voip services by just invoking the API.

I'm not very sure whether this usecase is same as sec 4.3.1 as there is no =
protocol architecture shown:

         browser--------webservers---web services
         (Javascript)
             |
             ----------------voipservers-----VoIP entity (browser)

This architecture facilites for webdeveloper to choose the different vendor=
 for webservers & voipservers. It is possible for webserver & voipserver co=
-located but not mandatory. This architecture is slightly different from dr=
aft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC Trapezoid).  Please le=
t me know your opinion on the same.


My opinion is that as long as the "voipservers" use HTTP-based signaling an=
d RTCWEB-compatible VoIP, then that's fine, and indistinguishable from othe=
r use cases. But if the "voipservers" are existing SIP servers then no, I d=
on't want additional inflexible code in the browser that turns it into a SI=
P phone.

Matthew Kaufman

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Matthew/Christer,

I asked this usecase for two reasons:

1) Whether any new authentication/security/privacy consideration has to
be taken into account in case separate server involved for real-time
traffic alone. The issues may be
 a) The single authentication will not work
 b) The same identity may not apply.=20
I'm curious to know whether it will translate into new requirement in
browser or in API.

2) I understand that your proposal is to introduce new gateway (RTCWEB
to VoIP GW) for making the interop for every session even between two
RTCWeb servers /RTCWeb client to other existing VoIP clients. As your
proposal, I'm seeing IETF recommends two real-time protocol with
different solution for the same service depends upon application in the
system. For example, any solution developer has to consider the
following topology:

a) Desktop application interacts with VoIP server (use SIP)
        application <------->SIP server
b) Browser application interacts with VoIP server (use HTTP)
        browser<------>RTCweb server & SIP client (new GW)<------>SIP
server

New Gateway is outside the scope of RTCWeb and IETF which provides the
opportunity for innovative way of building gateways but may leads to
interop issue due to mapping between RTCWebserver & SIP client w.r.t
mid-call services. Also, I'm not seeing JavaScript API design as the
limiting factor for this choice.=20

Could you please explain the strong reasons for browser *MUST NOT*
interop with any existing SIP entity or other VOIP entity directly.  =09

Thanks
Partha

> -----Original Message-----
> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com]=20
> Sent: Sunday, August 07, 2011 3:50 AM
> To: Matthew Kaufman; Parthasarathi R (partr)
> Cc: rtcweb@ietf.org
> Subject: RE: [rtcweb] Usecase & architecture: Browser=20
> application with separate webserver & voipserver
>=20
> Hi,
>=20
> If I understand Matthew correctly, he is saing that, from the=20
> browser's perspective, the VoIP server is acting as a web server.
>=20
> I agree, and I don't see any new browser requirements derived=20
> from that.
>=20
> (There might be requirements on the VoIP server, but that is=20
> outside the scope of RTCWEB).
>=20
> Regards,
>=20
> Christer
>=20
> ________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On=20
> Behalf Of Matthew Kaufman [matthew.kaufman@skype.net]
> Sent: Saturday, August 06, 2011 7:36 PM
> To: Parthasarathi R (partr)
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Usecase & architecture: Browser=20
> application with separate webserver & voipserver
>=20
> On 8/6/2011 6:19 AM, Parthasarathi R (partr) wrote:
> Hi all,
>=20
> browser application should have the mechanism by which it=20
> interacts with voipserver directly instead of depend on the=20
> webserver. This usecase provides the flexibility for the=20
> webdeveloper to focus on the webdevelopment and use the=20
> existing voipservers for voip services by just invoking the API.
>=20
> I'm not very sure whether this usecase is same as sec 4.3.1=20
> as there is no protocol architecture shown:
>=20
>          browser--------webservers---web services
>          (Javascript)
>              |
>              ----------------voipservers-----VoIP entity (browser)
>=20
> This architecture facilites for webdeveloper to choose the=20
> different vendor for webservers & voipservers. It is possible=20
> for webserver & voipserver co-located but not mandatory. This=20
> architecture is slightly different from=20
> draft-rosenberg-rtcweb-framework-00 fig 2 (Browser RTC=20
> Trapezoid).  Please let me know your opinion on the same.
>=20
>=20
> My opinion is that as long as the "voipservers" use=20
> HTTP-based signaling and RTCWEB-compatible VoIP, then that's=20
> fine, and indistinguishable from other use cases. But if the=20
> "voipservers" are existing SIP servers then no, I don't want=20
> additional inflexible code in the browser that turns it into=20
> a SIP phone.
>=20
> Matthew Kaufman
>=20

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On 08/06/11 11:43, Parthasarathi R (partr) wrote:
> Matthew,
>
> I'm seeing two aspects here
>
> 1) API: Level of control has to be given to the Javascript programmer.
> Here, the discussion has to focus on the abstraction by which
> application shall be developed easily without much understanding the
> protocol intricacies. Even API shall be provided to the extent that
> username&  destination host (xyz@abc.com) as a input shall trigger the
> session creation with default codec.
Partha,

two comments:
- I do not see a requirement that all scenarios use usernames.
- in "xyz@abc.com", "xyz" is not necessarily an username, and "abc.com" 
is not necessarily a hostname.
Even when identity is mediated via the Domain Name System, there may be 
no host with that name.

> 2) Protocol: The protocol selection has to be based on better interop
> with other browsers and other VoIP entity also. I'm not favoring for
> creating different island of realtime protocol which has to interop with
> "n" of gateways.
>
> Please read inline for more comments.
>
> Thanks
> Partha
>
>
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
> Of Matthew Kaufman
> Sent: Saturday, August 06, 2011 12:24 AM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] codec and connection negotiation
>
> Now, to follow up on my previous message with a bit of opinion...
>
> I think we should not use SDP as the API,
>
> <partha>  It is fine to discuss which level of control has to be
> provided.</partha>
>
>   and we should not have the browser implementing SDP offer-answer.
> <partha>  I could not see the strong reason for not doing so.</partha>
>
> 1. If SDP offer-answer is all that is available to the Javascript
> programmer (and server programmer), it becomes very difficult to know
> what the full capability set is without complex probing. This
> significantly complicates the building of future innovative applications
> on top of the browser as platform. Many of the current applications that
> show off browser capabilities do so by using capabilities that were
> already present for other reasons, and we can expect the same innovation
> to occur here, if we provide the tools.
>
> 2. Using SDP offer-answer has the advantage of reusing all the IETF
> standards work that occurs to define SDP when a new codec comes along.
> But it also has the *disadvantage* of having to wait for the IETF to
> produce these standards, which may be incomplete, or unable to control
> parameters which are necessary for web applications.
> <partha>  I prefer standard over proprietary mechanism because standard
> may delay but proprietary is normal uncontrolled and impossible to
> interop with other webservers</partha>
>
> 3. Anything that is missing in SDP (example: forcing the Opus codec to
> "music" mode for encoding) will still need to be exposed as a Javascript
> API on the encoder. Thus we end up with a mix of possibly-conflicting
> settings that are adjusted via the explicit API and the opaque SDP API.
> <partha>  It is possible to integrate unknown SDP attribute&  known SDP
> attribute using the proper API design. It is the basic requirement in
> any SDP stack API today.</partha>
>
> 4. Other work in browsers (like recording camera and microphone to disk)
> will require the ability to directly control the encoders. If these APIs
> exist then there will definitely be conflicting settings. What happens
> if you feed in an SDP answer that requires VP8 encoding and then set the
> encoder to H.264 mode via the Javascript?
> <partha>  I'm reading your query as how to priortize the value in case
> extra environment change the value. It has to be worked out</partha>
>
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From csp@csperkins.org  Mon Aug  8 03:28:48 2011
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[inline]

On 5 Aug 2011, at 19:33, Matthew Kaufman wrote:
> I put this together to try to help myself, and perhaps others, =
understand the various ways in which codec and connection negotiation =
might be decomposed and then put together for RTCWEB applications. It is =
a bit of a stream of consciousness, but hopefully we can get some good =
discussion provoked.
>=20
> A. How to encode codec capabilities or choices
>=20
> A1 - Don't provide a canonical way of encoding codec settings. The =
choice of which codec and what parameters is encoded in an ad-hoc way =
that requires no standardization.
>=20
> A2 - Encode codec settings using SDP. The choice of which codec and =
what parameters is encoded using the syntax and semantics of SDP and =
reuses the SDP standardization work.
>=20
> A3 - Encode codec settings using a JSON encoding of SDP. The choice of =
which codec and what parameters is using the semantics of SDP but a JSON =
syntax. Reuses the SDP standardization plus new standards around how to =
encode SDP in JSON.

The codec name and parameters are not defined in terms of SDP. Rather, =
SDP uses the MIME type of the codec, and the registered parameters of =
that MIME type. I don't much care whether the syntax used to convey =
codec MIME types and parameters is SDP, JSON, XML, or whatever, but I =
strongly suggest reusing that existing database of codec MIME types =
(registered following RFC 3555), rather than trying to re-invent it.=20

Colin




> B. How to expose codec capabilities and controls
>=20
> B1. Expose codec capabilities and settings via individual Javascript =
APIs. For example one might say "camera.encodeMode =3D "H.264"; =
camera.encodeWidth =3D 640;" or "if (camera.canEncode("H.264")) ..."
>=20
> B2. Expose codec capabilities and settings via a combined Javascript =
API that concatenates all the capabilities and settings into a single =
object. Calling "camera.encodeCapabilities()" would return a large =
object that is a full enumeration of what it can do.
>=20
> B3. Expose codec capabilities and settings via a combined Javascript =
API that produces and consumes SDP strings.
>=20
> B4. Don't expose codec capbilities and settings via Javascript, rather =
handle this outside of Javascript.
>=20
> C. How to agree on which codec and settings to use
>=20
> C1. The server receives information about the capabilities, decides =
what settings should be used, and communicates them to the endpoints
>=20
> C2. The endpoints agree using offer-answer, using the server as a =
communication channel
>=20
> C3. The endpoints agree using offer-answer, directly between the two =
endpoints
>=20
>=20
> (There's obviously a few additional nuances, like whether to use =
something like RFC 5939 SDP capability negotiation, that aren't =
well-captured above.)
>=20
> Now, with the above, we find that some of the models that have been =
discussed are combinations of the above choices.
>=20
> X1 - Use A2, B3, C2 - The endpoints generate SDP for offer-answer by =
firing an SDP event, having it sent via a server to the far end, where =
it is injected as SDP into the Javascript.
>=20
> X2 - Use A1, B1, C1 - The endpoints run Javascript that inspects their =
capabilities, sends that to the server which decides what is best, and =
sends back information that the Javascript uses to set up the encoders.
>=20
> X3 - Use A2, B4, C3 - The endpoints establish a communication channel =
using ICE and an out-of-band channel, they then use SDP inside of SIP to =
directly negotiate using offer-answer, the server has no information =
about what was chosen and does not participate.
>=20
> We also find that almost all the models can map, though with varying =
degrees of pain, to others.
>=20
> For instance:
>=20
> Y1 - Use A2, B1, C2 - The endpoints run Javascript that inspects their =
capabilities and encodes it into an SDP string. This is sent via the =
server to the far end, where the SDP is parsed in Javascript and the =
individual APIs called to implement SDP offer-answer.
>=20
> Y2 - Use A1, B3, C1 - The endpoints generate SDP for offer-answer by =
firing an SDP event. The endpoint then runs Javascript that extracts =
from the SDP all the individual parameters and re-encodes them using an =
ad-hoc scheme. The server determines what each end should do and sends =
ad-hoc messages to the endpoints which then generate the SDP answer that =
causes the desired outcome.
>=20
> It should be noted that extracting a full set of capabilities when =
only offer-answer is available via the API is particularly painful, as =
it might be necessary to explore a very wide space of fake offers to get =
answers that clarify things like "can do G.722 audio but only if not =
doing H.264 video".
>=20
> We might be able to learn from some previous experiences:
>=20
> 1. Web-based email. Web browsers run Javascript that uses ad-hoc =
signaling to/from the web site in order to send and receive email. The =
browser does not have SMTP, POP, or IMAP implementations, or even know =
how to parse RFC822 headers on its own. This argues for A1, B1 (or maybe =
B2), C1.
>=20
> 2. Web-based image display. Web browsers send an "Accept" header via =
HTTP indicating which MIME types they can handle. Knowing whether a =
browser can do JPEG is as simple as looking for "image/jpeg" in the =
header. This is probably at the wrong layer for sophisticated web =
applications (as this isn't about what comes back over HTTP, but what =
can be supported over a separate channel), but might be appropriate if =
we could agree on a small number of supported codecs and simply have the =
browser tell the server in a similar manner whether or not it can do =
RTCWEB.
>=20
> 3. SIP. SIP endpoints use SDP directly between the endpoints (though =
possibly with intermediaries that rarely change the SDP) to do =
offer-answer. This is most like A2, B3 (or B4), C3 (or C2).
>=20
> 4. MGCP. MGCP uses SDP for both capabilities and settings, but the =
server is in control. This is most like A2, B3, C1.
>=20
> 5. H.323 (H.245) uses a protocol other than SDP for both capabilities =
and settings, but the server is in control. This is like A3 (or some =
other encoding), B2, C1
>=20
> As for connection negotiation, most of the separation previously also =
applies... again whether to put the ICE candidates into SDP or not, =
whether to have APIs that take blobs or individual calls, etc.
>=20
> But the other issue that arises when we start talking about connection =
negotiation is that if you choose SDP offer-answer for codec =
negotiation, and SDP for ICE candidates, there is the temptation to =
combine the SDP for the two, thus conflating the process of establishing =
a logical channel over which various types of media might flow, and =
establishing just which media should be sent over that channel.
>=20
> Matthew Kaufman
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




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Harald,

<snip>
> two comments:
> - I do not see a requirement that all scenarios use usernames.
> - in "xyz@abc.com", "xyz" is not necessarily an username, and=20
> "abc.com"=20
> is not necessarily a hostname.
> Even when identity is mediated via the Domain Name System,=20
> there may be no host with that name.
</snip>

I provided username just as an example and the identity discussion is
not related to this mail thread. Intention of my example in the mail
thread is to highlight high level API possible from Javascript.=20

I'm fine with any identity mechanism as long as multiple RTCWeb servers
are able to interop with the given identity.

Thanks
Partha

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
> Sent: Monday, August 08, 2011 2:29 PM
> To: Parthasarathi R (partr)
> Cc: Matthew Kaufman; rtcweb@ietf.org
> Subject: Re: [rtcweb] codec and connection negotiation
>=20
> On 08/06/11 11:43, Parthasarathi R (partr) wrote:
> > Matthew,
> >
> > I'm seeing two aspects here
> >
> > 1) API: Level of control has to be given to the Javascript=20
> programmer.
> > Here, the discussion has to focus on the abstraction by which=20
> > application shall be developed easily without much=20
> understanding the=20
> > protocol intricacies. Even API shall be provided to the extent that=20
> > username&  destination host (xyz@abc.com) as a input shall=20
> trigger the=20
> > session creation with default codec.
> Partha,
>=20
> two comments:
> - I do not see a requirement that all scenarios use usernames.
> - in "xyz@abc.com", "xyz" is not necessarily an username, and=20
> "abc.com"=20
> is not necessarily a hostname.
> Even when identity is mediated via the Domain Name System,=20
> there may be no host with that name.
>=20
> > 2) Protocol: The protocol selection has to be based on=20
> better interop=20
> > with other browsers and other VoIP entity also. I'm not=20
> favoring for=20
> > creating different island of realtime protocol which has to interop=20
> > with "n" of gateways.
> >
> > Please read inline for more comments.
> >
> > Thanks
> > Partha
> >
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On=20
> > Behalf Of Matthew Kaufman
> > Sent: Saturday, August 06, 2011 12:24 AM
> > To: rtcweb@ietf.org
> > Subject: Re: [rtcweb] codec and connection negotiation
> >
> > Now, to follow up on my previous message with a bit of opinion...
> >
> > I think we should not use SDP as the API,
> >
> > <partha>  It is fine to discuss which level of control has to be=20
> > provided.</partha>
> >
> >   and we should not have the browser implementing SDP offer-answer.
> > <partha>  I could not see the strong reason for not doing=20
> so.</partha>
> >
> > 1. If SDP offer-answer is all that is available to the Javascript=20
> > programmer (and server programmer), it becomes very=20
> difficult to know=20
> > what the full capability set is without complex probing. This=20
> > significantly complicates the building of future innovative=20
> > applications on top of the browser as platform. Many of the current=20
> > applications that show off browser capabilities do so by using=20
> > capabilities that were already present for other reasons,=20
> and we can=20
> > expect the same innovation to occur here, if we provide the tools.
> >
> > 2. Using SDP offer-answer has the advantage of reusing all the IETF=20
> > standards work that occurs to define SDP when a new codec=20
> comes along.
> > But it also has the *disadvantage* of having to wait for=20
> the IETF to=20
> > produce these standards, which may be incomplete, or unable=20
> to control=20
> > parameters which are necessary for web applications.
> > <partha>  I prefer standard over proprietary mechanism because=20
> > standard may delay but proprietary is normal uncontrolled and=20
> > impossible to interop with other webservers</partha>
> >
> > 3. Anything that is missing in SDP (example: forcing the=20
> Opus codec to=20
> > "music" mode for encoding) will still need to be exposed as a=20
> > Javascript API on the encoder. Thus we end up with a mix of=20
> > possibly-conflicting settings that are adjusted via the=20
> explicit API and the opaque SDP API.
> > <partha>  It is possible to integrate unknown SDP attribute&  known=20
> > SDP attribute using the proper API design. It is the basic=20
> requirement=20
> > in any SDP stack API today.</partha>
> >
> > 4. Other work in browsers (like recording camera and microphone to=20
> > disk) will require the ability to directly control the encoders. If=20
> > these APIs exist then there will definitely be conflicting=20
> settings.=20
> > What happens if you feed in an SDP answer that requires VP8=20
> encoding=20
> > and then set the encoder to H.264 mode via the Javascript?
> > <partha>  I'm reading your query as how to priortize the=20
> value in case=20
> > extra environment change the value. It has to be worked out</partha>
> >
> >
> > Matthew Kaufman
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
>=20

From harald@alvestrand.no  Mon Aug  8 05:51:38 2011
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Subject: Re: [rtcweb] New use case: Large multiparty session
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On 08/03/11 15:34, Arno Bakker wrote:
> On 03/08/2011 14:51, Randell Jesup wrote:
>> On 8/3/2011 4:36 AM, Arno Bakker wrote:
> >>
>>> assume a client is involved in a large multiparty session but is
>>> unable (or unwilling) to upload his signals to all participants.
>>>
>>> This suggests forwarding support from the network is required, e.g.
>>> some relay server, multicast, or the future IETF peer-to-peer streaming
>>> protocol (PPSP).
>>
>> So the use case is:
>>
>> User is part of a multiparty (3 or greater) session, but for one of
>> several reasons (such as
>> available upstream bandwidth) cannot send media to all other
>> participants ("mesh" conferencing).
>> This requires that another node, either a central network node (such as
>> a conferencing server)
>> or another member in the session, relay or mix the user's media to
>> distribute to the rest of the
>> members of the session.
>>
>
>
> Yes. Although your wording suggests a single relay node (central or 
> member) is used. Please leave the option of true peer-to-peer 
> distribution (or IP multicast) of the signal open.
In draft-ietf-rtcweb-use-cases-and-requirements-01.txt, the "central 
server" model is listed in section 4.3.3, "Video conferencing system 
with central server".

I am strongly opposed to adding use cases (which in turn generate 
requirements) to this first iteration of RTCWEB that we do not know how 
to fulfil with any currently deployed standard technology, so I would 
definitely want to leave the "peer to peer distribution or IP multicast" 
out of the picture for now. It can be added to a working appendix of 
"Use cases that have been discussed, but will not be considered as 
requirements in this iteration of RTCWEB".

From harald@alvestrand.no  Mon Aug  8 05:55:17 2011
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On 08/03/11 14:12, Parthasarathi R (partr) wrote:
> Hi Magnus,
>
> The current RTCWeb overview, usecase&  requirement document does not ha=
ve architecture diagrams. Please clarify which one of the RTCWeb WG item =
below will cover RTCWeb architecture.
If draft-ietf-rtcweb-overview does not describe the architecture in=20
sufficient detail, it needs updating. It is the document in the document =

hierarchy that should have this role.

Some parts of it are not explicit because no consensus has been=20
forthcoming, others because I have not understood yet what people need=20
to hear.

What do you think it should say?
> Thanks
> Partha
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus Westerlund
>> Sent: Tuesday, August 02, 2011 5:47 PM
>> To: rtcweb@ietf.org
>> Subject: [rtcweb] New Milestones for RTCWEB WG
>>
>> WG,
>>
>> As raised in the WG meeting last week we plan to submit new
>> milestones.
>> These is to make the time line a bit more realistic to
>> actually be meet.
>> They also try to adjust to the picked document structure.
>>
>> Here they are:
>> Oct 2011 Send Use Cases
>> (draft-ietf-rtcweb-use-cases-and-requirements),
>> Overview (draft-ietf-rtcweb-overview), Security and Privacy Documents
>> (I-D) to W3C
>>
>> Dec 2011 Send Use Cases (draft-ietf-rtcweb-use-cases-and-requirements)=

>> document to IESG for publication as Informational
>>
>> Apr 2012 Send W3C input on what requirements the current
>> solution has on
>> API(s)
>>
>> May 2012 Send Overview (draft-ietf-rtcweb-overview), Security
>> and Privacy Model documents to IESG for publication as Informational
>>
>> May 2012 Send Signalling and Negotiation, and NAT Traversal
>> to IESG for publication as Proposed Standard
>>
>> Jun 2012 Send Media Transport, Media Processing, and Codecs
>> to IESG for publication as Proposed Standard
>>
>> Jun 2012 Send Datagram Transport for non-media data to IESG
>> for publication as Proposed Standard
>>
>> These will soon be submitted for official publication. If you
>> have any issues please raise them on the list and we can discuss them.=

>>
>> cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------=

>> Multimedia Technologies, Ericsson Research EAB/TVM
>> ----------------------------------------------------------------------=

>> Ericsson AB                | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden| mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------=

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



From harald@alvestrand.no  Mon Aug  8 06:09:00 2011
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Trying to doodle on the opinions (the alternative lists were quite 
useful to clear the mind - I liked them a lot, and will return to them!)

On 08/05/11 20:53, Matthew Kaufman wrote:
> Now, to follow up on my previous message with a bit of opinion...
>
> I think we should not use SDP as the API, and we should not have the 
> browser implementing SDP offer-answer.
>
> 1. If SDP offer-answer is all that is available to the Javascript 
> programmer (and server programmer), it becomes very difficult to know 
> what the full capability set is without complex probing. This 
> significantly complicates the building of future innovative 
> applications on top of the browser as platform. Many of the current 
> applications that show off browser capabilities do so by using 
> capabilities that were already present for other reasons, and we can 
> expect the same innovation to occur here, if we provide the tools.
This may be heretical ... but do we actually need to enumerate all 
capabilities?
It seems to me that we'll have exactly 2 groups of capabilities:
- Routine ones, which are generated automagically when saying "I want 
something working"
- Ones needed to "show off" specific new capabilities

The platforms, which are likely reusing code from outside WebRTC, may 
support many capabilities that are never required for any scenario 
(especially those that have been added over the years to support 
backwards compatibility in very baroque scenarios). Exposing those will 
just add clutter, not add quality.

The "show-offs" will be capable of probing.
>
> 2. Using SDP offer-answer has the advantage of reusing all the IETF 
> standards work that occurs to define SDP when a new codec comes along. 
> But it also has the *disadvantage* of having to wait for the IETF to 
> produce these standards, which may be incomplete, or unable to control 
> parameters which are necessary for web applications.
Hmmm.... since when have people *actually* waited for the IETF process 
to finish before starting to use newly defined capabilities?
This may argue for looking again at the IANA procedures for SDP, and 
institute something like the "provisional registries" that were 
introduced for email/web.
>
> 3. Anything that is missing in SDP (example: forcing the Opus codec to 
> "music" mode for encoding) will still need to be exposed as a 
> Javascript API on the encoder. Thus we end up with a mix of 
> possibly-conflicting settings that are adjusted via the explicit API 
> and the opaque SDP API.
Won't these be things people want in SDP too? See above.
>
> 4. Other work in browsers (like recording camera and microphone to 
> disk) will require the ability to directly control the encoders. If 
> these APIs exist then there will definitely be conflicting settings. 
> What happens if you feed in an SDP answer that requires VP8 encoding 
> and then set the encoder to H.264 mode via the Javascript?
Transcoding? :-)
The same problem exists today when trying to use cameras for multiple 
things concurrently.
>
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From pkyzivat@alum.mit.edu  Mon Aug  8 12:04:34 2011
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Date: Mon, 08 Aug 2011 15:04:58 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] codec and connection negotiation
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I also have some observations, without conclusions, on this decision:

SDP o/a has some well known limitations. It would be nice if rtcweb was 
not saddled with them. For instance, the offer can list multiple codecs, 
but the answerer is free to accept more than one. In that case any of 
those in both offer and answer can be used. But there are UAs that 
support multiple codecs, but only one at a time. O/A doesn't support 
that well.

Unfortunately, relaxing the limitations may break interoperation with 
non-rtcweb devices. For instance if you have an rtcweb client initiating 
a call to a pstn destination, the gateway will likely have the limitation.

Querying for capabilities before initiating a call could, in *some* 
cases transcend that incompatibility. But not always. In general it 
isn't easy to guarantee that the UA you reach with a query will be the 
same one you reach when you try to establish a call. If not, the 
capabilities may be wrong.

SDP has needed modernizing for ages. But SDPng went down in flames. 
Attempting to do something like that again would probably not be a good 
plan for rtcweb.

It might be easier to go with SDP and o/a in order to ease interop, and 
at the same time provide a capability query mechanism, also built on SDP 
that could be used optionally, with the understanding that it might not 
be available in all cases.

	Thanks,
	paul


On 8/8/11 9:09 AM, Harald Alvestrand wrote:
> Trying to doodle on the opinions (the alternative lists were quite
> useful to clear the mind - I liked them a lot, and will return to them!)
>
> On 08/05/11 20:53, Matthew Kaufman wrote:
>> Now, to follow up on my previous message with a bit of opinion...
>>
>> I think we should not use SDP as the API, and we should not have the
>> browser implementing SDP offer-answer.
>>
>> 1. If SDP offer-answer is all that is available to the Javascript
>> programmer (and server programmer), it becomes very difficult to know
>> what the full capability set is without complex probing. This
>> significantly complicates the building of future innovative
>> applications on top of the browser as platform. Many of the current
>> applications that show off browser capabilities do so by using
>> capabilities that were already present for other reasons, and we can
>> expect the same innovation to occur here, if we provide the tools.
> This may be heretical ... but do we actually need to enumerate all
> capabilities?
> It seems to me that we'll have exactly 2 groups of capabilities:
> - Routine ones, which are generated automagically when saying "I want
> something working"
> - Ones needed to "show off" specific new capabilities
>
> The platforms, which are likely reusing code from outside WebRTC, may
> support many capabilities that are never required for any scenario
> (especially those that have been added over the years to support
> backwards compatibility in very baroque scenarios). Exposing those will
> just add clutter, not add quality.
>
> The "show-offs" will be capable of probing.
>>
>> 2. Using SDP offer-answer has the advantage of reusing all the IETF
>> standards work that occurs to define SDP when a new codec comes along.
>> But it also has the *disadvantage* of having to wait for the IETF to
>> produce these standards, which may be incomplete, or unable to control
>> parameters which are necessary for web applications.
> Hmmm.... since when have people *actually* waited for the IETF process
> to finish before starting to use newly defined capabilities?
> This may argue for looking again at the IANA procedures for SDP, and
> institute something like the "provisional registries" that were
> introduced for email/web.
>>
>> 3. Anything that is missing in SDP (example: forcing the Opus codec to
>> "music" mode for encoding) will still need to be exposed as a
>> Javascript API on the encoder. Thus we end up with a mix of
>> possibly-conflicting settings that are adjusted via the explicit API
>> and the opaque SDP API.
> Won't these be things people want in SDP too? See above.
>>
>> 4. Other work in browsers (like recording camera and microphone to
>> disk) will require the ability to directly control the encoders. If
>> these APIs exist then there will definitely be conflicting settings.
>> What happens if you feed in an SDP answer that requires VP8 encoding
>> and then set the encoder to H.264 mode via the Javascript?
> Transcoding? :-)
> The same problem exists today when trying to use cameras for multiple
> things concurrently.
>>
>>
>> Matthew Kaufman
>> _______________________________________________
>> 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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I think we have to distinguish between the use of SDP in the API versus use=
 in signaling. =20

When you say "attempting to do something like that again"=2C I presume that=
 you are referring to an attempt to develop a replacement for SDP that woul=
d be used on the wire between an RTCWEB implementation and a legacy impleme=
ntation.   Given that legacy implementations won't support a replacement=2C=
 it's hard to argue against use of SDP in that context.=20

However=2C that doesn't necessarily argue for use of SDP (or something sema=
ntically equivalent) in the API.   The danger we face in adopting SDP withi=
n the API is we will be saddled with the limitations of SDP while not makin=
g use of all of its capabilities.  We saw hints of this already in the IETF=
 81 discussion -- where questions arose as to whether the JSON SDP blob was=
 opaque or editable=2C how faithful the API is to existing SDP usage (inclu=
ding offer/answer=2C ICE)=2C etc.   As a data point=2C I'd note that few if=
 any of the existing Web-based realtime APIs have chosen to utilize the SDP=
 format=2C and that attempts to "profile" SDP usage to avoid corner cases h=
ave encountered issues=2C particularly when dealing with PSTN interop. =20


> Date: Mon=2C 8 Aug 2011 15:04:58 -0400
> From: pkyzivat@alum.mit.edu
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] codec and connection negotiation
>=20
> I also have some observations=2C without conclusions=2C on this decision:
>=20
> SDP o/a has some well known limitations. It would be nice if rtcweb was=20
> not saddled with them. For instance=2C the offer can list multiple codecs=
=2C=20
> but the answerer is free to accept more than one. In that case any of=20
> those in both offer and answer can be used. But there are UAs that=20
> support multiple codecs=2C but only one at a time. O/A doesn't support=20
> that well.
>=20
> Unfortunately=2C relaxing the limitations may break interoperation with=20
> non-rtcweb devices. For instance if you have an rtcweb client initiating=
=20
> a call to a pstn destination=2C the gateway will likely have the limitati=
on.
>=20
> Querying for capabilities before initiating a call could=2C in *some*=20
> cases transcend that incompatibility. But not always. In general it=20
> isn't easy to guarantee that the UA you reach with a query will be the=20
> same one you reach when you try to establish a call. If not=2C the=20
> capabilities may be wrong.
>=20
> SDP has needed modernizing for ages. But SDPng went down in flames.=20
> Attempting to do something like that again would probably not be a good=20
> plan for rtcweb.
>=20
> It might be easier to go with SDP and o/a in order to ease interop=2C and=
=20
> at the same time provide a capability query mechanism=2C also built on SD=
P=20
> that could be used optionally=2C with the understanding that it might not=
=20
> be available in all cases.
>=20
> 	Thanks=2C
> 	paul
>=20
>=20
> On 8/8/11 9:09 AM=2C Harald Alvestrand wrote:
> > Trying to doodle on the opinions (the alternative lists were quite
> > useful to clear the mind - I liked them a lot=2C and will return to the=
m!)
> >
> > On 08/05/11 20:53=2C Matthew Kaufman wrote:
> >> Now=2C to follow up on my previous message with a bit of opinion...
> >>
> >> I think we should not use SDP as the API=2C and we should not have the
> >> browser implementing SDP offer-answer.
> >>
> >> 1. If SDP offer-answer is all that is available to the Javascript
> >> programmer (and server programmer)=2C it becomes very difficult to kno=
w
> >> what the full capability set is without complex probing. This
> >> significantly complicates the building of future innovative
> >> applications on top of the browser as platform. Many of the current
> >> applications that show off browser capabilities do so by using
> >> capabilities that were already present for other reasons=2C and we can
> >> expect the same innovation to occur here=2C if we provide the tools.
> > This may be heretical ... but do we actually need to enumerate all
> > capabilities?
> > It seems to me that we'll have exactly 2 groups of capabilities:
> > - Routine ones=2C which are generated automagically when saying "I want
> > something working"
> > - Ones needed to "show off" specific new capabilities
> >
> > The platforms=2C which are likely reusing code from outside WebRTC=2C m=
ay
> > support many capabilities that are never required for any scenario
> > (especially those that have been added over the years to support
> > backwards compatibility in very baroque scenarios). Exposing those will
> > just add clutter=2C not add quality.
> >
> > The "show-offs" will be capable of probing.
> >>
> >> 2. Using SDP offer-answer has the advantage of reusing all the IETF
> >> standards work that occurs to define SDP when a new codec comes along.
> >> But it also has the *disadvantage* of having to wait for the IETF to
> >> produce these standards=2C which may be incomplete=2C or unable to con=
trol
> >> parameters which are necessary for web applications.
> > Hmmm.... since when have people *actually* waited for the IETF process
> > to finish before starting to use newly defined capabilities?
> > This may argue for looking again at the IANA procedures for SDP=2C and
> > institute something like the "provisional registries" that were
> > introduced for email/web.
> >>
> >> 3. Anything that is missing in SDP (example: forcing the Opus codec to
> >> "music" mode for encoding) will still need to be exposed as a
> >> Javascript API on the encoder. Thus we end up with a mix of
> >> possibly-conflicting settings that are adjusted via the explicit API
> >> and the opaque SDP API.
> > Won't these be things people want in SDP too? See above.
> >>
> >> 4. Other work in browsers (like recording camera and microphone to
> >> disk) will require the ability to directly control the encoders. If
> >> these APIs exist then there will definitely be conflicting settings.
> >> What happens if you feed in an SDP answer that requires VP8 encoding
> >> and then set the encoder to H.264 mode via the Javascript?
> > Transcoding? :-)
> > The same problem exists today when trying to use cameras for multiple
> > things concurrently.
> >>
> >>
> >> Matthew Kaufman
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
I think we have to distinguish between the use of SDP in the API versus use=
 in signaling.&nbsp=3B <br><br>When you say "attempting to do something lik=
e that again"=2C I presume that you are referring to an attempt to develop =
a replacement for SDP that would be used on the wire between an RTCWEB impl=
ementation and a legacy implementation.&nbsp=3B&nbsp=3B Given that legacy i=
mplementations won't support a replacement=2C it's hard to argue against us=
e of SDP in that context. <br><br>However=2C that doesn't necessarily argue=
 for use of SDP (or something semantically equivalent) in the API.&nbsp=3B&=
nbsp=3B The danger we face in adopting SDP within the API is we will be sad=
dled with the limitations of SDP while not making use of all of its capabil=
ities.&nbsp=3B We saw hints of this already in the IETF 81 discussion -- wh=
ere questions arose as to whether the JSON SDP blob was opaque or editable=
=2C how faithful the API is to existing SDP usage (including offer/answer=
=2C ICE)=2C etc. &nbsp=3B As a data point=2C I'd note that few if any of th=
e existing Web-based realtime APIs have chosen to utilize the SDP format=2C=
 and that attempts to "profile" SDP usage to avoid corner cases have encoun=
tered issues=2C particularly when dealing with PSTN interop.&nbsp=3B <br><b=
r><br><div>&gt=3B Date: Mon=2C 8 Aug 2011 15:04:58 -0400<br>&gt=3B From: pk=
yzivat@alum.mit.edu<br>&gt=3B To: rtcweb@ietf.org<br>&gt=3B Subject: Re: [r=
tcweb] codec and connection negotiation<br>&gt=3B <br>&gt=3B I also have so=
me observations=2C without conclusions=2C on this decision:<br>&gt=3B <br>&=
gt=3B SDP o/a has some well known limitations. It would be nice if rtcweb w=
as <br>&gt=3B not saddled with them. For instance=2C the offer can list mul=
tiple codecs=2C <br>&gt=3B but the answerer is free to accept more than one=
. In that case any of <br>&gt=3B those in both offer and answer can be used=
. But there are UAs that <br>&gt=3B support multiple codecs=2C but only one=
 at a time. O/A doesn't support <br>&gt=3B that well.<br>&gt=3B <br>&gt=3B =
Unfortunately=2C relaxing the limitations may break interoperation with <br=
>&gt=3B non-rtcweb devices. For instance if you have an rtcweb client initi=
ating <br>&gt=3B a call to a pstn destination=2C the gateway will likely ha=
ve the limitation.<br>&gt=3B <br>&gt=3B Querying for capabilities before in=
itiating a call could=2C in *some* <br>&gt=3B cases transcend that incompat=
ibility. But not always. In general it <br>&gt=3B isn't easy to guarantee t=
hat the UA you reach with a query will be the <br>&gt=3B same one you reach=
 when you try to establish a call. If not=2C the <br>&gt=3B capabilities ma=
y be wrong.<br>&gt=3B <br>&gt=3B SDP has needed modernizing for ages. But S=
DPng went down in flames. <br>&gt=3B Attempting to do something like that a=
gain would probably not be a good <br>&gt=3B plan for rtcweb.<br>&gt=3B <br=
>&gt=3B It might be easier to go with SDP and o/a in order to ease interop=
=2C and <br>&gt=3B at the same time provide a capability query mechanism=2C=
 also built on SDP <br>&gt=3B that could be used optionally=2C with the und=
erstanding that it might not <br>&gt=3B be available in all cases.<br>&gt=
=3B <br>&gt=3B 	Thanks=2C<br>&gt=3B 	paul<br>&gt=3B <br>&gt=3B <br>&gt=3B O=
n 8/8/11 9:09 AM=2C Harald Alvestrand wrote:<br>&gt=3B &gt=3B Trying to doo=
dle on the opinions (the alternative lists were quite<br>&gt=3B &gt=3B usef=
ul to clear the mind - I liked them a lot=2C and will return to them!)<br>&=
gt=3B &gt=3B<br>&gt=3B &gt=3B On 08/05/11 20:53=2C Matthew Kaufman wrote:<b=
r>&gt=3B &gt=3B&gt=3B Now=2C to follow up on my previous message with a bit=
 of opinion...<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B I think we sho=
uld not use SDP as the API=2C and we should not have the<br>&gt=3B &gt=3B&g=
t=3B browser implementing SDP offer-answer.<br>&gt=3B &gt=3B&gt=3B<br>&gt=
=3B &gt=3B&gt=3B 1. If SDP offer-answer is all that is available to the Jav=
ascript<br>&gt=3B &gt=3B&gt=3B programmer (and server programmer)=2C it bec=
omes very difficult to know<br>&gt=3B &gt=3B&gt=3B what the full capability=
 set is without complex probing. This<br>&gt=3B &gt=3B&gt=3B significantly =
complicates the building of future innovative<br>&gt=3B &gt=3B&gt=3B applic=
ations on top of the browser as platform. Many of the current<br>&gt=3B &gt=
=3B&gt=3B applications that show off browser capabilities do so by using<br=
>&gt=3B &gt=3B&gt=3B capabilities that were already present for other reaso=
ns=2C and we can<br>&gt=3B &gt=3B&gt=3B expect the same innovation to occur=
 here=2C if we provide the tools.<br>&gt=3B &gt=3B This may be heretical ..=
. but do we actually need to enumerate all<br>&gt=3B &gt=3B capabilities?<b=
r>&gt=3B &gt=3B It seems to me that we'll have exactly 2 groups of capabili=
ties:<br>&gt=3B &gt=3B - Routine ones=2C which are generated automagically =
when saying "I want<br>&gt=3B &gt=3B something working"<br>&gt=3B &gt=3B - =
Ones needed to "show off" specific new capabilities<br>&gt=3B &gt=3B<br>&gt=
=3B &gt=3B The platforms=2C which are likely reusing code from outside WebR=
TC=2C may<br>&gt=3B &gt=3B support many capabilities that are never require=
d for any scenario<br>&gt=3B &gt=3B (especially those that have been added =
over the years to support<br>&gt=3B &gt=3B backwards compatibility in very =
baroque scenarios). Exposing those will<br>&gt=3B &gt=3B just add clutter=
=2C not add quality.<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B The "show-offs" will=
 be capable of probing.<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B 2. Us=
ing SDP offer-answer has the advantage of reusing all the IETF<br>&gt=3B &g=
t=3B&gt=3B standards work that occurs to define SDP when a new codec comes =
along.<br>&gt=3B &gt=3B&gt=3B But it also has the *disadvantage* of having =
to wait for the IETF to<br>&gt=3B &gt=3B&gt=3B produce these standards=2C w=
hich may be incomplete=2C or unable to control<br>&gt=3B &gt=3B&gt=3B param=
eters which are necessary for web applications.<br>&gt=3B &gt=3B Hmmm.... s=
ince when have people *actually* waited for the IETF process<br>&gt=3B &gt=
=3B to finish before starting to use newly defined capabilities?<br>&gt=3B =
&gt=3B This may argue for looking again at the IANA procedures for SDP=2C a=
nd<br>&gt=3B &gt=3B institute something like the "provisional registries" t=
hat were<br>&gt=3B &gt=3B introduced for email/web.<br>&gt=3B &gt=3B&gt=3B<=
br>&gt=3B &gt=3B&gt=3B 3. Anything that is missing in SDP (example: forcing=
 the Opus codec to<br>&gt=3B &gt=3B&gt=3B "music" mode for encoding) will s=
till need to be exposed as a<br>&gt=3B &gt=3B&gt=3B Javascript API on the e=
ncoder. Thus we end up with a mix of<br>&gt=3B &gt=3B&gt=3B possibly-confli=
cting settings that are adjusted via the explicit API<br>&gt=3B &gt=3B&gt=
=3B and the opaque SDP API.<br>&gt=3B &gt=3B Won't these be things people w=
ant in SDP too? See above.<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B 4.=
 Other work in browsers (like recording camera and microphone to<br>&gt=3B =
&gt=3B&gt=3B disk) will require the ability to directly control the encoder=
s. If<br>&gt=3B &gt=3B&gt=3B these APIs exist then there will definitely be=
 conflicting settings.<br>&gt=3B &gt=3B&gt=3B What happens if you feed in a=
n SDP answer that requires VP8 encoding<br>&gt=3B &gt=3B&gt=3B and then set=
 the encoder to H.264 mode via the Javascript?<br>&gt=3B &gt=3B Transcoding=
? :-)<br>&gt=3B &gt=3B The same problem exists today when trying to use cam=
eras for multiple<br>&gt=3B &gt=3B things concurrently.<br>&gt=3B &gt=3B&gt=
=3B<br>&gt=3B &gt=3B&gt=3B<br>&gt=3B &gt=3B&gt=3B Matthew Kaufman<br>&gt=3B=
 &gt=3B&gt=3B _______________________________________________<br>&gt=3B &gt=
=3B&gt=3B rtcweb mailing list<br>&gt=3B &gt=3B&gt=3B rtcweb@ietf.org<br>&gt=
=3B &gt=3B&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br>&gt=3B &gt=
=3B&gt=3B<br>&gt=3B &gt=3B<br>&gt=3B &gt=3B _______________________________=
________________<br>&gt=3B &gt=3B rtcweb mailing list<br>&gt=3B &gt=3B rtcw=
eb@ietf.org<br>&gt=3B &gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<b=
r>&gt=3B &gt=3B<br>&gt=3B <br>&gt=3B ______________________________________=
_________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb@ietf.org<br>&gt=3B=
 https://www.ietf.org/mailman/listinfo/rtcweb<br></div> 		 	   		  </div></=
body>
</html>=

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From pkyzivat@alum.mit.edu  Tue Aug  9 09:41:21 2011
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On 8/9/11 12:19 PM, Bernard Aboba wrote:
> I think we have to distinguish between the use of SDP in the API versus
> use in signaling.
>
> When you say "attempting to do something like that again", I presume
> that you are referring to an attempt to develop a replacement for SDP
> that would be used on the wire between an RTCWEB implementation and a
> legacy implementation. Given that legacy implementations won't support a
> replacement, it's hard to argue against use of SDP in that context.

Yes, I was talking about attempting to revive SDPng or something comparable.

> However, that doesn't necessarily argue for use of SDP (or something
> semantically equivalent) in the API. The danger we face in adopting SDP
> within the API is we will be saddled with the limitations of SDP while
> not making use of all of its capabilities. We saw hints of this already
> in the IETF 81 discussion -- where questions arose as to whether the
> JSON SDP blob was opaque or editable, how faithful the API is to
> existing SDP usage (including offer/answer, ICE), etc. As a data point,
> I'd note that few if any of the existing Web-based realtime APIs have
> chosen to utilize the SDP format, and that attempts to "profile" SDP
> usage to avoid corner cases have encountered issues, particularly when
> dealing with PSTN interop.

Certainly it would be possible to provide an API that had something 
semantically equivalent to SDP while using a different syntax. But thata 
comes at the price of having to then track future extensions to SDP.

An alternative is to profile/subset SDP and then perhaps develop a new 
representation semantically equivalent to that. The trouble with that is 
interoperation with things that don't fit the subset. Going that way 
will probably require making it pretty easy to extend the subset when 
things that had been excluded are found to be needed.

(It gets worse if you decide to "add a few things" to the API that 
aren't part of standard SDP.)

	Thanks,
	Paul

>  > Date: Mon, 8 Aug 2011 15:04:58 -0400
>  > From: pkyzivat@alum.mit.edu
>  > To: rtcweb@ietf.org
>  > Subject: Re: [rtcweb] codec and connection negotiation
>  >
>  > I also have some observations, without conclusions, on this decision:
>  >
>  > SDP o/a has some well known limitations. It would be nice if rtcweb was
>  > not saddled with them. For instance, the offer can list multiple codecs,
>  > but the answerer is free to accept more than one. In that case any of
>  > those in both offer and answer can be used. But there are UAs that
>  > support multiple codecs, but only one at a time. O/A doesn't support
>  > that well.
>  >
>  > Unfortunately, relaxing the limitations may break interoperation with
>  > non-rtcweb devices. For instance if you have an rtcweb client initiating
>  > a call to a pstn destination, the gateway will likely have the
> limitation.
>  >
>  > Querying for capabilities before initiating a call could, in *some*
>  > cases transcend that incompatibility. But not always. In general it
>  > isn't easy to guarantee that the UA you reach with a query will be the
>  > same one you reach when you try to establish a call. If not, the
>  > capabilities may be wrong.
>  >
>  > SDP has needed modernizing for ages. But SDPng went down in flames.
>  > Attempting to do something like that again would probably not be a good
>  > plan for rtcweb.
>  >
>  > It might be easier to go with SDP and o/a in order to ease interop, and
>  > at the same time provide a capability query mechanism, also built on SDP
>  > that could be used optionally, with the understanding that it might not
>  > be available in all cases.
>  >
>  > Thanks,
>  > paul
>  >
>  >
>  > On 8/8/11 9:09 AM, Harald Alvestrand wrote:
>  > > Trying to doodle on the opinions (the alternative lists were quite
>  > > useful to clear the mind - I liked them a lot, and will return to
> them!)
>  > >
>  > > On 08/05/11 20:53, Matthew Kaufman wrote:
>  > >> Now, to follow up on my previous message with a bit of opinion...
>  > >>
>  > >> I think we should not use SDP as the API, and we should not have the
>  > >> browser implementing SDP offer-answer.
>  > >>
>  > >> 1. If SDP offer-answer is all that is available to the Javascript
>  > >> programmer (and server programmer), it becomes very difficult to know
>  > >> what the full capability set is without complex probing. This
>  > >> significantly complicates the building of future innovative
>  > >> applications on top of the browser as platform. Many of the current
>  > >> applications that show off browser capabilities do so by using
>  > >> capabilities that were already present for other reasons, and we can
>  > >> expect the same innovation to occur here, if we provide the tools.
>  > > This may be heretical ... but do we actually need to enumerate all
>  > > capabilities?
>  > > It seems to me that we'll have exactly 2 groups of capabilities:
>  > > - Routine ones, which are generated automagically when saying "I want
>  > > something working"
>  > > - Ones needed to "show off" specific new capabilities
>  > >
>  > > The platforms, which are likely reusing code from outside WebRTC, may
>  > > support many capabilities that are never required for any scenario
>  > > (especially those that have been added over the years to support
>  > > backwards compatibility in very baroque scenarios). Exposing those will
>  > > just add clutter, not add quality.
>  > >
>  > > The "show-offs" will be capable of probing.
>  > >>
>  > >> 2. Using SDP offer-answer has the advantage of reusing all the IETF
>  > >> standards work that occurs to define SDP when a new codec comes along.
>  > >> But it also has the *disadvantage* of having to wait for the IETF to
>  > >> produce these standards, which may be incomplete, or unable to control
>  > >> parameters which are necessary for web applications.
>  > > Hmmm.... since when have people *actually* waited for the IETF process
>  > > to finish before starting to use newly defined capabilities?
>  > > This may argue for looking again at the IANA procedures for SDP, and
>  > > institute something like the "provisional registries" that were
>  > > introduced for email/web.
>  > >>
>  > >> 3. Anything that is missing in SDP (example: forcing the Opus codec to
>  > >> "music" mode for encoding) will still need to be exposed as a
>  > >> Javascript API on the encoder. Thus we end up with a mix of
>  > >> possibly-conflicting settings that are adjusted via the explicit API
>  > >> and the opaque SDP API.
>  > > Won't these be things people want in SDP too? See above.
>  > >>
>  > >> 4. Other work in browsers (like recording camera and microphone to
>  > >> disk) will require the ability to directly control the encoders. If
>  > >> these APIs exist then there will definitely be conflicting settings.
>  > >> What happens if you feed in an SDP answer that requires VP8 encoding
>  > >> and then set the encoder to H.264 mode via the Javascript?
>  > > Transcoding? :-)
>  > > The same problem exists today when trying to use cameras for multiple
>  > > things concurrently.
>  > >>
>  > >>
>  > >> Matthew Kaufman
>  > >> _______________________________________________
>  > >> 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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Paul Kyzivat said:=20
> Certainly it would be possible to provide an API that had something=20
> semantically equivalent to SDP while using a different syntax. But thata=
=20
> comes at the price of having to then track future extensions to SDP.
>=20
> An alternative is to profile/subset SDP and then perhaps develop a new=20
> representation semantically equivalent to that. The trouble with that is=
=20
> interoperation with things that don't fit the subset. Going that way=20
> will probably require making it pretty easy to extend the subset when=20
> things that had been excluded are found to be needed.
>=20
> (It gets worse if you decide to "add a few things" to the API that=20
> aren't part of standard SDP.)

[BA]  Given the discussion of IETF 81=2C my concern is that the API is base=
d on  an SDP and ICE profile that is implicit and not explicit.    In a num=
ber of cases=2C it would appear that this implicit profile may ignore norma=
tive requirements (including MUSTs) in RFC 3264=2C 5245 and other relevant =
RFCs. =20
The danger in continuing down that road is  that different implementations =
will make different assumptions about the profile=2C and the result will be=
 implementations that are not only incompatible with each other=2C but also=
 with existing implementations of the wire protocols.    		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
<div>Paul Kyzivat said:&nbsp=3B</div><div><br></div><div>&gt=3B Certainly i=
t would be possible to provide an API that had something <br>&gt=3B semanti=
cally equivalent to SDP while using a different syntax. But thata <br>&gt=
=3B comes at the price of having to then track future extensions to SDP.<br=
>&gt=3B <br>&gt=3B An alternative is to profile/subset SDP and then perhaps=
 develop a new <br>&gt=3B representation semantically equivalent to that. T=
he trouble with that is <br>&gt=3B interoperation with things that don't fi=
t the subset. Going that way <br>&gt=3B will probably require making it pre=
tty easy to extend the subset when <br>&gt=3B things that had been excluded=
 are found to be needed.<br>&gt=3B <br>&gt=3B (It gets worse if you decide =
to "add a few things" to the API that <br>&gt=3B aren't part of standard SD=
P.)<br><br></div><div>[BA] &nbsp=3BGiven the discussion of IETF 81=2C my co=
ncern is that the API is based on &nbsp=3Ban SDP and ICE profile that is im=
plicit and not explicit. &nbsp=3B &nbsp=3BIn a number of cases=2C it would =
appear that this implicit profile may ignore normative requirements (includ=
ing MUSTs) in RFC 3264=2C 5245 and other relevant RFCs. &nbsp=3B</div><div>=
<br></div><div>The danger in continuing down that road is &nbsp=3Bthat diff=
erent implementations will make different assumptions about the profile=2C =
and the result will be implementations that are not only incompatible with =
each other=2C but also with existing implementations of the wire protocols.=
 &nbsp=3B&nbsp=3B</div> 		 	   		  </div></body>
</html>=

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Leaving all the problems behind mentioned here for SDP (in full agreement),
the bigger picture is that apps may have so many other types of data to
exchange, besides codecs.
For example numbers, sizes and types of screens, keyboard/touch devices to
optimize user experience, data for mashup parts of the screen, say surround
sound(?), etc.
Folks not wedded to SDP use metadata to exchange app info. Why should RTC
continue to inhabit a world of its own, even in the browser?

Na=EFve question:
Does this separate use of SDP to convey RTC metadata not contradict the
target universality of apps in the browser?

Thanks, Henry


On 8/9/11 12:18 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:

> Paul Kyzivat said:
>=20
>> > Certainly it would be possible to provide an API that had something
>> > semantically equivalent to SDP while using a different syntax. But tha=
ta
>> > comes at the price of having to then track future extensions to SDP.
>> >=20
>> > An alternative is to profile/subset SDP and then perhaps develop a new
>> > representation semantically equivalent to that. The trouble with that =
is
>> > interoperation with things that don't fit the subset. Going that way
>> > will probably require making it pretty easy to extend the subset when
>> > things that had been excluded are found to be needed.
>> >=20
>> > (It gets worse if you decide to "add a few things" to the API that
>> > aren't part of standard SDP.)
>=20
> [BA]  Given the discussion of IETF 81, my concern is that the API is base=
d on
> an SDP and ICE profile that is implicit and not explicit.    In a number =
of
> cases, it would appear that this implicit profile may ignore normative
> requirements (including MUSTs) in RFC 3264, 5245 and other relevant RFCs.
>=20
> The danger in continuing down that road is  that different implementation=
s
> will make different assumptions about the profile, and the result will be
> implementations that are not only incompatible with each other, but also =
with
> existing implementations of the wire protocols.
>       =20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--B_3395764779_2078863
Content-type: text/html;
	charset="ISO-8859-1"
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<HTML>
<HEAD>
<TITLE>Re: [rtcweb] codec and connection negotiation</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>Leaving all the problems behind mentioned here for SDP (in full agreement)=
, the bigger picture is that apps may have so many other types of data to ex=
change, besides codecs.<BR>
For example numbers, sizes and types of screens, keyboard/touch devices to =
optimize user experience, data for mashup parts of the screen, say surround =
sound(?), etc.<BR>
Folks not wedded to SDP use metadata to exchange app info. Why should RTC c=
ontinue to inhabit a world of its own, even in the browser?<BR>
<BR>
Na&iuml;ve question:<BR>
Does this separate use of SDP to convey RTC metadata not contradict the tar=
get universality of apps in the browser?<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 8/9/11 12:18 PM, &quot;Bernard Aboba&quot; &lt;<a href=3D"bernard_aboba@ho=
tmail.com">bernard_aboba@hotmail.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT SIZE=3D"1"><FONT FACE=3D"Tahoma, Verdana, Helve=
tica, Arial"><SPAN STYLE=3D'font-size:10pt'>Paul Kyzivat said: <BR>
<BR>
&gt; Certainly it would be possible to provide an API that had something <B=
R>
&gt; semantically equivalent to SDP while using a different syntax. But tha=
ta <BR>
&gt; comes at the price of having to then track future extensions to SDP.<B=
R>
&gt; <BR>
&gt; An alternative is to profile/subset SDP and then perhaps develop a new=
 <BR>
&gt; representation semantically equivalent to that. The trouble with that =
is <BR>
&gt; interoperation with things that don't fit the subset. Going that way <=
BR>
&gt; will probably require making it pretty easy to extend the subset when =
<BR>
&gt; things that had been excluded are found to be needed.<BR>
&gt; <BR>
&gt; (It gets worse if you decide to &quot;add a few things&quot; to the AP=
I that <BR>
&gt; aren't part of standard SDP.)<BR>
<BR>
[BA] &nbsp;Given the discussion of IETF 81, my concern is that the API is b=
ased on &nbsp;an SDP and ICE profile that is implicit and not explicit. &nbs=
p;&nbsp;&nbsp;In a number of cases, it would appear that this implicit profi=
le may ignore normative requirements (including MUSTs) in RFC 3264, 5245 and=
 other relevant RFCs. &nbsp;<BR>
<BR>
The danger in continuing down that road is &nbsp;that different implementat=
ions will make different assumptions about the profile, and the result will =
be implementations that are not only incompatible with each other, but also =
with existing implementations of the wire protocols. &nbsp;&nbsp;<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>
<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"95%"></SPAN></FONT></FONT><FONT FACE=3D"Cons=
olas, Courier New, Courier"><SPAN STYLE=3D'font-size:13pt'>___________________=
____________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
</SPAN></FONT></BLOCKQUOTE>
</BODY>
</HTML>


--B_3395764779_2078863--



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On 08/10/2011 02:59 AM, Henry Sinnreich wrote:
> Leaving all the problems behind mentioned here for SDP (in full 
> agreement), the bigger picture is that apps may have so many other 
> types of data to exchange, besides codecs.
> For example numbers, sizes and types of screens, keyboard/touch 
> devices to optimize user experience, data for mashup parts of the 
> screen, say surround sound(?), etc.
> Folks not wedded to SDP use metadata to exchange app info. Why should 
> RTC continue to inhabit a world of its own, even in the browser?

> Naïve question:
> Does this separate use of SDP to convey RTC metadata not contradict 
> the target universality of apps in the browser?

Sorry, I can't parse that sentence.

On one possible parsing, my response would be "no more than the use of 
UTF-8 to convey human names constrain the naming conventions of humanity".
(Prince, in his "symbol" phase, might not like it, though. No way to 
please them all.)

Seriously, if codec/stream-related metadata needs to be conveyed in one 
standard format, it seems that we're either using SDP or reinventing it. 
And so far, the reinventors seem to be unable to gather a consensus to 
go forward.

Other metadata doesn't seem (yet) to require one standard format.


>
> Thanks, Henry
>
>
> On 8/9/11 12:18 PM, "Bernard Aboba" <bernard_aboba@hotmail.com> wrote:
>
>     Paul Kyzivat said:
>
>     > Certainly it would be possible to provide an API that had something
>     > semantically equivalent to SDP while using a different syntax.
>     But thata
>     > comes at the price of having to then track future extensions to SDP.
>     >
>     > An alternative is to profile/subset SDP and then perhaps develop
>     a new
>     > representation semantically equivalent to that. The trouble with
>     that is
>     > interoperation with things that don't fit the subset. Going that way
>     > will probably require making it pretty easy to extend the subset
>     when
>     > things that had been excluded are found to be needed.
>     >
>     > (It gets worse if you decide to "add a few things" to the API that
>     > aren't part of standard SDP.)
>
>     [BA]  Given the discussion of IETF 81, my concern is that the API
>     is based on  an SDP and ICE profile that is implicit and not
>     explicit.    In a number of cases, it would appear that this
>     implicit profile may ignore normative requirements (including
>     MUSTs) in RFC 3264, 5245 and other relevant RFCs.
>
>     The danger in continuing down that road is  that different
>     implementations will make different assumptions about the profile,
>     and the result will be implementations that are not only
>     incompatible with each other, but also with existing
>     implementations of the wire protocols.
>
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     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 08/10/2011 02:59 AM, Henry Sinnreich wrote:
    <blockquote cite="mid:CA67422A.1CAD4%25henry.sinnreich@gmail.com"
      type="cite">
      <title>Re: [rtcweb] codec and connection negotiation</title>
      <font face="Calibri, Verdana, Helvetica, Arial"><span
          style="font-size: 13pt;">Leaving all the problems behind
          mentioned here for SDP (in full agreement), the bigger picture
          is that apps may have so many other types of data to exchange,
          besides codecs.<br>
          For example numbers, sizes and types of screens,
          keyboard/touch devices to optimize user experience, data for
          mashup parts of the screen, say surround sound(?), etc.<br>
          Folks not wedded to SDP use metadata to exchange app info. Why
          should RTC continue to inhabit a world of its own, even in the
          browser?<br>
        </span></font></blockquote>
    <br>
    <blockquote cite="mid:CA67422A.1CAD4%25henry.sinnreich@gmail.com"
      type="cite"><font face="Calibri, Verdana, Helvetica, Arial"><span
          style="font-size: 13pt;">
          Na&iuml;ve question:<br>
          Does this separate use of SDP to convey RTC metadata not
          contradict the target universality of apps in the browser?<br>
        </span></font></blockquote>
    <br>
    Sorry, I can't parse that sentence.<br>
    <br>
    On one possible parsing, my response would be "no more than the use
    of UTF-8 to convey human names constrain the naming conventions of
    humanity".<br>
    (Prince, in his "symbol" phase, might not like it, though. No way to
    please them all.)<br>
    <br>
    Seriously, if codec/stream-related metadata needs to be conveyed in
    one standard format, it seems that we're either using SDP or
    reinventing it. And so far, the reinventors seem to be unable to
    gather a consensus to go forward.<br>
    <br>
    Other metadata doesn't seem (yet) to require one standard format.<br>
    <br>
    <br>
    <blockquote cite="mid:CA67422A.1CAD4%25henry.sinnreich@gmail.com"
      type="cite"><font face="Calibri, Verdana, Helvetica, Arial"><span
          style="font-size: 13pt;">
          <br>
          Thanks, Henry<br>
          <br>
          <br>
          On 8/9/11 12:18 PM, "Bernard Aboba" &lt;<a
            moz-do-not-send="true" href="bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;
          wrote:<br>
          <br>
        </span></font>
      <blockquote><font size="1"><font face="Tahoma, Verdana, Helvetica,
            Arial"><span style="font-size: 10pt;">Paul Kyzivat said: <br>
              <br>
              &gt; Certainly it would be possible to provide an API that
              had something <br>
              &gt; semantically equivalent to SDP while using a
              different syntax. But thata <br>
              &gt; comes at the price of having to then track future
              extensions to SDP.<br>
              &gt; <br>
              &gt; An alternative is to profile/subset SDP and then
              perhaps develop a new <br>
              &gt; representation semantically equivalent to that. The
              trouble with that is <br>
              &gt; interoperation with things that don't fit the subset.
              Going that way <br>
              &gt; will probably require making it pretty easy to extend
              the subset when <br>
              &gt; things that had been excluded are found to be needed.<br>
              &gt; <br>
              &gt; (It gets worse if you decide to "add a few things" to
              the API that <br>
              &gt; aren't part of standard SDP.)<br>
              <br>
              [BA] &nbsp;Given the discussion of IETF 81, my concern is that
              the API is based on &nbsp;an SDP and ICE profile that is
              implicit and not explicit. &nbsp;&nbsp;&nbsp;In a number of cases, it
              would appear that this implicit profile may ignore
              normative requirements (including MUSTs) in RFC 3264, 5245
              and other relevant RFCs. &nbsp;<br>
              <br>
              The danger in continuing down that road is &nbsp;that different
              implementations will make different assumptions about the
              profile, and the result will be implementations that are
              not only incompatible with each other, but also with
              existing implementations of the wire protocols. &nbsp;&nbsp;<br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<br>
              <br>
              <hr size="3" width="95%" align="CENTER"></span></font></font><font
          face="Consolas, Courier New, Courier"><span style="font-size:
            13pt;">_______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </span></font></blockquote>
      <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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From Markus.Isomaki@nokia.com  Wed Aug 10 00:27:35 2011
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Hi,

Yes, it is important to distinguish between the API and the protocol in thi=
s discussion. Standardizing the Javascript API is necessary, but standardiz=
ing the "on the wire" protocol is not. At least one of the requirements sho=
uld be that we leave enough flexibility about the protocol. A service provi=
der should be able to implement/use any protocol that does the job required=
 and works with the standard API offered by the browsers. SDP offer/answer =
could be one of those protocols.

The API itself is the critical piece. I'm not sure if there is any benefit =
in tying it to SDP, but certainly it should be possible to implement SDP of=
fer/answer protocol in Javascript on top of it. That could be one requireme=
nt for it, and something that could be even proved by a real implementation=
.

So, in summary: Avoid strict dependency to SDP, but it must be possible to =
implement SDP offer/answer protocol on top of the standardized API.

Markus


From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 ext Bernard Aboba
Sent: 09 August, 2011 19:19
To: pkyzivat@alum.mit.edu; rtcweb@ietf.org
Subject: Re: [rtcweb] codec and connection negotiation

I think we have to distinguish between the use of SDP in the API versus use=
 in signaling.

When you say "attempting to do something like that again", I presume that y=
ou are referring to an attempt to develop a replacement for SDP that would =
be used on the wire between an RTCWEB implementation and a legacy implement=
ation.   Given that legacy implementations won't support a replacement, it'=
s hard to argue against use of SDP in that context.

However, that doesn't necessarily argue for use of SDP (or something semant=
ically equivalent) in the API.   The danger we face in adopting SDP within =
the API is we will be saddled with the limitations of SDP while not making =
use of all of its capabilities.  We saw hints of this already in the IETF 8=
1 discussion -- where questions arose as to whether the JSON SDP blob was o=
paque or editable, how faithful the API is to existing SDP usage (including=
 offer/answer, ICE), etc.   As a data point, I'd note that few if any of th=
e existing Web-based realtime APIs have chosen to utilize the SDP format, a=
nd that attempts to "profile" SDP usage to avoid corner cases have encounte=
red issues, particularly when dealing with PSTN interop.

> Date: Mon, 8 Aug 2011 15:04:58 -0400
> From: pkyzivat@alum.mit.edu<mailto:pkyzivat@alum.mit.edu>
> To: rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> Subject: Re: [rtcweb] codec and connection negotiation
>
> I also have some observations, without conclusions, on this decision:
>
> SDP o/a has some well known limitations. It would be nice if rtcweb was
> not saddled with them. For instance, the offer can list multiple codecs,
> but the answerer is free to accept more than one. In that case any of
> those in both offer and answer can be used. But there are UAs that
> support multiple codecs, but only one at a time. O/A doesn't support
> that well.
>
> Unfortunately, relaxing the limitations may break interoperation with
> non-rtcweb devices. For instance if you have an rtcweb client initiating
> a call to a pstn destination, the gateway will likely have the limitation=
.
>
> Querying for capabilities before initiating a call could, in *some*
> cases transcend that incompatibility. But not always. In general it
> isn't easy to guarantee that the UA you reach with a query will be the
> same one you reach when you try to establish a call. If not, the
> capabilities may be wrong.
>
> SDP has needed modernizing for ages. But SDPng went down in flames.
> Attempting to do something like that again would probably not be a good
> plan for rtcweb.
>
> It might be easier to go with SDP and o/a in order to ease interop, and
> at the same time provide a capability query mechanism, also built on SDP
> that could be used optionally, with the understanding that it might not
> be available in all cases.
>
> Thanks,
> paul
>
>
> On 8/8/11 9:09 AM, Harald Alvestrand wrote:
> > Trying to doodle on the opinions (the alternative lists were quite
> > useful to clear the mind - I liked them a lot, and will return to them!=
)
> >
> > On 08/05/11 20:53, Matthew Kaufman wrote:
> >> Now, to follow up on my previous message with a bit of opinion...
> >>
> >> I think we should not use SDP as the API, and we should not have the
> >> browser implementing SDP offer-answer.
> >>
> >> 1. If SDP offer-answer is all that is available to the Javascript
> >> programmer (and server programmer), it becomes very difficult to know
> >> what the full capability set is without complex probing. This
> >> significantly complicates the building of future innovative
> >> applications on top of the browser as platform. Many of the current
> >> applications that show off browser capabilities do so by using
> >> capabilities that were already present for other reasons, and we can
> >> expect the same innovation to occur here, if we provide the tools.
> > This may be heretical ... but do we actually need to enumerate all
> > capabilities?
> > It seems to me that we'll have exactly 2 groups of capabilities:
> > - Routine ones, which are generated automagically when saying "I want
> > something working"
> > - Ones needed to "show off" specific new capabilities
> >
> > The platforms, which are likely reusing code from outside WebRTC, may
> > support many capabilities that are never required for any scenario
> > (especially those that have been added over the years to support
> > backwards compatibility in very baroque scenarios). Exposing those will
> > just add clutter, not add quality.
> >
> > The "show-offs" will be capable of probing.
> >>
> >> 2. Using SDP offer-answer has the advantage of reusing all the IETF
> >> standards work that occurs to define SDP when a new codec comes along.
> >> But it also has the *disadvantage* of having to wait for the IETF to
> >> produce these standards, which may be incomplete, or unable to control
> >> parameters which are necessary for web applications.
> > Hmmm.... since when have people *actually* waited for the IETF process
> > to finish before starting to use newly defined capabilities?
> > This may argue for looking again at the IANA procedures for SDP, and
> > institute something like the "provisional registries" that were
> > introduced for email/web.
> >>
> >> 3. Anything that is missing in SDP (example: forcing the Opus codec to
> >> "music" mode for encoding) will still need to be exposed as a
> >> Javascript API on the encoder. Thus we end up with a mix of
> >> possibly-conflicting settings that are adjusted via the explicit API
> >> and the opaque SDP API.
> > Won't these be things people want in SDP too? See above.
> >>
> >> 4. Other work in browsers (like recording camera and microphone to
> >> disk) will require the ability to directly control the encoders. If
> >> these APIs exist then there will definitely be conflicting settings.
> >> What happens if you feed in an SDP answer that requires VP8 encoding
> >> and then set the encoder to H.264 mode via the Javascript?
> > Transcoding? :-)
> > The same problem exists today when trying to use cameras for multiple
> > things concurrently.
> >>
> >>
> >> Matthew Kaufman
> >> _______________________________________________
> >> rtcweb mailing list
> >> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> >> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

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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">Hi,<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">Yes, it is important to d=
istinguish between the API and the protocol in this discussion. Standardizi=
ng the Javascript API is necessary, but standardizing the
 &#8220;on the wire&#8221; protocol is not. At least one of the requirement=
s should be that we leave enough flexibility about the protocol. A service =
provider should be able to implement/use any protocol that does the job req=
uired and works with the standard API offered
 by the browsers. SDP offer/answer could be one of those protocols.<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">The API itself is the cri=
tical piece. I&#8217;m not sure if there is any benefit in tying it to SDP,=
 but certainly it should be possible to implement SDP offer/answer
 protocol in Javascript on top of it. That could be one requirement for it,=
 and something that could be even proved by a real implementation.<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">So, in summary: Avoid str=
ict dependency to SDP, but it must be possible to implement SDP offer/answe=
r protocol on top of the standardized API.<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">Markus<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"><o:p>&nbsp;</o:p></span><=
/p>
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<div>
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0cm 0cm">
<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>ext Bernard Aboba<br>
<b>Sent:</b> 09 August, 2011 19:19<br>
<b>To:</b> pkyzivat@alum.mit.edu; rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] codec and connection negotiation<o:p></o:p></s=
pan></p>
</div>
</div>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<div>
<p class=3D"MsoNormal" style=3D"margin-bottom:12.0pt"><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">I think w=
e have to distinguish between the use of SDP in the API versus use in signa=
ling.&nbsp;
<br>
<br>
When you say &quot;attempting to do something like that again&quot;, I pres=
ume that you are referring to an attempt to develop a replacement for SDP t=
hat would be used on the wire between an RTCWEB implementation and a legacy=
 implementation.&nbsp;&nbsp; Given that legacy implementations
 won't support a replacement, it's hard to argue against use of SDP in that=
 context.
<br>
<br>
However, that doesn't necessarily argue for use of SDP (or something semant=
ically equivalent) in the API.&nbsp;&nbsp; The danger we face in adopting S=
DP within the API is we will be saddled with the limitations of SDP while n=
ot making use of all of its capabilities.&nbsp;
 We saw hints of this already in the IETF 81 discussion -- where questions =
arose as to whether the JSON SDP blob was opaque or editable, how faithful =
the API is to existing SDP usage (including offer/answer, ICE), etc. &nbsp;=
 As a data point, I'd note that few if
 any of the existing Web-based realtime APIs have chosen to utilize the SDP=
 format, and that attempts to &quot;profile&quot; SDP usage to avoid corner=
 cases have encountered issues, particularly when dealing with PSTN interop=
.&nbsp;
<br>
<br>
<o:p></o:p></span></p>
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:10.0pt;font-family:&quot;Ta=
homa&quot;,&quot;sans-serif&quot;">&gt; Date: Mon, 8 Aug 2011 15:04:58 -040=
0<br>
&gt; From: <a href=3D"mailto:pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</=
a><br>
&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] codec and connection negotiation<br>
&gt; <br>
&gt; I also have some observations, without conclusions, on this decision:<=
br>
&gt; <br>
&gt; SDP o/a has some well known limitations. It would be nice if rtcweb wa=
s <br>
&gt; not saddled with them. For instance, the offer can list multiple codec=
s, <br>
&gt; but the answerer is free to accept more than one. In that case any of =
<br>
&gt; those in both offer and answer can be used. But there are UAs that <br=
>
&gt; support multiple codecs, but only one at a time. O/A doesn't support <=
br>
&gt; that well.<br>
&gt; <br>
&gt; Unfortunately, relaxing the limitations may break interoperation with =
<br>
&gt; non-rtcweb devices. For instance if you have an rtcweb client initiati=
ng <br>
&gt; a call to a pstn destination, the gateway will likely have the limitat=
ion.<br>
&gt; <br>
&gt; Querying for capabilities before initiating a call could, in *some* <b=
r>
&gt; cases transcend that incompatibility. But not always. In general it <b=
r>
&gt; isn't easy to guarantee that the UA you reach with a query will be the=
 <br>
&gt; same one you reach when you try to establish a call. If not, the <br>
&gt; capabilities may be wrong.<br>
&gt; <br>
&gt; SDP has needed modernizing for ages. But SDPng went down in flames. <b=
r>
&gt; Attempting to do something like that again would probably not be a goo=
d <br>
&gt; plan for rtcweb.<br>
&gt; <br>
&gt; It might be easier to go with SDP and o/a in order to ease interop, an=
d <br>
&gt; at the same time provide a capability query mechanism, also built on S=
DP <br>
&gt; that could be used optionally, with the understanding that it might no=
t <br>
&gt; be available in all cases.<br>
&gt; <br>
&gt; Thanks,<br>
&gt; paul<br>
&gt; <br>
&gt; <br>
&gt; On 8/8/11 9:09 AM, Harald Alvestrand wrote:<br>
&gt; &gt; Trying to doodle on the opinions (the alternative lists were quit=
e<br>
&gt; &gt; useful to clear the mind - I liked them a lot, and will return to=
 them!)<br>
&gt; &gt;<br>
&gt; &gt; On 08/05/11 20:53, Matthew Kaufman wrote:<br>
&gt; &gt;&gt; Now, to follow up on my previous message with a bit of opinio=
n...<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; I think we should not use SDP as the API, and we should not h=
ave the<br>
&gt; &gt;&gt; browser implementing SDP offer-answer.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; 1. If SDP offer-answer is all that is available to the Javasc=
ript<br>
&gt; &gt;&gt; programmer (and server programmer), it becomes very difficult=
 to know<br>
&gt; &gt;&gt; what the full capability set is without complex probing. This=
<br>
&gt; &gt;&gt; significantly complicates the building of future innovative<b=
r>
&gt; &gt;&gt; applications on top of the browser as platform. Many of the c=
urrent<br>
&gt; &gt;&gt; applications that show off browser capabilities do so by usin=
g<br>
&gt; &gt;&gt; capabilities that were already present for other reasons, and=
 we can<br>
&gt; &gt;&gt; expect the same innovation to occur here, if we provide the t=
ools.<br>
&gt; &gt; This may be heretical ... but do we actually need to enumerate al=
l<br>
&gt; &gt; capabilities?<br>
&gt; &gt; It seems to me that we'll have exactly 2 groups of capabilities:<=
br>
&gt; &gt; - Routine ones, which are generated automagically when saying &qu=
ot;I want<br>
&gt; &gt; something working&quot;<br>
&gt; &gt; - Ones needed to &quot;show off&quot; specific new capabilities<b=
r>
&gt; &gt;<br>
&gt; &gt; The platforms, which are likely reusing code from outside WebRTC,=
 may<br>
&gt; &gt; support many capabilities that are never required for any scenari=
o<br>
&gt; &gt; (especially those that have been added over the years to support<=
br>
&gt; &gt; backwards compatibility in very baroque scenarios). Exposing thos=
e will<br>
&gt; &gt; just add clutter, not add quality.<br>
&gt; &gt;<br>
&gt; &gt; The &quot;show-offs&quot; will be capable of probing.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; 2. Using SDP offer-answer has the advantage of reusing all th=
e IETF<br>
&gt; &gt;&gt; standards work that occurs to define SDP when a new codec com=
es along.<br>
&gt; &gt;&gt; But it also has the *disadvantage* of having to wait for the =
IETF to<br>
&gt; &gt;&gt; produce these standards, which may be incomplete, or unable t=
o control<br>
&gt; &gt;&gt; parameters which are necessary for web applications.<br>
&gt; &gt; Hmmm.... since when have people *actually* waited for the IETF pr=
ocess<br>
&gt; &gt; to finish before starting to use newly defined capabilities?<br>
&gt; &gt; This may argue for looking again at the IANA procedures for SDP, =
and<br>
&gt; &gt; institute something like the &quot;provisional registries&quot; t=
hat were<br>
&gt; &gt; introduced for email/web.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; 3. Anything that is missing in SDP (example: forcing the Opus=
 codec to<br>
&gt; &gt;&gt; &quot;music&quot; mode for encoding) will still need to be ex=
posed as a<br>
&gt; &gt;&gt; Javascript API on the encoder. Thus we end up with a mix of<b=
r>
&gt; &gt;&gt; possibly-conflicting settings that are adjusted via the expli=
cit API<br>
&gt; &gt;&gt; and the opaque SDP API.<br>
&gt; &gt; Won't these be things people want in SDP too? See above.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; 4. Other work in browsers (like recording camera and micropho=
ne to<br>
&gt; &gt;&gt; disk) will require the ability to directly control the encode=
rs. If<br>
&gt; &gt;&gt; these APIs exist then there will definitely be conflicting se=
ttings.<br>
&gt; &gt;&gt; What happens if you feed in an SDP answer that requires VP8 e=
ncoding<br>
&gt; &gt;&gt; and then set the encoder to H.264 mode via the Javascript?<br=
>
&gt; &gt; Transcoding? :-)<br>
&gt; &gt; The same problem exists today when trying to use cameras for mult=
iple<br>
&gt; &gt; things concurrently.<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt;<br>
&gt; &gt;&gt; Matthew Kaufman<br>
&gt; &gt;&gt; _______________________________________________<br>
&gt; &gt;&gt; rtcweb mailing list<br>
&gt; &gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; &gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">http=
s://www.ietf.org/mailman/listinfo/rtcweb</a><br>
&gt; &gt;&gt;<br>
&gt; &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; &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><o:p></o:p></span></p>
</div>
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On 8/10/11 3:27 AM, Markus.Isomaki@nokia.com wrote:
> Hi,
>
> Yes, it is important to distinguish between the API and the protocol in
> this discussion. Standardizing the Javascript API is necessary, but
> standardizing the “on the wire” protocol is not. At least one of the
> requirements should be that we leave enough flexibility about the
> protocol.  A service provider should be able to implement/use any
> protocol that does the job required and works with the standard API
> offered by the browsers.  SDP offer/answer could be one of those protocols.

SDP is *not* a protocol - it is a data format.
Offer/answer is, more or less, a protocol. It can, in principle, be used 
with other formats.

However, as things have evolved, there is a strong coupling between the 
SDP data format and the o/a protocol. For instance, IIRC, there are 
various extensions to SDP that come with their own rules for how o/a is 
applied to those pieces.

> The API itself is the critical piece. I’m not sure if there is any
> benefit in tying it to SDP,

Well, at the end of the day, an API must convey the critical data. It 
doesn't necessarily need to encode the data the same way it is encoded 
in SDP, and it doesn't need to convey that data for o/a in the same way 
that SIP does, but it does need to be able to represent the full range 
of SDP and the full semantics of o/a. And it needs to be flexible enough 
to do that as SDP evolves.

> but certainly it should be possible to
> implement SDP offer/answer protocol in Javascript on top of it. That
> could be one requirement for it, and something that could be even proved
> by a real implementation.

Certainly a real implementation is a useful existence proof. But it is 
not a proof of completeness. Completeness will be accomplished by 
defining a full mapping between the API and SDP + o/a.

*One* way to accomplish a complete mapping between the API and the SDP 
representation is to pass complete SDP in the API. Other ways are 
possible, but will be more work to define.

	Thanks,
	Paul

> So, in summary: Avoid strict dependency to SDP, but it must be possible
> to implement SDP offer/answer protocol on top of the standardized API.
>
> Markus
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On
> Behalf Of *ext Bernard Aboba
> *Sent:* 09 August, 2011 19:19
> *To:* pkyzivat@alum.mit.edu; rtcweb@ietf.org
> *Subject:* Re: [rtcweb] codec and connection negotiation
>
> I think we have to distinguish between the use of SDP in the API versus
> use in signaling.
>
> When you say "attempting to do something like that again", I presume
> that you are referring to an attempt to develop a replacement for SDP
> that would be used on the wire between an RTCWEB implementation and a
> legacy implementation. Given that legacy implementations won't support a
> replacement, it's hard to argue against use of SDP in that context.
>
> However, that doesn't necessarily argue for use of SDP (or something
> semantically equivalent) in the API. The danger we face in adopting SDP
> within the API is we will be saddled with the limitations of SDP while
> not making use of all of its capabilities. We saw hints of this already
> in the IETF 81 discussion -- where questions arose as to whether the
> JSON SDP blob was opaque or editable, how faithful the API is to
> existing SDP usage (including offer/answer, ICE), etc. As a data point,
> I'd note that few if any of the existing Web-based realtime APIs have
> chosen to utilize the SDP format, and that attempts to "profile" SDP
> usage to avoid corner cases have encountered issues, particularly when
> dealing with PSTN interop.
>
>>  Date: Mon, 8 Aug 2011 15:04:58 -0400
>>  From: pkyzivat@alum.mit.edu <mailto:pkyzivat@alum.mit.edu>
>>  To: rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>  Subject: Re: [rtcweb] codec and connection negotiation
>>
>>  I also have some observations, without conclusions, on this decision:
>>
>>  SDP o/a has some well known limitations. It would be nice if rtcweb was
>>  not saddled with them. For instance, the offer can list multiple codecs,
>>  but the answerer is free to accept more than one. In that case any of
>>  those in both offer and answer can be used. But there are UAs that
>>  support multiple codecs, but only one at a time. O/A doesn't support
>>  that well.
>>
>>  Unfortunately, relaxing the limitations may break interoperation with
>>  non-rtcweb devices. For instance if you have an rtcweb client initiating
>>  a call to a pstn destination, the gateway will likely have the limitation.
>>
>>  Querying for capabilities before initiating a call could, in *some*
>>  cases transcend that incompatibility. But not always. In general it
>>  isn't easy to guarantee that the UA you reach with a query will be the
>>  same one you reach when you try to establish a call. If not, the
>>  capabilities may be wrong.
>>
>>  SDP has needed modernizing for ages. But SDPng went down in flames.
>>  Attempting to do something like that again would probably not be a good
>>  plan for rtcweb.
>>
>>  It might be easier to go with SDP and o/a in order to ease interop, and
>>  at the same time provide a capability query mechanism, also built on SDP
>>  that could be used optionally, with the understanding that it might not
>>  be available in all cases.
>>
>>  Thanks,
>>  paul
>>
>>
>>  On 8/8/11 9:09 AM, Harald Alvestrand wrote:
>>  > Trying to doodle on the opinions (the alternative lists were quite
>>  > useful to clear the mind - I liked them a lot, and will return to them!)
>>  >
>>  > On 08/05/11 20:53, Matthew Kaufman wrote:
>>  >> Now, to follow up on my previous message with a bit of opinion...
>>  >>
>>  >> I think we should not use SDP as the API, and we should not have the
>>  >> browser implementing SDP offer-answer.
>>  >>
>>  >> 1. If SDP offer-answer is all that is available to the Javascript
>>  >> programmer (and server programmer), it becomes very difficult to know
>>  >> what the full capability set is without complex probing. This
>>  >> significantly complicates the building of future innovative
>>  >> applications on top of the browser as platform. Many of the current
>>  >> applications that show off browser capabilities do so by using
>>  >> capabilities that were already present for other reasons, and we can
>>  >> expect the same innovation to occur here, if we provide the tools.
>>  > This may be heretical ... but do we actually need to enumerate all
>>  > capabilities?
>>  > It seems to me that we'll have exactly 2 groups of capabilities:
>>  > - Routine ones, which are generated automagically when saying "I want
>>  > something working"
>>  > - Ones needed to "show off" specific new capabilities
>>  >
>>  > The platforms, which are likely reusing code from outside WebRTC, may
>>  > support many capabilities that are never required for any scenario
>>  > (especially those that have been added over the years to support
>>  > backwards compatibility in very baroque scenarios). Exposing those will
>>  > just add clutter, not add quality.
>>  >
>>  > The "show-offs" will be capable of probing.
>>  >>
>>  >> 2. Using SDP offer-answer has the advantage of reusing all the IETF
>>  >> standards work that occurs to define SDP when a new codec comes along.
>>  >> But it also has the *disadvantage* of having to wait for the IETF to
>>  >> produce these standards, which may be incomplete, or unable to control
>>  >> parameters which are necessary for web applications.
>>  > Hmmm.... since when have people *actually* waited for the IETF process
>>  > to finish before starting to use newly defined capabilities?
>>  > This may argue for looking again at the IANA procedures for SDP, and
>>  > institute something like the "provisional registries" that were
>>  > introduced for email/web.
>>  >>
>>  >> 3. Anything that is missing in SDP (example: forcing the Opus codec to
>>  >> "music" mode for encoding) will still need to be exposed as a
>>  >> Javascript API on the encoder. Thus we end up with a mix of
>>  >> possibly-conflicting settings that are adjusted via the explicit API
>>  >> and the opaque SDP API.
>>  > Won't these be things people want in SDP too? See above.
>>  >>
>>  >> 4. Other work in browsers (like recording camera and microphone to
>>  >> disk) will require the ability to directly control the encoders. If
>>  >> these APIs exist then there will definitely be conflicting settings.
>>  >> What happens if you feed in an SDP answer that requires VP8 encoding
>>  >> and then set the encoder to H.264 mode via the Javascript?
>>  > Transcoding? :-)
>>  > The same problem exists today when trying to use cameras for multiple
>>  > things concurrently.
>>  >>
>>  >>
>>  >> Matthew Kaufman
>>  >> _______________________________________________
>>  >> rtcweb mailing list
>>  >> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>  >> https://www.ietf.org/mailman/listinfo/rtcweb
>>  >>
>>  >
>>  > _______________________________________________
>>  > rtcweb mailing list
>>  > rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>  > https://www.ietf.org/mailman/listinfo/rtcweb
>>  >
>>
>>  _______________________________________________
>>  rtcweb mailing list
>>  rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>  https://www.ietf.org/mailman/listinfo/rtcweb
>


From randell-ietf@jesup.org  Wed Aug 10 07:13:57 2011
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On 8/10/2011 8:33 AM, Paul Kyzivat wrote:

> On 8/10/11 3:27 AM, Markus.Isomaki@nokia.com wrote:
>> The API itself is the critical piece. I’m not sure if there is any
>> benefit in tying it to SDP,
>
> Well, at the end of the day, an API must convey the critical data. It
> doesn't necessarily need to encode the data the same way it is encoded
> in SDP, and it doesn't need to convey that data for o/a in the same way
> that SIP does, but it does need to be able to represent the full range
> of SDP and the full semantics of o/a. And it needs to be flexible enough
> to do that as SDP evolves.

So: this theoretical format we're considering must:

a) be a superset of the functionality of SDP + o/a
b) be able to provide equivalent ability to future extensions of SDP + o/a
c) be reasonably straightforwardly convertible to/from SDP+o/a in order to
    make gatewaying feasible (this could involve only allowing a subset to
    be easily converted, at the cost of poorer gateway abilities and experience)
d) be completely vetted in time for the timeline for this spec

I can't see how anything other than either SDP+o/a or
alternate-serialization-of-SDP + o/a could meet those goals.  An alternate
serialization of SDP *might* allow for extended capabilities outside of the
ability for SDP to express (an escape valve, if you will).  I expect a
alternate serialization could be speced out in the timeframe.

Even if we decide to relax the "superset" and allow a subset of current SDP
functionality, defining that and making sure it handles the cases we want
would be tough even in a longer timeframe, unless we devise something very
simple but extensible, and live with reduced functionality in gateway
situations (and more complex gateways probably to cover) until such extensions
are done.




If someone has a real non-SDP proposal, get it on the the floor *NOW*.
Otherwise you're just waving your hands.


>> but certainly it should be possible to
>> implement SDP offer/answer protocol in Javascript on top of it. That
>> could be one requirement for it, and something that could be even proved
>> by a real implementation.
>
> Certainly a real implementation is a useful existence proof. But it is
> not a proof of completeness. Completeness will be accomplished by
> defining a full mapping between the API and SDP + o/a.
>
> *One* way to accomplish a complete mapping between the API and the SDP
> representation is to pass complete SDP in the API. Other ways are
> possible, but will be more work to define.

Exactly.

>
>      Thanks,
>      Paul
>
>> So, in summary: Avoid strict dependency to SDP, but it must be possible
>> to implement SDP offer/answer protocol on top of the standardized API.
>>
>> Markus

-- 
Randell Jesup
randell-ietf@jesup.org


From harald@alvestrand.no  Wed Aug 10 07:15:48 2011
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In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend 
one part of the scenario "4.3.3 Video conferencing system with central 
server".

I would like to add one more paragraph:

"All participant are authenticated by the central server, and authorized 
to connect to the central server. The participants are identified to 
each other by the central server, and the participants do not have 
access to each others' credentials such as e-mail addresses or login IDs".

This is necessary in order to drive use cases that resemble Google 
Hangout, where it is a requirement that people are able to participate 
without disclosing their Google login IDs to each other.
(in the particular case of Hangout, the display name on their profile 
*is* disclosed ... but that's a different matter)

The reason I think this is important is that it feeds directly into the 
discussion of what WebRTC needs to authorize: The final source or 
destination of media, or the identity of the handler at the first hop. 
In at least the case of Hangouts, the requirement is to *not* authorize 
the final source or destination.

Not sure yet how to formulate that as a requirement, and not sure yet if 
it applies to the cases without a central server, such as 4.2.6. We may 
have to decide.

                         Harald



From pkyzivat@alum.mit.edu  Wed Aug 10 07:50:57 2011
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Harald,

There was a lot of consideration of similar issues in the discussions 
around draft-ietf-simple-chat-*. What I took away is that conferencing 
implementations often want something like this, and they all spin it in 
slightly different ways. I *suspect* that in many cases the differences 
were gratuitous at the start, but that ultimately the chosen semantics 
become entrenched. As a result, it may be necessary to support an 
assortment of identity semantics - some of which may not make a lot of 
sense when looked at closely and critically.

This centered on "nicknames" to be visible in the conference, and to be 
used to support private communication between individuals in the 
conference without disclosing their actual addresses. (I guess the 
display names you mention below may be an instance of this.) We got into 
all sorts of discussions of scope of uniqueness of nicknames.

	Thanks,
	Paul

On 8/10/11 10:16 AM, Harald Alvestrand wrote:
> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
> one part of the scenario "4.3.3 Video conferencing system with central
> server".
>
> I would like to add one more paragraph:
>
> "All participant are authenticated by the central server, and authorized
> to connect to the central server. The participants are identified to
> each other by the central server, and the participants do not have
> access to each others' credentials such as e-mail addresses or login IDs".
>
> This is necessary in order to drive use cases that resemble Google
> Hangout, where it is a requirement that people are able to participate
> without disclosing their Google login IDs to each other.
> (in the particular case of Hangout, the display name on their profile
> *is* disclosed ... but that's a different matter)
>
> The reason I think this is important is that it feeds directly into the
> discussion of what WebRTC needs to authorize: The final source or
> destination of media, or the identity of the handler at the first hop.
> In at least the case of Hangouts, the requirement is to *not* authorize
> the final source or destination.
>
> Not sure yet how to formulate that as a requirement, and not sure yet if
> it applies to the cases without a central server, such as 4.2.6. We may
> have to decide.
>
> Harald
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From randell-ietf@jesup.org  Wed Aug 10 08:20:42 2011
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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 8/10/2011 10:16 AM, Harald Alvestrand wrote:

> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
> one part of the scenario "4.3.3 Video conferencing system with central
> server".
> I would like to add one more paragraph:
> "All participant are authenticated by the central server, and authorized
> to connect to the central server. The participants are identified to
> each other by the central server, and the participants do not have
> access to each others' credentials such as e-mail addresses or login IDs".
> This is necessary in order to drive use cases that resemble Google
> Hangout, where it is a requirement that people are able to participate
> without disclosing their Google login IDs to each other.
> (in the particular case of Hangout, the display name on their profile
> *is* disclosed ... but that's a different matter)
> The reason I think this is important is that it feeds directly into the
> discussion of what WebRTC needs to authorize: The final source or
> destination of media, or the identity of the handler at the first hop.
> In at least the case of Hangouts, the requirement is to *not* authorize
> the final source or destination.
Could this be handled by defining this case as a central server which is
the target of the "call", and which provides in response 1 or more video/
audio streams to the caller.  So you're only authenticating the server, and
the server is responsible for stripping and replacing the authentication
info for the forwarded streams.  The conferencing aspect is merely an aspect
of the service the server provides.  It might merge the video streams,
selectively forward them depending on speaker or on preference of the viewer,
and do similar things with audio.

While in theory it's largely covered by the point-to-point case (for example,
a point-to-point call should support multiple streams), a separate
use-case is still useful here since it both highlights conferencing-specific
issues (such as larger numbers of streams, etc).

We should consider ways for a client to indicate how many simultaneous decodes
it's willing/able to handle.  This lets the server be smart about what it
sends (it may be as simple as rejecting video streams on re-negotiation of o/a
- and that may help handle things such as where the number of streams supported
varies with the type of encoding/resolution/frame-rate, or other client-side loads).

For example, as people join a "Hangout", the server would re-negotiate to add
a video/audio pair (using SDP grouping I assume).  If the client can't handle
another one, it could reject the add (or renegotiate to lower the complexity/size
of existing streams, perhaps).  The server might then send the N most "interesting"
streams, or combine streams, etc.  SDP+o/a may not be optimal here but I think it
would get the job done.

Note that this requires the central server to decrypt and re-encrypt the
streams (as I believe Harald's wording above would).

The "and authorized to connect to the central server" I think is unneeded -
that's up to the policy of the server; there's no reason it can't allow
arbitrary (non-authorized) connections if the service/server so choose.
The same for authenticating the users - that's up to the server/service.
One can run encryption without authenticating the users.

-- 
Randell Jesup
randell-ietf@jesup.org


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I have taken some of the information I learned from the discussions in 
Quebec City about the issues of putting video and audio into the same 
RTP session and created a draft from them outlining a solution to the 
problem of signalling this configuration using SDP.

Comments welcome, including the recommended context for processing the 
document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.

                       Harald

-------- Original Message --------
Subject: 	I-D Action: draft-alvestrand-one-rtp-00.txt
Date: 	Thu, 11 Aug 2011 04:28:09 -0700
From: 	internet-drafts@ietf.org
Reply-To: 	internet-drafts@ietf.org
To: 	i-d-announce@ietf.org



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

	Title           : SDP Grouping for Single RTP Sessions
	Author(s)       : Harald Tveit Alvestrand
	Filename        : draft-alvestrand-one-rtp-00.txt
	Pages           : 8
	Date            : 2011-08-11

    This document describes an extension to the Session Description
    Protocol (SDP) to describe RTP sessions where media of multiple top
    level types, for example audio and video, are carried in the same RTP
    session.

    This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
    consideration.



A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
_______________________________________________
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<html>
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    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#ffffff" text="#000000">
    I have taken some of the information I learned from the discussions
    in Quebec City about the issues of putting video and audio into the
    same RTP session and created a draft from them outlining a solution
    to the problem of signalling this configuration using SDP.<br>
    <br>
    Comments welcome, including the recommended context for processing
    the document; in a few days I'll send the same note to AVTCORE
    and/or MMUSIC.<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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>I-D Action: draft-alvestrand-one-rtp-00.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Thu, 11 Aug 2011 04:28:09 -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">Reply-To:
          </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:i-d-announce@ietf.org">i-d-announce@ietf.org</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : SDP Grouping for Single RTP Sessions
	Author(s)       : Harald Tveit Alvestrand
	Filename        : draft-alvestrand-one-rtp-00.txt
	Pages           : 8
	Date            : 2011-08-11

   This document describes an extension to the Session Description
   Protocol (SDP) to describe RTP sessions where media of multiple top
   level types, for example audio and video, are carried in the same RTP
   session.

   This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
   consideration.



A URL for this Internet-Draft is:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a>

Internet-Drafts are also available by anonymous FTP at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a>

This Internet-Draft can be retrieved at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a>
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Internet-Draft directories: <a class="moz-txt-link-freetext" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a>
or <a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a>

</pre>
  </body>
</html>

--------------040406040802010401070204--

From stefan.lk.hakansson@ericsson.com  Thu Aug 11 04:36:52 2011
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Thu, 11 Aug 2011 13:33:51 +0200
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Harald Alvestrand wrote:
>In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>one part of the scenario "4.3.3 Video conferencing system with central
>server".
>
>I would like to add one more paragraph:
>
>"All participant are authenticated by the central server, and authorized
>to connect to the central server. The participants are identified to
>each other by the central server, and the participants do not have
>access to each others' credentials such as e-mail addresses or login IDs".
I think this paragraph makes a lot of sense, and would be happy to add it. =
However, I=92m not 100% convinced that it would add requirements that are i=
n scope for webrtc or rtcweb.

When writing up this use case, the architecture in mind was centred around =
a web server that carries out the functionality of serving the web app, han=
dling users, authenticating them, authorising them, allowing them to commun=
icate and so on. That web server would control the central (media) server, =
which in turn is responsible only for establishing connections for RTC with=
 browsers, mixing audio and selecting video between the users (browsers) se=
lected by the web server, etc.

This would mean that user management, including determining what user ident=
ity is revealed to others, is controlled by the web server. I guess this is=
 done already today for many services. What we will add is the possibility =
to communicate using audio and video without plug-ins.

Does this make sense or not?

Stefan=

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On 08/11/11 13:33, Stefan Håkansson LK wrote:
> Harald Alvestrand wrote:
>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>> one part of the scenario "4.3.3 Video conferencing system with central
>> server".
>>
>> I would like to add one more paragraph:
>>
>> "All participant are authenticated by the central server, and authorized
>> to connect to the central server. The participants are identified to
>> each other by the central server, and the participants do not have
>> access to each others' credentials such as e-mail addresses or login IDs".
> I think this paragraph makes a lot of sense, and would be happy to add it. However, I’m not 100% convinced that it would add requirements that are in scope for webrtc or rtcweb.
>
> When writing up this use case, the architecture in mind was centred around a web server that carries out the functionality of serving the web app, handling users, authenticating them, authorising them, allowing them to communicate and so on. That web server would control the central (media) server, which in turn is responsible only for establishing connections for RTC with browsers, mixing audio and selecting video between the users (browsers) selected by the web server, etc.
>
> This would mean that user management, including determining what user identity is revealed to others, is controlled by the web server. I guess this is done already today for many services. What we will add is the possibility to communicate using audio and video without plug-ins.
>
> Does this make sense or not?
This makes sense. I just want it to be explicit.
It's relevant for the discussion EKR brought up about "who do I 
authorize when I say OK to using a camera" - if it's the far end of the 
connection or the service I'm connecting to.

In the centralized server case, with the added text, it's definitely the 
service, and I'd like to keep it that way. So this is text added in 
order to make sure we don't generate a requirement....


From stefan.lk.hakansson@ericsson.com  Thu Aug 11 07:00:35 2011
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From: =?Windows-1252?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Date: Thu, 11 Aug 2011 16:00:03 +0200
Thread-Topic: [rtcweb] Use case change request: Identity in multiuser calls
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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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OK! Then I think we have the same view. Will you work on the wording, or sh=
ould we start with the current one?=20

Stefan
________________________________________
From: Harald Alvestrand [harald@alvestrand.no]
Sent: Thursday, August 11, 2011 1:47 PM
To: Stefan H=E5kansson LK
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls

On 08/11/11 13:33, Stefan H=E5kansson LK wrote:
> Harald Alvestrand wrote:
>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>> one part of the scenario "4.3.3 Video conferencing system with central
>> server".
>>
>> I would like to add one more paragraph:
>>
>> "All participant are authenticated by the central server, and authorized
>> to connect to the central server. The participants are identified to
>> each other by the central server, and the participants do not have
>> access to each others' credentials such as e-mail addresses or login IDs=
".
> I think this paragraph makes a lot of sense, and would be happy to add it=
. However, I=92m not 100% convinced that it would add requirements that are=
 in scope for webrtc or rtcweb.
>
> When writing up this use case, the architecture in mind was centred aroun=
d a web server that carries out the functionality of serving the web app, h=
andling users, authenticating them, authorising them, allowing them to comm=
unicate and so on. That web server would control the central (media) server=
, which in turn is responsible only for establishing connections for RTC wi=
th browsers, mixing audio and selecting video between the users (browsers) =
selected by the web server, etc.
>
> This would mean that user management, including determining what user ide=
ntity is revealed to others, is controlled by the web server. I guess this =
is done already today for many services. What we will add is the possibilit=
y to communicate using audio and video without plug-ins.
>
> Does this make sense or not?
This makes sense. I just want it to be explicit.
It's relevant for the discussion EKR brought up about "who do I
authorize when I say OK to using a camera" - if it's the far end of the
connection or the service I'm connecting to.

In the centralized server case, with the added text, it's definitely the
service, and I'd like to keep it that way. So this is text added in
order to make sure we don't generate a requirement....


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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 08/11/11 16:00, Stefan Håkansson LK wrote:
> OK! Then I think we have the same view. Will you work on the wording, or should we start with the current one?
I'm happy if you just insert the paragraph I suggested. Feel free to 
wordsmith, though.

> Stefan
> ________________________________________
> From: Harald Alvestrand [harald@alvestrand.no]
> Sent: Thursday, August 11, 2011 1:47 PM
> To: Stefan Håkansson LK
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
>
> On 08/11/11 13:33, Stefan Håkansson LK wrote:
>> Harald Alvestrand wrote:
>>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>>> one part of the scenario "4.3.3 Video conferencing system with central
>>> server".
>>>
>>> I would like to add one more paragraph:
>>>
>>> "All participant are authenticated by the central server, and authorized
>>> to connect to the central server. The participants are identified to
>>> each other by the central server, and the participants do not have
>>> access to each others' credentials such as e-mail addresses or login IDs".
>> I think this paragraph makes a lot of sense, and would be happy to add it. However, I’m not 100% convinced that it would add requirements that are in scope for webrtc or rtcweb.
>>
>> When writing up this use case, the architecture in mind was centred around a web server that carries out the functionality of serving the web app, handling users, authenticating them, authorising them, allowing them to communicate and so on. That web server would control the central (media) server, which in turn is responsible only for establishing connections for RTC with browsers, mixing audio and selecting video between the users (browsers) selected by the web server, etc.
>>
>> This would mean that user management, including determining what user identity is revealed to others, is controlled by the web server. I guess this is done already today for many services. What we will add is the possibility to communicate using audio and video without plug-ins.
>>
>> Does this make sense or not?
> This makes sense. I just want it to be explicit.
> It's relevant for the discussion EKR brought up about "who do I
> authorize when I say OK to using a camera" - if it's the far end of the
> connection or the service I'm connecting to.
>
> In the centralized server case, with the added text, it's definitely the
> service, and I'd like to keep it that way. So this is text added in
> order to make sure we don't generate a requirement....
>
>


From harald@alvestrand.no  Thu Aug 11 08:08:27 2011
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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 08/10/11 17:19, Randell Jesup wrote:
> On 8/10/2011 10:16 AM, Harald Alvestrand wrote:
>
>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>> one part of the scenario "4.3.3 Video conferencing system with central
>> server".
>> I would like to add one more paragraph:
>> "All participant are authenticated by the central server, and authorized
>> to connect to the central server. The participants are identified to
>> each other by the central server, and the participants do not have
>> access to each others' credentials such as e-mail addresses or login 
>> IDs".
>> This is necessary in order to drive use cases that resemble Google
>> Hangout, where it is a requirement that people are able to participate
>> without disclosing their Google login IDs to each other.
>> (in the particular case of Hangout, the display name on their profile
>> *is* disclosed ... but that's a different matter)
>> The reason I think this is important is that it feeds directly into the
>> discussion of what WebRTC needs to authorize: The final source or
>> destination of media, or the identity of the handler at the first hop.
>> In at least the case of Hangouts, the requirement is to *not* authorize
>> the final source or destination.
> Could this be handled by defining this case as a central server which is
> the target of the "call", and which provides in response 1 or more video/
> audio streams to the caller.  So you're only authenticating the 
> server, and
> the server is responsible for stripping and replacing the authentication
> info for the forwarded streams.  The conferencing aspect is merely an 
> aspect
> of the service the server provides.  It might merge the video streams,
> selectively forward them depending on speaker or on preference of the 
> viewer,
> and do similar things with audio.
I think parts of that is an implementation strategy for the scenario in 
4.3.3 - it doesn't have to be that way, so we shouldn't define it as 
part of the scenario.
>
> While in theory it's largely covered by the point-to-point case (for 
> example,
> a point-to-point call should support multiple streams), a separate
> use-case is still useful here since it both highlights 
> conferencing-specific
> issues (such as larger numbers of streams, etc).
>
> We should consider ways for a client to indicate how many simultaneous 
> decodes
> it's willing/able to handle.  This lets the server be smart about what it
> sends (it may be as simple as rejecting video streams on 
> re-negotiation of o/a
> - and that may help handle things such as where the number of streams 
> supported
> varies with the type of encoding/resolution/frame-rate, or other 
> client-side loads).
If we do multiple video streams inside an RTP session, we don't need 
re-negotiation of o/a to add more streams. We might consider the 
proposed extensions in 
draft-westerlund-avtcore-multistream-and-simulcast to add information 
about number of streams to O/A (he suggests that for max backwards 
compatibility, we should assume that no more than one stream per RTP 
session can be handled unless the extension is used, but that might be 
an overly conservative suggestion).

>
> For example, as people join a "Hangout", the server would re-negotiate 
> to add
> a video/audio pair (using SDP grouping I assume).  If the client can't 
> handle
> another one, it could reject the add (or renegotiate to lower the 
> complexity/size
> of existing streams, perhaps).  The server might then send the N most 
> "interesting"
> streams, or combine streams, etc.  SDP+o/a may not be optimal here but 
> I think it
> would get the job done.
Hangouts put all the video streams into one RTP session, and all the 
audio streams into another RTP session. Renegotiation would be needed if 
it didn't.
>
> Note that this requires the central server to decrypt and re-encrypt the
> streams (as I believe Harald's wording above would).
Not necessarily - if key negotiation goes via the server, the server 
could make sure the key used in the server-to-client direction is the 
same key as that used in the client-to-server direction - but if you 
want to put 2 media streams into the same RTP session, and do keying per 
RTP session, the server would have to decrypt-then-encrypt.
>
>
> The "and authorized to connect to the central server" I think is 
> unneeded -
> that's up to the policy of the server; there's no reason it can't allow
> arbitrary (non-authorized) connections if the service/server so choose.
> The same for authenticating the users - that's up to the server/service.
> One can run encryption without authenticating the users.
>
Yup - when I was writing "authorized to connect", I was thinking of 
anonymous/open access as "policy: everyone's authorized". Similarly, 
authentication - not authenticating is usually easier than 
authenticating, so we don't need to add extra text to cover the 
unauthenticated case.
(showing my age: FTP doesn't really do unauthenticated mode; it just has 
the convention that "anonymous" has a blank password....)

                  Harald


From henry.sinnreich@gmail.com  Thu Aug 11 09:06:37 2011
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>The participants are identified to
> each other by the central server

There may be no contradiction, but just to make sure the scenario, such as a
confidential business meeting is included (not necessarily a social network
scenario). 

A confidential conference MUST assure that all participants can also be
safely identified first to each other, not necessarily by the server, end to
end, to the full satisfaction of all the participants, who would otherwise
refuse to continue  to speak up or continue to attend at all.
Who provides for the e2e identification? A 3rd party? Inside the app?

Did I understand correctly there is no contradiction?
Any problem in clarifying this?

Thanks, Henry


On 8/10/11 9:16 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:

> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
> one part of the scenario "4.3.3 Video conferencing system with central
> server".
> 
> I would like to add one more paragraph:
> 
> "All participant are authenticated by the central server, and authorized
> to connect to the central server. The participants are identified to
> each other by the central server, and the participants do not have
> access to each others' credentials such as e-mail addresses or login IDs".
> 
> This is necessary in order to drive use cases that resemble Google
> Hangout, where it is a requirement that people are able to participate
> without disclosing their Google login IDs to each other.
> (in the particular case of Hangout, the display name on their profile
> *is* disclosed ... but that's a different matter)
> 
> The reason I think this is important is that it feeds directly into the
> discussion of what WebRTC needs to authorize: The final source or
> destination of media, or the identity of the handler at the first hop.
> In at least the case of Hangouts, the requirement is to *not* authorize
> the final source or destination.
> 
> Not sure yet how to formulate that as a requirement, and not sure yet if
> it applies to the cases without a central server, such as 4.2.6. We may
> have to decide.
> 
>                          Harald
> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



From pkyzivat@alum.mit.edu  Thu Aug 11 09:50:24 2011
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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>> The participants are identified to
>> each other by the central server
>
> There may be no contradiction, but just to make sure the scenario, such as a
> confidential business meeting is included (not necessarily a social network
> scenario).
>
> A confidential conference MUST assure that all participants can also be
> safely identified first to each other, not necessarily by the server, end to
> end, to the full satisfaction of all the participants, who would otherwise
> refuse to continue  to speak up or continue to attend at all.
> Who provides for the e2e identification? A 3rd party? Inside the app?
>
> Did I understand correctly there is no contradiction?
> Any problem in clarifying this?

This is an example of the sort of thing I was speaking of in my earlier 
reply - that there are differing levels of identity that may be desired.

In some cases, participants may want to be fully anonymous. But even 
then, it will typically be important that one participant be able to 
identify *which* of the other (anonymous) participants it is receiving 
media from. (E.g. by noting which participant in the roster is the 
current speaker - anon1, anon2, ...)

In other cases the participants *true* identities are hidden, but 
handles/nicknames may be used to identify them in the roster and in 
chats. ("snoopy" and "red barron" rather than anon1 and anon2.)

But in that case there are issues of how the handles are associated with 
participants. It could be that handles are permanently assigned to real 
users by the server. Then you can be confident that when talking to 
"snoopy" it is always someone well defined - same as the last time - 
without awareness of who that person *actually* is.

Alternatively, the server might allow handles to be used on a fcfs basis 
per conference. Or it might allow handles to be non-unique, so that you 
could have two "snoopy"s at the same time, bound to unrelated users.

Henry - what sort of mechanism did you have in mind for these anonymous 
participants to identify themselves to one another? Are you talking 
about the exchange of credentials using the server as an intermediary in 
the exchange? Or did you have in mind an exchange outside the server. 
(E.g. Bob establishes a secure connection to Alice and over it tells her 
"I'm anon2 in conference foo right now"?

	Thanks,
	Paul

> Thanks, Henry
>
>
> On 8/10/11 9:16 AM, "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>
>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>> one part of the scenario "4.3.3 Video conferencing system with central
>> server".
>>
>> I would like to add one more paragraph:
>>
>> "All participant are authenticated by the central server, and authorized
>> to connect to the central server. The participants are identified to
>> each other by the central server, and the participants do not have
>> access to each others' credentials such as e-mail addresses or login IDs".
>>
>> This is necessary in order to drive use cases that resemble Google
>> Hangout, where it is a requirement that people are able to participate
>> without disclosing their Google login IDs to each other.
>> (in the particular case of Hangout, the display name on their profile
>> *is* disclosed ... but that's a different matter)
>>
>> The reason I think this is important is that it feeds directly into the
>> discussion of what WebRTC needs to authorize: The final source or
>> destination of media, or the identity of the handler at the first hop.
>> In at least the case of Hangouts, the requirement is to *not* authorize
>> the final source or destination.
>>
>> Not sure yet how to formulate that as a requirement, and not sure yet if
>> it applies to the cases without a central server, such as 4.2.6. We may
>> have to decide.
>>
>>                           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
>


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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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+1.

(Incidentally, this IdM model is consistent with the ITU-T Identity 
Management requirements for next generation networks.)

Igor

On 8/11/2011 7:47 AM, Harald Alvestrand wrote:
> On 08/11/11 13:33, Stefan Håkansson LK wrote:
>> Harald Alvestrand wrote:
>>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>>> one part of the scenario "4.3.3 Video conferencing system with central
>>> server".
>>>
>>> I would like to add one more paragraph:
>>>
>>> "All participant are authenticated by the central server, and 
>>> authorized
>>> to connect to the central server. The participants are identified to
>>> each other by the central server, and the participants do not have
>>> access to each others' credentials such as e-mail addresses or login 
>>> IDs".
>> I think this paragraph makes a lot of sense, and would be happy to 
>> add it. However, I’m not 100% convinced that it would add 
>> requirements that are in scope for webrtc or rtcweb.
>>
>> When writing up this use case, the architecture in mind was centred 
>> around a web server that carries out the functionality of serving the 
>> web app, handling users, authenticating them, authorising them, 
>> allowing them to communicate and so on. That web server would control 
>> the central (media) server, which in turn is responsible only for 
>> establishing connections for RTC with browsers, mixing audio and 
>> selecting video between the users (browsers) selected by the web 
>> server, etc.
>>
>> This would mean that user management, including determining what user 
>> identity is revealed to others, is controlled by the web server. I 
>> guess this is done already today for many services. What we will add 
>> is the possibility to communicate using audio and video without 
>> plug-ins.
>>
>> Does this make sense or not?
> This makes sense. I just want it to be explicit.
> It's relevant for the discussion EKR brought up about "who do I 
> authorize when I say OK to using a camera" - if it's the far end of 
> the connection or the service I'm connecting to.
>
> In the centralized server case, with the added text, it's definitely 
> the service, and I'd like to keep it that way. So this is text added 
> in order to make sure we don't generate a requirement....
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From ted.ietf@gmail.com  Thu Aug 11 11:15:10 2011
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Thanks for everyone who participated in the Doodle Poll.  Pending
approval of the ADs, the interim meeting will be held on September 8,
2011 at 7:00 A.M. PDT; we've allotted 4 hours, and the details of
participation will follow.

As is obvious from that date, we're hoping to make progress on a very
aggressive scale here, and if you owe a document it would be good to
get started on it as soon as possible!

thanks,

Ted, Cullen, Magnus

From pkyzivat@alum.mit.edu  Thu Aug 11 11:46:28 2011
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On 8/11/11 7:32 AM, Harald Alvestrand wrote:
> I have taken some of the information I learned from the discussions in
> Quebec City about the issues of putting video and audio into the same
> RTP session and created a draft from them outlining a solution to the
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.

Harald,

A few questions/comments:

1) If ICE is being used, how do you expect it to be handled on the 2nd 
m-line? Since one of the goals of multiplexing on a single rtp session 
is to minimize ICE processing, one would hope only to do ICE on the 
first of the grouped m-lines. But if the answerer doesn't support this 
grouping, then the ICE will be needed on the others.

2) The draft only shows this mechanism being used to combine one audio 
and one video. Do you also imagine it being used to represent multiple 
audio and/or multiple video streams that might otherwise be handled 
independently? (I see no reason why it couldn't be used that way.) This 
would allow description of different properties for each one.

3) In the example, for an entity that doesn't understand TOGETHER, you 
still show the a=mid lines. This is appropriate for an answerer that 
*does* understand a=group. However an answerer that doesn't implement 
rfc3388 would presumably omit the a=mid lines from the answer. So the 
offerer had better be prepared for that as well.

	Thanks,
	Paul

From henry.sinnreich@gmail.com  Thu Aug 11 12:11:59 2011
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From: Henry Sinnreich <henry.sinnreich@gmail.com>
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> Henry - what sort of mechanism did you have in mind for these anonymous
> participants to identify themselves to one another? Are you talking
> about the exchange of credentials using the server as an intermediary in
> the exchange? Or did you have in mind an exchange outside the server.
> (E.g. Bob establishes a secure connection to Alice and over it tells her
> "I'm anon2 in conference foo right now"?

The most credible e2e identification that comes to my mind is Phil
Zimmermann's Zfone technology:

http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
http://zfoneproject.com/

Definitely not relying on any "trusted" server :-)

Thanks, Henry

On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu> wrote:

> On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>> The participants are identified to
>>> each other by the central server
>> 
>> There may be no contradiction, but just to make sure the scenario, such as a
>> confidential business meeting is included (not necessarily a social network
>> scenario).
>> 
>> A confidential conference MUST assure that all participants can also be
>> safely identified first to each other, not necessarily by the server, end to
>> end, to the full satisfaction of all the participants, who would otherwise
>> refuse to continue  to speak up or continue to attend at all.
>> Who provides for the e2e identification? A 3rd party? Inside the app?
>> 
>> Did I understand correctly there is no contradiction?
>> Any problem in clarifying this?
> 
> This is an example of the sort of thing I was speaking of in my earlier
> reply - that there are differing levels of identity that may be desired.
> 
> In some cases, participants may want to be fully anonymous. But even
> then, it will typically be important that one participant be able to
> identify *which* of the other (anonymous) participants it is receiving
> media from. (E.g. by noting which participant in the roster is the
> current speaker - anon1, anon2, ...)
> 
> In other cases the participants *true* identities are hidden, but
> handles/nicknames may be used to identify them in the roster and in
> chats. ("snoopy" and "red barron" rather than anon1 and anon2.)
> 
> But in that case there are issues of how the handles are associated with
> participants. It could be that handles are permanently assigned to real
> users by the server. Then you can be confident that when talking to
> "snoopy" it is always someone well defined - same as the last time -
> without awareness of who that person *actually* is.
> 
> Alternatively, the server might allow handles to be used on a fcfs basis
> per conference. Or it might allow handles to be non-unique, so that you
> could have two "snoopy"s at the same time, bound to unrelated users.
> 
> Henry - what sort of mechanism did you have in mind for these anonymous
> participants to identify themselves to one another? Are you talking
> about the exchange of credentials using the server as an intermediary in
> the exchange? Or did you have in mind an exchange outside the server.
> (E.g. Bob establishes a secure connection to Alice and over it tells her
> "I'm anon2 in conference foo right now"?
> 
> Thanks,
> Paul
> 
>> Thanks, Henry
>> 
>> 
>> On 8/10/11 9:16 AM, "Harald Alvestrand"<harald@alvestrand.no>  wrote:
>> 
>>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>>> one part of the scenario "4.3.3 Video conferencing system with central
>>> server".
>>> 
>>> I would like to add one more paragraph:
>>> 
>>> "All participant are authenticated by the central server, and authorized
>>> to connect to the central server. The participants are identified to
>>> each other by the central server, and the participants do not have
>>> access to each others' credentials such as e-mail addresses or login IDs".
>>> 
>>> This is necessary in order to drive use cases that resemble Google
>>> Hangout, where it is a requirement that people are able to participate
>>> without disclosing their Google login IDs to each other.
>>> (in the particular case of Hangout, the display name on their profile
>>> *is* disclosed ... but that's a different matter)
>>> 
>>> The reason I think this is important is that it feeds directly into the
>>> discussion of what WebRTC needs to authorize: The final source or
>>> destination of media, or the identity of the handler at the first hop.
>>> In at least the case of Hangouts, the requirement is to *not* authorize
>>> the final source or destination.
>>> 
>>> Not sure yet how to formulate that as a requirement, and not sure yet if
>>> it applies to the cases without a central server, such as 4.2.6. We may
>>> have to decide.
>>> 
>>>                           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
>> 
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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Well...

(I thought that the lawful intercept issue that came up several times is 
pertinent here.)  But note that this was a parenthetical remark!

A non-parenthetical one is based on the quote from the cited draft:

"In the development of ZRTP, it was realized that there are scenarios
    in which the media can not be encrypted end-to-end.  For example,
    when a client has a trusted server or PBX which provides media
    services in the path.  For these cases, ZRTP developed mechanisms for
    handling a "trusted MiTM" which can terminate than reoriginate the
    SRTP encryption."



So, if we end up with a "trusted MITM," why not have a trusted central 
server in the first place?

Igor

P.S.:  I have nothing against ZRTP, which has done a thorough and honest 
job.  I question its applicability to the agreed use cases. Personally I 
think we need an all-encompassing solution.


On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>> Henry - what sort of mechanism did you have in mind for these anonymous
>> participants to identify themselves to one another? Are you talking
>> about the exchange of credentials using the server as an intermediary in
>> the exchange? Or did you have in mind an exchange outside the server.
>> (E.g. Bob establishes a secure connection to Alice and over it tells her
>> "I'm anon2 in conference foo right now"?
> The most credible e2e identification that comes to my mind is Phil
> Zimmermann's Zfone technology:
>
> http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
> http://zfoneproject.com/
>
> Definitely not relying on any "trusted" server :-)
>
> Thanks, Henry
>
> On 8/11/11 11:50 AM, "Paul Kyzivat"<pkyzivat@alum.mit.edu>  wrote:
>
>> On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>>> The participants are identified to
>>>> each other by the central server
>>> There may be no contradiction, but just to make sure the scenario, such as a
>>> confidential business meeting is included (not necessarily a social network
>>> scenario).
>>>
>>> A confidential conference MUST assure that all participants can also be
>>> safely identified first to each other, not necessarily by the server, end to
>>> end, to the full satisfaction of all the participants, who would otherwise
>>> refuse to continue  to speak up or continue to attend at all.
>>> Who provides for the e2e identification? A 3rd party? Inside the app?
>>>
>>> Did I understand correctly there is no contradiction?
>>> Any problem in clarifying this?
>> This is an example of the sort of thing I was speaking of in my earlier
>> reply - that there are differing levels of identity that may be desired.
>>
>> In some cases, participants may want to be fully anonymous. But even
>> then, it will typically be important that one participant be able to
>> identify *which* of the other (anonymous) participants it is receiving
>> media from. (E.g. by noting which participant in the roster is the
>> current speaker - anon1, anon2, ...)
>>
>> In other cases the participants *true* identities are hidden, but
>> handles/nicknames may be used to identify them in the roster and in
>> chats. ("snoopy" and "red barron" rather than anon1 and anon2.)
>>
>> But in that case there are issues of how the handles are associated with
>> participants. It could be that handles are permanently assigned to real
>> users by the server. Then you can be confident that when talking to
>> "snoopy" it is always someone well defined - same as the last time -
>> without awareness of who that person *actually* is.
>>
>> Alternatively, the server might allow handles to be used on a fcfs basis
>> per conference. Or it might allow handles to be non-unique, so that you
>> could have two "snoopy"s at the same time, bound to unrelated users.
>>
>> Henry - what sort of mechanism did you have in mind for these anonymous
>> participants to identify themselves to one another? Are you talking
>> about the exchange of credentials using the server as an intermediary in
>> the exchange? Or did you have in mind an exchange outside the server.
>> (E.g. Bob establishes a secure connection to Alice and over it tells her
>> "I'm anon2 in conference foo right now"?
>>
>> Thanks,
>> Paul
>>
>>> Thanks, Henry
>>>
>>>
>>> On 8/10/11 9:16 AM, "Harald Alvestrand"<harald@alvestrand.no>   wrote:
>>>
>>>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>>>> one part of the scenario "4.3.3 Video conferencing system with central
>>>> server".
>>>>
>>>> I would like to add one more paragraph:
>>>>
>>>> "All participant are authenticated by the central server, and authorized
>>>> to connect to the central server. The participants are identified to
>>>> each other by the central server, and the participants do not have
>>>> access to each others' credentials such as e-mail addresses or login IDs".
>>>>
>>>> This is necessary in order to drive use cases that resemble Google
>>>> Hangout, where it is a requirement that people are able to participate
>>>> without disclosing their Google login IDs to each other.
>>>> (in the particular case of Hangout, the display name on their profile
>>>> *is* disclosed ... but that's a different matter)
>>>>
>>>> The reason I think this is important is that it feeds directly into the
>>>> discussion of what WebRTC needs to authorize: The final source or
>>>> destination of media, or the identity of the handler at the first hop.
>>>> In at least the case of Hangouts, the requirement is to *not* authorize
>>>> the final source or destination.
>>>>
>>>> Not sure yet how to formulate that as a requirement, and not sure yet if
>>>> it applies to the cases without a central server, such as 4.2.6. We may
>>>> have to decide.
>>>>
>>>>                            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
>>>
>> _______________________________________________
>> 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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    Well...<br>
    <br>
    (I thought that the lawful intercept issue that came up several
    times is pertinent here.)&nbsp; But note that this was a parenthetical
    remark!<br>
    <br>
    A non-parenthetical one is based on the quote from the cited draft:<br>
    <br>
    <span class="Apple-style-span" style="color: rgb(0, 0, 0);
      font-family: 'Times New Roman'; 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; font-size:
      medium;">
      <pre style="word-wrap: break-word; white-space: pre-wrap;">"In the development of ZRTP, it was realized that there are scenarios
   in which the media can not be encrypted end-to-end.  For example,
   when a client has a trusted server or PBX which provides media
   services in the path.  For these cases, ZRTP developed mechanisms for
   handling a "trusted MiTM" which can terminate than reoriginate the
   SRTP encryption." </pre>
    </span><br>
    <br>
    So, if we end up with a "trusted MITM," why not have a trusted
    central server in the first place?<br>
    <br>
    Igor<br>
    <br>
    P.S.:&nbsp; I have nothing against ZRTP, which has done a thorough and
    honest job.&nbsp; I question its applicability to the agreed use cases.
    Personally I think we need an all-encompassing solution.<br>
    <br>
    <br>
    On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
    <blockquote cite="mid:CA69938B.1CB98%25henry.sinnreich@gmail.com"
      type="cite">
      <blockquote type="cite">
        <pre wrap="">Henry - what sort of mechanism did you have in mind for these anonymous
participants to identify themselves to one another? Are you talking
about the exchange of credentials using the server as an intermediary in
the exchange? Or did you have in mind an exchange outside the server.
(E.g. Bob establishes a secure connection to Alice and over it tells her
"I'm anon2 in conference foo right now"?
</pre>
      </blockquote>
      <pre wrap="">
The most credible e2e identification that comes to my mind is Phil
Zimmermann's Zfone technology:

<a class="moz-txt-link-freetext" href="http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt">http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt</a>
<a class="moz-txt-link-freetext" href="http://zfoneproject.com/">http://zfoneproject.com/</a>

Definitely not relying on any "trusted" server :-)

Thanks, Henry

On 8/11/11 11:50 AM, "Paul Kyzivat" <a class="moz-txt-link-rfc2396E" href="mailto:pkyzivat@alum.mit.edu">&lt;pkyzivat@alum.mit.edu&gt;</a> wrote:

</pre>
      <blockquote type="cite">
        <pre wrap="">On 8/11/11 12:07 PM, Henry Sinnreich wrote:
</pre>
        <blockquote type="cite">
          <blockquote type="cite">
            <pre wrap="">The participants are identified to
each other by the central server
</pre>
          </blockquote>
          <pre wrap="">
There may be no contradiction, but just to make sure the scenario, such as a
confidential business meeting is included (not necessarily a social network
scenario).

A confidential conference MUST assure that all participants can also be
safely identified first to each other, not necessarily by the server, end to
end, to the full satisfaction of all the participants, who would otherwise
refuse to continue  to speak up or continue to attend at all.
Who provides for the e2e identification? A 3rd party? Inside the app?

Did I understand correctly there is no contradiction?
Any problem in clarifying this?
</pre>
        </blockquote>
        <pre wrap="">
This is an example of the sort of thing I was speaking of in my earlier
reply - that there are differing levels of identity that may be desired.

In some cases, participants may want to be fully anonymous. But even
then, it will typically be important that one participant be able to
identify *which* of the other (anonymous) participants it is receiving
media from. (E.g. by noting which participant in the roster is the
current speaker - anon1, anon2, ...)

In other cases the participants *true* identities are hidden, but
handles/nicknames may be used to identify them in the roster and in
chats. ("snoopy" and "red barron" rather than anon1 and anon2.)

But in that case there are issues of how the handles are associated with
participants. It could be that handles are permanently assigned to real
users by the server. Then you can be confident that when talking to
"snoopy" it is always someone well defined - same as the last time -
without awareness of who that person *actually* is.

Alternatively, the server might allow handles to be used on a fcfs basis
per conference. Or it might allow handles to be non-unique, so that you
could have two "snoopy"s at the same time, bound to unrelated users.

Henry - what sort of mechanism did you have in mind for these anonymous
participants to identify themselves to one another? Are you talking
about the exchange of credentials using the server as an intermediary in
the exchange? Or did you have in mind an exchange outside the server.
(E.g. Bob establishes a secure connection to Alice and over it tells her
"I'm anon2 in conference foo right now"?

Thanks,
Paul

</pre>
        <blockquote type="cite">
          <pre wrap="">Thanks, Henry


On 8/10/11 9:16 AM, "Harald Alvestrand"<a class="moz-txt-link-rfc2396E" href="mailto:harald@alvestrand.no">&lt;harald@alvestrand.no&gt;</a>  wrote:

</pre>
          <blockquote type="cite">
            <pre wrap="">In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
one part of the scenario "4.3.3 Video conferencing system with central
server".

I would like to add one more paragraph:

"All participant are authenticated by the central server, and authorized
to connect to the central server. The participants are identified to
each other by the central server, and the participants do not have
access to each others' credentials such as e-mail addresses or login IDs".

This is necessary in order to drive use cases that resemble Google
Hangout, where it is a requirement that people are able to participate
without disclosing their Google login IDs to each other.
(in the particular case of Hangout, the display name on their profile
*is* disclosed ... but that's a different matter)

The reason I think this is important is that it feeds directly into the
discussion of what WebRTC needs to authorize: The final source or
destination of media, or the identity of the handler at the first hop.
In at least the case of Hangouts, the requirement is to *not* authorize
the final source or destination.

Not sure yet how to formulate that as a requirement, and not sure yet if
it applies to the cases without a central server, such as 4.2.6. We may
have to decide.

                          Harald


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

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

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

_______________________________________________
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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Date: Thu, 11 Aug 2011 16:09:43 -0500
From: Henry Sinnreich <henry.sinnreich@gmail.com>
To: <igor.faynberg@alcatel-lucent.com>
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Thread-Topic: [rtcweb] Use case change request: Identity in multiuser calls
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Cc: rtcweb@ietf.org, Phil Zimmermann <prz@mit.edu>
Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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> So, if we end up with a "trusted MITM," why not have a trusted central server
in the first place?

draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support various
other scenarios, but how I understand it, no PBX or any intermediary is
required.
The default scenario is e2e.

Maybe the authors copied here can clarify.

Thanks, Henry



On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com>
wrote:

>    Well...
>  
>  (I thought that the lawful intercept issue that came up several times is
> pertinent here.)  But note that this was a parenthetical remark!
>  
>  A non-parenthetical one is based on the quote from the cited draft:
>  
>   
> "In the development of ZRTP, it was realized that there are scenarios
>    in which the media can not be encrypted end-to-end.  For example,
>    when a client has a trusted server or PBX which provides media
>    services in the path.  For these cases, ZRTP developed mechanisms for
>    handling a "trusted MiTM" which can terminate than reoriginate the
>    SRTP encryption."
>  
>  
>  So, if we end up with a "trusted MITM," why not have a trusted central server
> in the first place?
>  
>  Igor
>  
>  P.S.:  I have nothing against ZRTP, which has done a thorough and honest job.
> I question its applicability to the agreed use cases. Personally I think we
> need an all-encompassing solution.
>  
>  
>  On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>>  
>>>  
>>> Henry - what sort of mechanism did you have in mind for these anonymous
>>> participants to identify themselves to one another? Are you talking
>>> about the exchange of credentials using the server as an intermediary in
>>> the exchange? Or did you have in mind an exchange outside the server.
>>> (E.g. Bob establishes a secure connection to Alice and over it tells her
>>> "I'm anon2 in conference foo right now"?
>>>  
>>  
>> 
>> The most credible e2e identification that comes to my mind is Phil
>> Zimmermann's Zfone technology:
>> 
>> http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>> http://zfoneproject.com/
>> 
>> Definitely not relying on any "trusted" server :-)
>> 
>> Thanks, Henry
>> 
>> On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>> <mailto:pkyzivat@alum.mit.edu>  wrote:
>> 
>>  
>>>  
>>> On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>>  
>>>>  
>>>>>  
>>>>> The participants are identified to
>>>>> each other by the central server
>>>>>  
>>>>  
>>>> 
>>>> There may be no contradiction, but just to make sure the scenario, such as
>>>> a
>>>> confidential business meeting is included (not necessarily a social network
>>>> scenario).
>>>> 
>>>> A confidential conference MUST assure that all participants can also be
>>>> safely identified first to each other, not necessarily by the server, end
>>>> to
>>>> end, to the full satisfaction of all the participants, who would otherwise
>>>> refuse to continue  to speak up or continue to attend at all.
>>>> Who provides for the e2e identification? A 3rd party? Inside the app?
>>>> 
>>>> Did I understand correctly there is no contradiction?
>>>> Any problem in clarifying this?
>>>>  
>>>  
>>> 
>>> This is an example of the sort of thing I was speaking of in my earlier
>>> reply - that there are differing levels of identity that may be desired.
>>> 
>>> In some cases, participants may want to be fully anonymous. But even
>>> then, it will typically be important that one participant be able to
>>> identify *which* of the other (anonymous) participants it is receiving
>>> media from. (E.g. by noting which participant in the roster is the
>>> current speaker - anon1, anon2, ...)
>>> 
>>> In other cases the participants *true* identities are hidden, but
>>> handles/nicknames may be used to identify them in the roster and in
>>> chats. ("snoopy" and "red barron" rather than anon1 and anon2.)
>>> 
>>> But in that case there are issues of how the handles are associated with
>>> participants. It could be that handles are permanently assigned to real
>>> users by the server. Then you can be confident that when talking to
>>> "snoopy" it is always someone well defined - same as the last time -
>>> without awareness of who that person *actually* is.
>>> 
>>> Alternatively, the server might allow handles to be used on a fcfs basis
>>> per conference. Or it might allow handles to be non-unique, so that you
>>> could have two "snoopy"s at the same time, bound to unrelated users.
>>> 
>>> Henry - what sort of mechanism did you have in mind for these anonymous
>>> participants to identify themselves to one another? Are you talking
>>> about the exchange of credentials using the server as an intermediary in
>>> the exchange? Or did you have in mind an exchange outside the server.
>>> (E.g. Bob establishes a secure connection to Alice and over it tells her
>>> "I'm anon2 in conference foo right now"?
>>> 
>>> Thanks,
>>> Paul
>>> 
>>>  
>>>>  
>>>> Thanks, Henry
>>>> 
>>>> 
>>>> On 8/10/11 9:16 AM, "Harald Alvestrand"<harald@alvestrand.no>
>>>> <mailto:harald@alvestrand.no>   wrote:
>>>> 
>>>>  
>>>>>  
>>>>> In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend
>>>>> one part of the scenario "4.3.3 Video conferencing system with central
>>>>> server".
>>>>> 
>>>>> I would like to add one more paragraph:
>>>>> 
>>>>> "All participant are authenticated by the central server, and authorized
>>>>> to connect to the central server. The participants are identified to
>>>>> each other by the central server, and the participants do not have
>>>>> access to each others' credentials such as e-mail addresses or login IDs".
>>>>> 
>>>>> This is necessary in order to drive use cases that resemble Google
>>>>> Hangout, where it is a requirement that people are able to participate
>>>>> without disclosing their Google login IDs to each other.
>>>>> (in the particular case of Hangout, the display name on their profile
>>>>> *is* disclosed ... but that's a different matter)
>>>>> 
>>>>> The reason I think this is important is that it feeds directly into the
>>>>> discussion of what WebRTC needs to authorize: The final source or
>>>>> destination of media, or the identity of the handler at the first hop.
>>>>> In at least the case of Hangouts, the requirement is to *not* authorize
>>>>> the final source or destination.
>>>>> 
>>>>> Not sure yet how to formulate that as a requirement, and not sure yet if
>>>>> it applies to the cases without a central server, such as 4.2.6. We may
>>>>> have to decide.
>>>>> 
>>>>>                           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
>>>> 
>>>>  
>>>  
>>> 
>>> _______________________________________________
>>> 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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<HTML>
<HEAD>
<TITLE>Re: [rtcweb] Use case change request: Identity in multiuser calls</T=
ITLE>
</HEAD>
<BODY>
<FONT SIZE=3D"2"><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D=
'font-size:12pt'>&gt; So, if we end up with a &quot;trusted MITM,&quot; why =
not have a trusted central server in the first place?<BR>
<BR>
draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support various o=
ther scenarios, but how I understand it, no PBX or any intermediary is requi=
red.<BR>
The default scenario is e2e.<BR>
<BR>
Maybe the authors copied here can clarify.<BR>
<BR>
Thanks, Henry<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'><BR>
<BR>
<BR>
On 8/11/11 2:39 PM, &quot;Igor Faynberg&quot; &lt;<a href=3D"igor.faynberg@al=
catel-lucent.com">igor.faynberg@alcatel-lucent.com</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> &nbsp;&nbsp;Well...<BR>
&nbsp;<BR>
&nbsp;(I thought that the lawful intercept issue that came up several times=
 is pertinent here.) &nbsp;But note that this was a parenthetical remark!<BR=
>
&nbsp;<BR>
&nbsp;A non-parenthetical one is based on the quote from the cited draft:<B=
R>
&nbsp;<BR>
&nbsp;</SPAN></FONT><FONT SIZE=3D"2"><FONT FACE=3D"Times New Roman"><SPAN STYLE=
=3D'font-size:12pt'> <BR>
&quot;In the development of ZRTP, it was realized that there are scenarios<=
BR>
&nbsp;&nbsp;&nbsp;in which the media can not be encrypted end-to-end. &nbsp=
;For example,<BR>
&nbsp;&nbsp;&nbsp;when a client has a trusted server or PBX which provides =
media<BR>
&nbsp;&nbsp;&nbsp;services in the path. &nbsp;For these cases, ZRTP develop=
ed mechanisms for<BR>
&nbsp;&nbsp;&nbsp;handling a &quot;trusted MiTM&quot; which can terminate t=
han reoriginate the<BR>
&nbsp;&nbsp;&nbsp;SRTP encryption.&quot; <BR>
&nbsp;<BR>
</SPAN></FONT></FONT><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN =
STYLE=3D'font-size:13pt'> <BR>
&nbsp;So, if we end up with a &quot;trusted MITM,&quot; why not have a trus=
ted central server in the first place?<BR>
&nbsp;<BR>
&nbsp;Igor<BR>
&nbsp;<BR>
&nbsp;P.S.: &nbsp;I have nothing against ZRTP, which has done a thorough an=
d honest job. &nbsp;I question its applicability to the agreed use cases. Pe=
rsonally I think we need an all-encompassing solution.<BR>
&nbsp;<BR>
&nbsp;<BR>
&nbsp;On 8/11/2011 3:11 PM, Henry Sinnreich wrote: <BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
Henry - what sort of mechanism did you have in mind for these anonymous<BR>
participants to identify themselves to one another? Are you talking<BR>
about the exchange of credentials using the server as an intermediary in<BR=
>
the exchange? Or did you have in mind an exchange outside the server.<BR>
(E.g. Bob establishes a secure connection to Alice and over it tells her<BR=
>
&quot;I'm anon2 in conference foo right now&quot;?<BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
The most credible e2e identification that comes to my mind is Phil<BR>
Zimmermann's Zfone technology:<BR>
<BR>
<a href=3D"http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt"=
>http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt</a><BR>
<a href=3D"http://zfoneproject.com/">http://zfoneproject.com/</a><BR>
<BR>
Definitely not relying on any &quot;trusted&quot; server :-)<BR>
<BR>
Thanks, Henry<BR>
<BR>
On 8/11/11 11:50 AM, &quot;Paul Kyzivat&quot; &lt;<a href=3D"pkyzivat@alum.mi=
t.edu">pkyzivat@alum.mit.edu</a>&gt; &lt;<a href=3D"mailto:pkyzivat@alum.mit.e=
du">mailto:pkyzivat@alum.mit.edu</a>&gt; &nbsp;wrote:<BR>
<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
On 8/11/11 12:07 PM, Henry Sinnreich wrote:<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
The participants are identified to<BR>
each other by the central server<BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
There may be no contradiction, but just to make sure the scenario, such as =
a<BR>
confidential business meeting is included (not necessarily a social network=
<BR>
scenario).<BR>
<BR>
A confidential conference MUST assure that all participants can also be<BR>
safely identified first to each other, not necessarily by the server, end t=
o<BR>
end, to the full satisfaction of all the participants, who would otherwise<=
BR>
refuse to continue &nbsp;to speak up or continue to attend at all.<BR>
Who provides for the e2e identification? A 3rd party? Inside the app?<BR>
<BR>
Did I understand correctly there is no contradiction?<BR>
Any problem in clarifying this?<BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
This is an example of the sort of thing I was speaking of in my earlier<BR>
reply - that there are differing levels of identity that may be desired.<BR=
>
<BR>
In some cases, participants may want to be fully anonymous. But even<BR>
then, it will typically be important that one participant be able to<BR>
identify *which* of the other (anonymous) participants it is receiving<BR>
media from. (E.g. by noting which participant in the roster is the<BR>
current speaker - anon1, anon2, ...)<BR>
<BR>
In other cases the participants *true* identities are hidden, but<BR>
handles/nicknames may be used to identify them in the roster and in<BR>
chats. (&quot;snoopy&quot; and &quot;red barron&quot; rather than anon1 and=
 anon2.)<BR>
<BR>
But in that case there are issues of how the handles are associated with<BR=
>
participants. It could be that handles are permanently assigned to real<BR>
users by the server. Then you can be confident that when talking to<BR>
&quot;snoopy&quot; it is always someone well defined - same as the last tim=
e -<BR>
without awareness of who that person *actually* is.<BR>
<BR>
Alternatively, the server might allow handles to be used on a fcfs basis<BR=
>
per conference. Or it might allow handles to be non-unique, so that you<BR>
could have two &quot;snoopy&quot;s at the same time, bound to unrelated use=
rs.<BR>
<BR>
Henry - what sort of mechanism did you have in mind for these anonymous<BR>
participants to identify themselves to one another? Are you talking<BR>
about the exchange of credentials using the server as an intermediary in<BR=
>
the exchange? Or did you have in mind an exchange outside the server.<BR>
(E.g. Bob establishes a secure connection to Alice and over it tells her<BR=
>
&quot;I'm anon2 in conference foo right now&quot;?<BR>
<BR>
Thanks,<BR>
Paul<BR>
<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
Thanks, Henry<BR>
<BR>
<BR>
On 8/10/11 9:16 AM, &quot;Harald Alvestrand&quot;&lt;<a href=3D"harald@alvest=
rand.no">harald@alvestrand.no</a>&gt; &lt;<a href=3D"mailto:harald@alvestrand.=
no">mailto:harald@alvestrand.no</a>&gt; &nbsp;&nbsp;wrote:<BR>
<BR>
&nbsp;<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> <BR>
In draft-ietf-rtcweb-use-cases-and-requirements, I would like to extend<BR>
one part of the scenario &quot;4.3.3 Video conferencing system with central=
<BR>
server&quot;.<BR>
<BR>
I would like to add one more paragraph:<BR>
<BR>
&quot;All participant are authenticated by the central server, and authoriz=
ed<BR>
to connect to the central server. The participants are identified to<BR>
each other by the central server, and the participants do not have<BR>
access to each others' credentials such as e-mail addresses or login IDs&qu=
ot;.<BR>
<BR>
This is necessary in order to drive use cases that resemble Google<BR>
Hangout, where it is a requirement that people are able to participate<BR>
without disclosing their Google login IDs to each other.<BR>
(in the particular case of Hangout, the display name on their profile<BR>
*is* disclosed ... but that's a different matter)<BR>
<BR>
The reason I think this is important is that it feeds directly into the<BR>
discussion of what WebRTC needs to authorize: The final source or<BR>
destination of media, or the identity of the handler at the first hop.<BR>
In at least the case of Hangouts, the requirement is to *not* authorize<BR>
the final source or destination.<BR>
<BR>
Not sure yet how to formulate that as a requirement, and not sure yet if<BR=
>
it applies to the cases without a central server, such as 4.2.6. We may<BR>
have to decide.<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;Harald<BR>
<BR>
<BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
<BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
<BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
&nbsp;<BR>
</SPAN></FONT></BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial">=
<SPAN STYLE=3D'font-size:13pt'> <BR>
<BR>
<BR>
_______________________________________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
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From randell-ietf@jesup.org  Thu Aug 11 14:18:24 2011
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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 8/11/2011 11:08 AM, Harald Alvestrand wrote:

> On 08/10/11 17:19, Randell Jesup wrote:
>> We should consider ways for a client to indicate how many simultaneous
>> decodes it's willing/able to handle.  This lets the server be smart about what it
>> sends (it may be as simple as rejecting video streams on re-negotiation of o/a
>> - and that may help handle things such as where the number of streams supported
>> varies with the type of encoding/resolution/frame-rate, or other client-side loads).
> If we do multiple video streams inside an RTP session, we don't need
> re-negotiation of o/a to add more streams. We might consider the
> proposed extensions in
> draft-westerlund-avtcore-multistream-and-simulcast to add information
> about number of streams to O/A (he suggests that for max backwards
> compatibility, we should assume that no more than one stream per RTP
> session can be handled unless the extension is used, but that might be
> an overly conservative suggestion).

My problem with that is that all streams are not equal.  I can handle a lot
more CIF streams than I can HD streams.  So while a suggestion is good, there
should be some way to say "too much", if not before a stream is added, then after
the fact (that might be good to have anyways).


>> For example, as people join a "Hangout", the server would re-negotiate
>> to add
>> a video/audio pair (using SDP grouping I assume).  If the client can't
>> handle
>> another one, it could reject the add (or renegotiate to lower the
>> complexity/size
>> of existing streams, perhaps).  The server might then send the N most
>> "interesting"
>> streams, or combine streams, etc.  SDP+o/a may not be optimal here but
>> I think it
>> would get the job done.
> Hangouts put all the video streams into one RTP session, and all the
> audio streams into another RTP session. Renegotiation would be needed if
> it didn't.
That assumes they all are using the same parameters/payloads (or ones agreed to
in the m= lines of the existing participants).  I'm not 100% sure that's a
valid assumption - and we need to be able to renegotiate at that point anyways
(new person doesn't support an optional codec being used, for example).  But on
the other hand maybe it's simpler to not try to handle all the funny edge cases
here and force/allow the clients to renegotiate if they don't like the state of
the call.

We should think through the case of multiple video sessions in a conference (example:
one for participants, one for presentations, and perhaps even a second camera for
some participants.)  I don't see any blocker here other than properly identifying which
is the "user" video and which is "presentation" and that may be a W3 issue or an
application issue even.


>> Note that this requires the central server to decrypt and re-encrypt the
>> streams (as I believe Harald's wording above would).
> Not necessarily - if key negotiation goes via the server, the server
> could make sure the key used in the server-to-client direction is the
> same key as that used in the client-to-server direction - but if you
> want to put 2 media streams into the same RTP session, and do keying per
> RTP session, the server would have to decrypt-then-encrypt.

Do we have enough control of the key selection (and the timing of it) to
do that?  Plus, I see that as an optimization - the server knows the key;
whether the keys are the same or different and if it actually decodes and
re-encodes are performance details (though perhaps important ones).

If the server doesn't know the key, we're in an end-to-end exchange scenario
where the server just acts as a forwarder, and the ends will authenticate each
incoming stream separately, and the server has much more limited ability to
interact with the conference.

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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Yes, definitely, this is a good course of action--to discuss it with the 
authors.

To this end, I will piggy-back another question.

Do I understand it right that an endpoint  would need to share a secret 
with every other endpoint it would need to authenticate or be 
authenticated by?  If not, how could DH exchange be authenticated (to 
avoid MiTM)?

Igor

On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
> > So, if we end up with a "trusted MITM," why not have a trusted 
> central server in the first place?
>
> draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support 
> various other scenarios, but how I understand it, no PBX or any 
> intermediary is required.
> The default scenario is e2e.
>
> Maybe the authors copied here can clarify.
>
> Thanks, Henry
>
>
>
> On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com> 
> wrote:
>
>       Well...
>
>      (I thought that the lawful intercept issue that came up several
>     times is pertinent here.)  But note that this was a parenthetical
>     remark!
>
>      A non-parenthetical one is based on the quote from the cited draft:
>
>
>     "In the development of ZRTP, it was realized that there are scenarios
>        in which the media can not be encrypted end-to-end.  For example,
>        when a client has a trusted server or PBX which provides media
>        services in the path.  For these cases, ZRTP developed
>     mechanisms for
>        handling a "trusted MiTM" which can terminate than reoriginate the
>        SRTP encryption."
>
>
>      So, if we end up with a "trusted MITM," why not have a trusted
>     central server in the first place?
>
>      Igor
>
>      P.S.:  I have nothing against ZRTP, which has done a thorough and
>     honest job.  I question its applicability to the agreed use cases.
>     Personally I think we need an all-encompassing solution.
>
>
>      On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>
>
>
>             Henry - what sort of mechanism did you have in mind for
>             these anonymous
>             participants to identify themselves to one another? Are
>             you talking
>             about the exchange of credentials using the server as an
>             intermediary in
>             the exchange? Or did you have in mind an exchange outside
>             the server.
>             (E.g. Bob establishes a secure connection to Alice and
>             over it tells her
>             "I'm anon2 in conference foo right now"?
>
>
>
>         The most credible e2e identification that comes to my mind is Phil
>         Zimmermann's Zfone technology:
>
>         http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>         http://zfoneproject.com/
>
>         Definitely not relying on any "trusted" server :-)
>
>         Thanks, Henry
>
>         On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>         <mailto:pkyzivat@alum.mit.edu>  wrote:
>
>
>
>             On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>
>
>
>                     The participants are identified to
>                     each other by the central server
>
>
>
>                 There may be no contradiction, but just to make sure
>                 the scenario, such as a
>                 confidential business meeting is included (not
>                 necessarily a social network
>                 scenario).
>
>                 A confidential conference MUST assure that all
>                 participants can also be
>                 safely identified first to each other, not necessarily
>                 by the server, end to
>                 end, to the full satisfaction of all the participants,
>                 who would otherwise
>                 refuse to continue  to speak up or continue to attend
>                 at all.
>                 Who provides for the e2e identification? A 3rd party?
>                 Inside the app?
>
>                 Did I understand correctly there is no contradiction?
>                 Any problem in clarifying this?
>
>
>
>             This is an example of the sort of thing I was speaking of
>             in my earlier
>             reply - that there are differing levels of identity that
>             may be desired.
>
>             In some cases, participants may want to be fully
>             anonymous. But even
>             then, it will typically be important that one participant
>             be able to
>             identify *which* of the other (anonymous) participants it
>             is receiving
>             media from. (E.g. by noting which participant in the
>             roster is the
>             current speaker - anon1, anon2, ...)
>
>             In other cases the participants *true* identities are
>             hidden, but
>             handles/nicknames may be used to identify them in the
>             roster and in
>             chats. ("snoopy" and "red barron" rather than anon1 and
>             anon2.)
>
>             But in that case there are issues of how the handles are
>             associated with
>             participants. It could be that handles are permanently
>             assigned to real
>             users by the server. Then you can be confident that when
>             talking to
>             "snoopy" it is always someone well defined - same as the
>             last time -
>             without awareness of who that person *actually* is.
>
>             Alternatively, the server might allow handles to be used
>             on a fcfs basis
>             per conference. Or it might allow handles to be
>             non-unique, so that you
>             could have two "snoopy"s at the same time, bound to
>             unrelated users.
>
>             Henry - what sort of mechanism did you have in mind for
>             these anonymous
>             participants to identify themselves to one another? Are
>             you talking
>             about the exchange of credentials using the server as an
>             intermediary in
>             the exchange? Or did you have in mind an exchange outside
>             the server.
>             (E.g. Bob establishes a secure connection to Alice and
>             over it tells her
>             "I'm anon2 in conference foo right now"?
>
>             Thanks,
>             Paul
>
>
>
>                 Thanks, Henry
>
>
>                 On 8/10/11 9:16 AM, "Harald
>                 Alvestrand"<harald@alvestrand.no>
>                 <mailto:harald@alvestrand.no>   wrote:
>
>
>
>                     In draft-ietf-rtcweb-use-cases-and-requirements, I
>                     would like to extend
>                     one part of the scenario "4.3.3 Video conferencing
>                     system with central
>                     server".
>
>                     I would like to add one more paragraph:
>
>                     "All participant are authenticated by the central
>                     server, and authorized
>                     to connect to the central server. The participants
>                     are identified to
>                     each other by the central server, and the
>                     participants do not have
>                     access to each others' credentials such as e-mail
>                     addresses or login IDs".
>
>                     This is necessary in order to drive use cases that
>                     resemble Google
>                     Hangout, where it is a requirement that people are
>                     able to participate
>                     without disclosing their Google login IDs to each
>                     other.
>                     (in the particular case of Hangout, the display
>                     name on their profile
>                     *is* disclosed ... but that's a different matter)
>
>                     The reason I think this is important is that it
>                     feeds directly into the
>                     discussion of what WebRTC needs to authorize: The
>                     final source or
>                     destination of media, or the identity of the
>                     handler at the first hop.
>                     In at least the case of Hangouts, the requirement
>                     is to *not* authorize
>                     the final source or destination.
>
>                     Not sure yet how to formulate that as a
>                     requirement, and not sure yet if
>                     it applies to the cases without a central server,
>                     such as 4.2.6. We may
>                     have to decide.
>
>                                               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
>
>
>
>
>             _______________________________________________
>             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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    Yes, definitely, this is a good course of action--to discuss it with
    the authors.<br>
    <br>
    To this end, I will piggy-back another question.<br>
    <br>
    Do I understand it right that an endpoint&nbsp; would need to share a
    secret with every other endpoint it would need to authenticate or be
    authenticated by?&nbsp; If not, how could DH exchange be authenticated
    (to avoid MiTM)?<br>
    <br>
    Igor<br>
    <br>
    On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
    <blockquote cite="mid:CA69AF47.1CBB0%25henry.sinnreich@gmail.com"
      type="cite">
      <title>Re: [rtcweb] Use case change request: Identity in multiuser
        calls</title>
      <font size="2"><font face="Calibri, Verdana, Helvetica, Arial"><span
            style="font-size: 12pt;">&gt; So, if we end up with a
            "trusted MITM," why not have a trusted central server in the
            first place?<br>
            <br>
            draft-johnston-rtcweb-media-privacy-00 says ZRTP can also
            support various other scenarios, but how I understand it, no
            PBX or any intermediary is required.<br>
            The default scenario is e2e.<br>
            <br>
            Maybe the authors copied here can clarify.<br>
            <br>
            Thanks, Henry<br>
          </span></font></font><font face="Calibri, Verdana, Helvetica,
        Arial"><span style="font-size: 13pt;"><br>
          <br>
          <br>
          On 8/11/11 2:39 PM, "Igor Faynberg" &lt;<a
            moz-do-not-send="true"
            href="igor.faynberg@alcatel-lucent.com">igor.faynberg@alcatel-lucent.com</a>&gt;
          wrote:<br>
          <br>
        </span></font>
      <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
            style="font-size: 13pt;"> &nbsp;&nbsp;Well...<br>
            &nbsp;<br>
            &nbsp;(I thought that the lawful intercept issue that came up
            several times is pertinent here.) &nbsp;But note that this was a
            parenthetical remark!<br>
            &nbsp;<br>
            &nbsp;A non-parenthetical one is based on the quote from the
            cited draft:<br>
            &nbsp;<br>
            &nbsp;</span></font><font size="2"><font face="Times New Roman"><span
              style="font-size: 12pt;"> <br>
              "In the development of ZRTP, it was realized that there
              are scenarios<br>
              &nbsp;&nbsp;&nbsp;in which the media can not be encrypted end-to-end.
              &nbsp;For example,<br>
              &nbsp;&nbsp;&nbsp;when a client has a trusted server or PBX which
              provides media<br>
              &nbsp;&nbsp;&nbsp;services in the path. &nbsp;For these cases, ZRTP developed
              mechanisms for<br>
              &nbsp;&nbsp;&nbsp;handling a "trusted MiTM" which can terminate than
              reoriginate the<br>
              &nbsp;&nbsp;&nbsp;SRTP encryption." <br>
              &nbsp;<br>
            </span></font></font><font face="Calibri, Verdana,
          Helvetica, Arial"><span style="font-size: 13pt;"> <br>
            &nbsp;So, if we end up with a "trusted MITM," why not have a
            trusted central server in the first place?<br>
            &nbsp;<br>
            &nbsp;Igor<br>
            &nbsp;<br>
            &nbsp;P.S.: &nbsp;I have nothing against ZRTP, which has done a
            thorough and honest job. &nbsp;I question its applicability to
            the agreed use cases. Personally I think we need an
            all-encompassing solution.<br>
            &nbsp;<br>
            &nbsp;<br>
            &nbsp;On 8/11/2011 3:11 PM, Henry Sinnreich wrote: <br>
          </span></font>
        <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
              style="font-size: 13pt;"> <br>
            </span></font>
          <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
                style="font-size: 13pt;"> <br>
                Henry - what sort of mechanism did you have in mind for
                these anonymous<br>
                participants to identify themselves to one another? Are
                you talking<br>
                about the exchange of credentials using the server as an
                intermediary in<br>
                the exchange? Or did you have in mind an exchange
                outside the server.<br>
                (E.g. Bob establishes a secure connection to Alice and
                over it tells her<br>
                "I'm anon2 in conference foo right now"?<br>
                &nbsp;<br>
              </span></font></blockquote>
          <font face="Calibri, Verdana, Helvetica, Arial"><span
              style="font-size: 13pt;"> <br>
              <br>
              The most credible e2e identification that comes to my mind
              is Phil<br>
              Zimmermann's Zfone technology:<br>
              <br>
              <a moz-do-not-send="true"
                href="http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt">http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt</a><br>
              <a moz-do-not-send="true" href="http://zfoneproject.com/">http://zfoneproject.com/</a><br>
              <br>
              Definitely not relying on any "trusted" server :-)<br>
              <br>
              Thanks, Henry<br>
              <br>
              On 8/11/11 11:50 AM, "Paul Kyzivat" &lt;<a
                moz-do-not-send="true" href="pkyzivat@alum.mit.edu">pkyzivat@alum.mit.edu</a>&gt;
              &lt;<a moz-do-not-send="true"
                href="mailto:pkyzivat@alum.mit.edu">mailto:pkyzivat@alum.mit.edu</a>&gt;
              &nbsp;wrote:<br>
              <br>
              &nbsp;<br>
            </span></font>
          <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
                style="font-size: 13pt;"> <br>
                On 8/11/11 12:07 PM, Henry Sinnreich wrote:<br>
                &nbsp;<br>
              </span></font>
            <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
                  style="font-size: 13pt;"> <br>
                </span></font>
              <blockquote><font face="Calibri, Verdana, Helvetica,
                  Arial"><span style="font-size: 13pt;"> <br>
                    The participants are identified to<br>
                    each other by the central server<br>
                    &nbsp;<br>
                  </span></font></blockquote>
              <font face="Calibri, Verdana, Helvetica, Arial"><span
                  style="font-size: 13pt;"> <br>
                  <br>
                  There may be no contradiction, but just to make sure
                  the scenario, such as a<br>
                  confidential business meeting is included (not
                  necessarily a social network<br>
                  scenario).<br>
                  <br>
                  A confidential conference MUST assure that all
                  participants can also be<br>
                  safely identified first to each other, not necessarily
                  by the server, end to<br>
                  end, to the full satisfaction of all the participants,
                  who would otherwise<br>
                  refuse to continue &nbsp;to speak up or continue to attend
                  at all.<br>
                  Who provides for the e2e identification? A 3rd party?
                  Inside the app?<br>
                  <br>
                  Did I understand correctly there is no contradiction?<br>
                  Any problem in clarifying this?<br>
                  &nbsp;<br>
                </span></font></blockquote>
            <font face="Calibri, Verdana, Helvetica, Arial"><span
                style="font-size: 13pt;"> <br>
                <br>
                This is an example of the sort of thing I was speaking
                of in my earlier<br>
                reply - that there are differing levels of identity that
                may be desired.<br>
                <br>
                In some cases, participants may want to be fully
                anonymous. But even<br>
                then, it will typically be important that one
                participant be able to<br>
                identify *which* of the other (anonymous) participants
                it is receiving<br>
                media from. (E.g. by noting which participant in the
                roster is the<br>
                current speaker - anon1, anon2, ...)<br>
                <br>
                In other cases the participants *true* identities are
                hidden, but<br>
                handles/nicknames may be used to identify them in the
                roster and in<br>
                chats. ("snoopy" and "red barron" rather than anon1 and
                anon2.)<br>
                <br>
                But in that case there are issues of how the handles are
                associated with<br>
                participants. It could be that handles are permanently
                assigned to real<br>
                users by the server. Then you can be confident that when
                talking to<br>
                "snoopy" it is always someone well defined - same as the
                last time -<br>
                without awareness of who that person *actually* is.<br>
                <br>
                Alternatively, the server might allow handles to be used
                on a fcfs basis<br>
                per conference. Or it might allow handles to be
                non-unique, so that you<br>
                could have two "snoopy"s at the same time, bound to
                unrelated users.<br>
                <br>
                Henry - what sort of mechanism did you have in mind for
                these anonymous<br>
                participants to identify themselves to one another? Are
                you talking<br>
                about the exchange of credentials using the server as an
                intermediary in<br>
                the exchange? Or did you have in mind an exchange
                outside the server.<br>
                (E.g. Bob establishes a secure connection to Alice and
                over it tells her<br>
                "I'm anon2 in conference foo right now"?<br>
                <br>
                Thanks,<br>
                Paul<br>
                <br>
                &nbsp;<br>
              </span></font>
            <blockquote><font face="Calibri, Verdana, Helvetica, Arial"><span
                  style="font-size: 13pt;"> <br>
                  Thanks, Henry<br>
                  <br>
                  <br>
                  On 8/10/11 9:16 AM, "Harald Alvestrand"&lt;<a
                    moz-do-not-send="true" href="harald@alvestrand.no">harald@alvestrand.no</a>&gt;
                  &lt;<a moz-do-not-send="true"
                    href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>&gt;
                  &nbsp;&nbsp;wrote:<br>
                  <br>
                  &nbsp;<br>
                </span></font>
              <blockquote><font face="Calibri, Verdana, Helvetica,
                  Arial"><span style="font-size: 13pt;"> <br>
                    In draft-ietf-rtcweb-use-cases-and-requirements, I
                    would like to extend<br>
                    one part of the scenario "4.3.3 Video conferencing
                    system with central<br>
                    server".<br>
                    <br>
                    I would like to add one more paragraph:<br>
                    <br>
                    "All participant are authenticated by the central
                    server, and authorized<br>
                    to connect to the central server. The participants
                    are identified to<br>
                    each other by the central server, and the
                    participants do not have<br>
                    access to each others' credentials such as e-mail
                    addresses or login IDs".<br>
                    <br>
                    This is necessary in order to drive use cases that
                    resemble Google<br>
                    Hangout, where it is a requirement that people are
                    able to participate<br>
                    without disclosing their Google login IDs to each
                    other.<br>
                    (in the particular case of Hangout, the display name
                    on their profile<br>
                    *is* disclosed ... but that's a different matter)<br>
                    <br>
                    The reason I think this is important is that it
                    feeds directly into the<br>
                    discussion of what WebRTC needs to authorize: The
                    final source or<br>
                    destination of media, or the identity of the handler
                    at the first hop.<br>
                    In at least the case of Hangouts, the requirement is
                    to *not* authorize<br>
                    the final source or destination.<br>
                    <br>
                    Not sure yet how to formulate that as a requirement,
                    and not sure yet if<br>
                    it applies to the cases without a central server,
                    such as 4.2.6. We may<br>
                    have to decide.<br>
                    <br>
                    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Harald<br>
                    <br>
                    <br>
                    _______________________________________________<br>
                    rtcweb mailing list<br>
                    <a moz-do-not-send="true" href="rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    <a moz-do-not-send="true"
                      href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                    &nbsp;<br>
                  </span></font></blockquote>
              <font face="Calibri, Verdana, Helvetica, Arial"><span
                  style="font-size: 13pt;"> <br>
                  <br>
                  <br>
                  _______________________________________________<br>
                  rtcweb mailing list<br>
                  <a moz-do-not-send="true" href="rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <a moz-do-not-send="true"
                    href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                  <br>
                  &nbsp;<br>
                </span></font></blockquote>
            <font face="Calibri, Verdana, Helvetica, Arial"><span
                style="font-size: 13pt;"> <br>
                <br>
                _______________________________________________<br>
                rtcweb mailing list<br>
                <a moz-do-not-send="true" href="rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <a moz-do-not-send="true"
                  href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
                &nbsp;<br>
              </span></font></blockquote>
          <font face="Calibri, Verdana, Helvetica, Arial"><span
              style="font-size: 13pt;"> <br>
              <br>
              <br>
              _______________________________________________<br>
              rtcweb mailing list<br>
              <a moz-do-not-send="true" href="rtcweb@ietf.org">rtcweb@ietf.org</a><br>
              <a moz-do-not-send="true"
                href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
              &nbsp;<br>
            </span></font></blockquote>
        <font face="Calibri, Verdana, Helvetica, Arial"><span
            style="font-size: 13pt;"> <br>
          </span></font></blockquote>
    </blockquote>
  </body>
</html>

--------------040007060300040702090708--

From pkyzivat@alum.mit.edu  Thu Aug 11 20:30:23 2011
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Cc: Phil Zimmermann <prz@mit.edu>, rtcweb@ietf.org
Subject: Re: [rtcweb] Use case change request: Identity in multiuser calls
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On 8/11/11 7:01 PM, Igor Faynberg wrote:
> Yes, definitely, this is a good course of action--to discuss it with the
> authors.
>
> To this end, I will piggy-back another question.
>
> Do I understand it right that an endpoint would need to share a secret
> with every other endpoint it would need to authenticate or be
> authenticated by? If not, how could DH exchange be authenticated (to
> avoid MiTM)?

Related question:

If the trusted server is a conference focus, and there is a single media 
stream between each participant and the focus, how is this secret 
exchange managed? each thing you send over that media stream is 
presumable mixed or otherwise distributed to all the other participants.

	Thanks,
	Paul

> Igor
>
> On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
>> > So, if we end up with a "trusted MITM," why not have a trusted
>> central server in the first place?
>>
>> draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support
>> various other scenarios, but how I understand it, no PBX or any
>> intermediary is required.
>> The default scenario is e2e.
>>
>> Maybe the authors copied here can clarify.
>>
>> Thanks, Henry
>>
>>
>>
>> On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com>
>> wrote:
>>
>>     Well...
>>
>>     (I thought that the lawful intercept issue that came up several
>>     times is pertinent here.) But note that this was a parenthetical
>>     remark!
>>
>>     A non-parenthetical one is based on the quote from the cited draft:
>>
>>
>>     "In the development of ZRTP, it was realized that there are scenarios
>>     in which the media can not be encrypted end-to-end. For example,
>>     when a client has a trusted server or PBX which provides media
>>     services in the path. For these cases, ZRTP developed mechanisms for
>>     handling a "trusted MiTM" which can terminate than reoriginate the
>>     SRTP encryption."
>>
>>
>>     So, if we end up with a "trusted MITM," why not have a trusted
>>     central server in the first place?
>>
>>     Igor
>>
>>     P.S.: I have nothing against ZRTP, which has done a thorough and
>>     honest job. I question its applicability to the agreed use cases.
>>     Personally I think we need an all-encompassing solution.
>>
>>
>>     On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>>
>>
>>
>>             Henry - what sort of mechanism did you have in mind for
>>             these anonymous
>>             participants to identify themselves to one another? Are
>>             you talking
>>             about the exchange of credentials using the server as an
>>             intermediary in
>>             the exchange? Or did you have in mind an exchange outside
>>             the server.
>>             (E.g. Bob establishes a secure connection to Alice and
>>             over it tells her
>>             "I'm anon2 in conference foo right now"?
>>
>>
>>
>>         The most credible e2e identification that comes to my mind is Phil
>>         Zimmermann's Zfone technology:
>>
>>         http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>>         http://zfoneproject.com/
>>
>>         Definitely not relying on any "trusted" server :-)
>>
>>         Thanks, Henry
>>
>>         On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>>         <mailto:pkyzivat@alum.mit.edu> wrote:
>>
>>
>>
>>             On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>
>>
>>
>>                     The participants are identified to
>>                     each other by the central server
>>
>>
>>
>>                 There may be no contradiction, but just to make sure
>>                 the scenario, such as a
>>                 confidential business meeting is included (not
>>                 necessarily a social network
>>                 scenario).
>>
>>                 A confidential conference MUST assure that all
>>                 participants can also be
>>                 safely identified first to each other, not necessarily
>>                 by the server, end to
>>                 end, to the full satisfaction of all the participants,
>>                 who would otherwise
>>                 refuse to continue to speak up or continue to attend
>>                 at all.
>>                 Who provides for the e2e identification? A 3rd party?
>>                 Inside the app?
>>
>>                 Did I understand correctly there is no contradiction?
>>                 Any problem in clarifying this?
>>
>>
>>
>>             This is an example of the sort of thing I was speaking of
>>             in my earlier
>>             reply - that there are differing levels of identity that
>>             may be desired.
>>
>>             In some cases, participants may want to be fully
>>             anonymous. But even
>>             then, it will typically be important that one participant
>>             be able to
>>             identify *which* of the other (anonymous) participants it
>>             is receiving
>>             media from. (E.g. by noting which participant in the
>>             roster is the
>>             current speaker - anon1, anon2, ...)
>>
>>             In other cases the participants *true* identities are
>>             hidden, but
>>             handles/nicknames may be used to identify them in the
>>             roster and in
>>             chats. ("snoopy" and "red barron" rather than anon1 and
>>             anon2.)
>>
>>             But in that case there are issues of how the handles are
>>             associated with
>>             participants. It could be that handles are permanently
>>             assigned to real
>>             users by the server. Then you can be confident that when
>>             talking to
>>             "snoopy" it is always someone well defined - same as the
>>             last time -
>>             without awareness of who that person *actually* is.
>>
>>             Alternatively, the server might allow handles to be used
>>             on a fcfs basis
>>             per conference. Or it might allow handles to be
>>             non-unique, so that you
>>             could have two "snoopy"s at the same time, bound to
>>             unrelated users.
>>
>>             Henry - what sort of mechanism did you have in mind for
>>             these anonymous
>>             participants to identify themselves to one another? Are
>>             you talking
>>             about the exchange of credentials using the server as an
>>             intermediary in
>>             the exchange? Or did you have in mind an exchange outside
>>             the server.
>>             (E.g. Bob establishes a secure connection to Alice and
>>             over it tells her
>>             "I'm anon2 in conference foo right now"?
>>
>>             Thanks,
>>             Paul
>>
>>
>>
>>                 Thanks, Henry
>>
>>
>>                 On 8/10/11 9:16 AM, "Harald
>>                 Alvestrand"<harald@alvestrand.no>
>>                 <mailto:harald@alvestrand.no> wrote:
>>
>>
>>
>>                     In draft-ietf-rtcweb-use-cases-and-requirements, I
>>                     would like to extend
>>                     one part of the scenario "4.3.3 Video conferencing
>>                     system with central
>>                     server".
>>
>>                     I would like to add one more paragraph:
>>
>>                     "All participant are authenticated by the central
>>                     server, and authorized
>>                     to connect to the central server. The participants
>>                     are identified to
>>                     each other by the central server, and the
>>                     participants do not have
>>                     access to each others' credentials such as e-mail
>>                     addresses or login IDs".
>>
>>                     This is necessary in order to drive use cases that
>>                     resemble Google
>>                     Hangout, where it is a requirement that people are
>>                     able to participate
>>                     without disclosing their Google login IDs to each
>>                     other.
>>                     (in the particular case of Hangout, the display
>>                     name on their profile
>>                     *is* disclosed ... but that's a different matter)
>>
>>                     The reason I think this is important is that it
>>                     feeds directly into the
>>                     discussion of what WebRTC needs to authorize: The
>>                     final source or
>>                     destination of media, or the identity of the
>>                     handler at the first hop.
>>                     In at least the case of Hangouts, the requirement
>>                     is to *not* authorize
>>                     the final source or destination.
>>
>>                     Not sure yet how to formulate that as a
>>                     requirement, and not sure yet if
>>                     it applies to the cases without a central server,
>>                     such as 4.2.6. We may
>>                     have to decide.
>>
>>                     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
>>
>>
>>
>>
>>             _______________________________________________
>>             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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Thanks, Harald, for this submission.

A nit (I think).  Unless I'm misunderstanding the purpose of the =
"a=3Dgroup:TOGETHER" attribute, the port numbers in the two example =
'answers' are wrong.

Example 1:
  Answer, from an entity that understands TOGETHER

         v=3D0
         o=3Dbob 2808844564 2808844564 IN IP4 host.biloxi.example.com
         s=3D
         c=3DIN IP4 host.biloxi.example.com
         t=3D0 0
         a=3Dgroup:TOGETHER foo bar
         m=3Daudio 49174 RTP/AVP 0
         a=3Dmid:foo
         b=3DAS:200
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         b=3DAS:1000
         a=3Drtpmap:32 MPV/90000
Should be (I think):
         ...
         a=3Dgroup:TOGETHER foo bar
         m=3Daudio 49170 RTP/AVP 0
         a=3Dmid:foo
         b=3DAS:200
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         b=3DAS:1000
         a=3Drtpmap:32 MPV/90000

Example 2:
   Answer, from an entity that understands grouping, but does not
   understand TOGETHER

         v=3D0
         o=3Dbob 2808844564 2808844564 IN IP4 host.biloxi.example.com
         s=3D
         c=3DIN IP4 host.biloxi.example.com
         t=3D0 0
         m=3Daudio 49174 RTP/AVP 0
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         b=3DAS:200
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         a=3Drtpmap:32 MPV/90000
         b=3DAS:1000
Should be (I think):
         ...
         m=3Daudio 49170 RTP/AVP 0
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         b=3DAS:200
         m=3Dvideo 51372 RTP/AVP 32
         a=3Dmid:bar
         a=3Drtpmap:32 MPV/90000
         b=3DAS:1000


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_3FB97A1A-E3EA-4156-904D-EFCB7F339470
Content-Transfer-Encoding: quoted-printable
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	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Thanks, Harald, for this submission.<div><br></div><div>A nit (I =
think). &nbsp;Unless I'm misunderstanding the purpose of the =
"a=3Dgroup:TOGETHER" attribute, the port numbers in the two example =
'answers' are wrong.</div><div><br></div><div>Example 1:</div><div><pre> =
 Answer, from an entity that understands TOGETHER

         v=3D0
         o=3Dbob 2808844564 2808844564 IN IP4 <a =
href=3D"http://host.biloxi.example.com">host.biloxi.example.com</a>
         s=3D
         c=3DIN IP4 <a =
href=3D"http://host.biloxi.example.com">host.biloxi.example.com</a>
         t=3D0 0
         a=3Dgroup:TOGETHER foo bar
         m=3Daudio 49174 RTP/AVP 0
         a=3Dmid:foo
         b=3DAS:200
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         b=3DAS:1000
         a=3Drtpmap:32 MPV/90000
</pre><pre><font class=3D"Apple-style-span" face=3D"Helvetica"><span =
class=3D"Apple-style-span" style=3D"white-space: normal;">Should be (I =
think):</span></font></pre><pre><span class=3D"Apple-style-span" =
style=3D"font-family: Helvetica; white-space: normal; "><pre>         =
...
         a=3Dgroup:TOGETHER foo bar
         m=3Daudio 49170 RTP/AVP 0
         a=3Dmid:foo
         b=3DAS:200
         a=3Drtpmap:0 PCMU/8000
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         b=3DAS:1000
         a=3Drtpmap:32 MPV/90000
</pre><div><br></div></span><font class=3D"Apple-style-span" =
face=3D"Helvetica"><div><div style=3D"white-space: normal; ">Example =
2:</div><div style=3D"white-space: normal; "><pre>   Answer, from an =
entity that understands grouping, but does not
   understand TOGETHER

         v=3D0
         o=3Dbob 2808844564 2808844564 IN IP4 <a =
href=3D"http://host.biloxi.example.com">host.biloxi.example.com</a>
         s=3D
         c=3DIN IP4 <a =
href=3D"http://host.biloxi.example.com">host.biloxi.example.com</a>
         t=3D0 0
         m=3Daudio 49174 RTP/AVP 0
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         b=3DAS:200
         m=3Dvideo 49170 RTP/AVP 32
         a=3Dmid:bar
         a=3Drtpmap:32 MPV/90000
         b=3DAS:1000
</pre><div>Should be (I think):</div></div></div></font><span =
class=3D"Apple-style-span" style=3D"font-family: Helvetica; white-space: =
normal; "><pre>         ...
         m=3Daudio 49170 RTP/AVP 0
         a=3Dmid:foo
         a=3Drtpmap:0 PCMU/8000
         b=3DAS:200
         m=3Dvideo 51372 RTP/AVP 32
         a=3Dmid:bar
         a=3Drtpmap:32 MPV/90000
         b=3DAS:1000</pre><div><br></div></span><font =
class=3D"Apple-style-span" face=3D"Helvetica"><div><div =
style=3D"white-space: normal; "></div></div></font></pre></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); 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; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); 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; ">Ross =
Finlayson<br>Live Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

--Apple-Mail=_3FB97A1A-E3EA-4156-904D-EFCB7F339470--

From harald@alvestrand.no  Thu Aug 11 23:45:12 2011
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On 08/11/11 20:47, Paul Kyzivat wrote:
> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>> I have taken some of the information I learned from the discussions in
>> Quebec City about the issues of putting video and audio into the same
>> RTP session and created a draft from them outlining a solution to the
>> problem of signalling this configuration using SDP.
>>
>> Comments welcome, including the recommended context for processing the
>> document; in a few days I'll send the same note to AVTCORE and/or 
>> MMUSIC.
>
> Harald,
>
> A few questions/comments:
>
> 1) If ICE is being used, how do you expect it to be handled on the 2nd 
> m-line? Since one of the goals of multiplexing on a single rtp session 
> is to minimize ICE processing, one would hope only to do ICE on the 
> first of the grouped m-lines. But if the answerer doesn't support this 
> grouping, then the ICE will be needed on the others.
I tried to be explicit that if negotiation is successful, only the ICE 
parameters from the first section listed in the A=group:TOGETHER would 
be used:

    The following parameters are taken from the first section only:

    o  Port number from the m= line

    o  All media-level attributes defined in RFC 5245 section 15.1 - this
       includes "candidate", "remote-candidates", "ice-mismatch", "ice-
       ufrag", "ice-pwd"

Suggestions for improving this language?

> 2) The draft only shows this mechanism being used to combine one audio 
> and one video. Do you also imagine it being used to represent multiple 
> audio and/or multiple video streams that might otherwise be handled 
> independently? (I see no reason why it couldn't be used that way.) 
> This would allow description of different properties for each one.
I certainly think that any number of m= sections could be combined this 
way, but in other discussions, I've seen people tend towards thinking of 
sections as "one m= section for all the audio channels, one m= section 
for all the video channels". Can you give an example of the usage you're 
thinking of?

I'm worried that we may get too many special rules about how parameters 
from sections are to be combined - multiple codecs with different PT 
numbers are fine, but other parameters could be confusing, so I'd like 
to see examples.
>
> 3) In the example, for an entity that doesn't understand TOGETHER, you 
> still show the a=mid lines. This is appropriate for an answerer that 
> *does* understand a=group. However an answerer that doesn't implement 
> rfc3388 would presumably omit the a=mid lines from the answer. So the 
> offerer had better be prepared for that as well.
Yes, that's a reasonable point. I'll add a third example answer.
>
>     Thanks,
>     Paul
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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On 08/12/11 05:37, Ross Finlayson wrote:
> Thanks, Harald, for this submission.
>
> A nit (I think).  Unless I'm misunderstanding the purpose of the 
> "a=group:TOGETHER" attribute, the port numbers in the two example 
> 'answers' are wrong.
I'm not sure. These examples came straight out of RFC 4317, including 
the port numbers; I noted the port number mismatch, and concluded 
(tentatively) that the port number of an offer means "I'll be using this 
port", while the port number of an answer means "I'll be using this 
port" - which means that they SHOULD be expected to be different.

When using ICE, the port number is moot anyway (the "candidates" 
overrides them), so the only purpose of the port number is that the 
special value zero in an answer means "offer not accepted".
> Example 1:
>    Answer, from an entity that understands TOGETHER
>
>           v=0
>           o=bob 2808844564 2808844564 IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>           s=
>           c=IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>           t=0 0
>           a=group:TOGETHER foo bar
>           m=audio 49174 RTP/AVP 0
>           a=mid:foo
>           b=AS:200
>           a=rtpmap:0 PCMU/8000
>           m=video 49170 RTP/AVP 32
>           a=mid:bar
>           b=AS:1000
>           a=rtpmap:32 MPV/90000
> Should be (I think):
>           ...
>           a=group:TOGETHER foo bar
>           m=audio 49170 RTP/AVP 0
>           a=mid:foo
>           b=AS:200
>           a=rtpmap:0 PCMU/8000
>           m=video 49170 RTP/AVP 32
>           a=mid:bar
>           b=AS:1000
>           a=rtpmap:32 MPV/90000
> Example 2:
>     Answer, from an entity that understands grouping, but does not
>     understand TOGETHER
>
>           v=0
>           o=bob 2808844564 2808844564 IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>           s=
>           c=IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>           t=0 0
>           m=audio 49174 RTP/AVP 0
>           a=mid:foo
>           a=rtpmap:0 PCMU/8000
>           b=AS:200
>           m=video 49170 RTP/AVP 32
>           a=mid:bar
>           a=rtpmap:32 MPV/90000
>           b=AS:1000
> Should be (I think):
>           ...
>           m=audio 49170 RTP/AVP 0
>           a=mid:foo
>           a=rtpmap:0 PCMU/8000
>           b=AS:200
>           m=video 51372 RTP/AVP 32
>           a=mid:bar
>           a=rtpmap:32 MPV/90000
>           b=AS:1000
>
> Ross Finlayson
> Live Networks, Inc.
> http://www.live555.com/
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    On 08/12/11 05:37, Ross Finlayson wrote:
    <blockquote
      cite="mid:E17059F7-DF14-4552-8D01-609D3E4BB77C@live555.com"
      type="cite">Thanks, Harald, for this submission.
      <div><br>
      </div>
      <div>A nit (I think). &nbsp;Unless I'm misunderstanding the purpose of
        the "a=group:TOGETHER" attribute, the port numbers in the two
        example 'answers' are wrong.</div>
    </blockquote>
    I'm not sure. These examples came straight out of RFC 4317,
    including the port numbers; I noted the port number mismatch, and
    concluded (tentatively) that the port number of an offer means "I'll
    be using this port", while the port number of an answer means "I'll
    be using this port" - which means that they SHOULD be expected to be
    different.<br>
    <br>
    When using ICE, the port number is moot anyway (the "candidates"
    overrides them), so the only purpose of the port number is that the
    special value zero in an answer means "offer not accepted".<br>
    <blockquote
      cite="mid:E17059F7-DF14-4552-8D01-609D3E4BB77C@live555.com"
      type="cite">
      <div>Example 1:</div>
      <div>
        <pre>  Answer, from an entity that understands TOGETHER

         v=0
         o=bob 2808844564 2808844564 IN IP4 <a moz-do-not-send="true" href="http://host.biloxi.example.com">host.biloxi.example.com</a>
         s=
         c=IN IP4 <a moz-do-not-send="true" href="http://host.biloxi.example.com">host.biloxi.example.com</a>
         t=0 0
         a=group:TOGETHER foo bar
         m=audio 49174 RTP/AVP 0
         a=mid:foo
         b=AS:200
         a=rtpmap:0 PCMU/8000
         m=video 49170 RTP/AVP 32
         a=mid:bar
         b=AS:1000
         a=rtpmap:32 MPV/90000
</pre>
        <pre><font class="Apple-style-span" face="Helvetica"><span class="Apple-style-span" style="white-space: normal;">Should be (I think):</span></font></pre>
        <pre><span class="Apple-style-span" style="font-family: Helvetica; white-space: normal;"><pre>         ...
         a=group:TOGETHER foo bar
         m=audio 49170 RTP/AVP 0
         a=mid:foo
         b=AS:200
         a=rtpmap:0 PCMU/8000
         m=video 49170 RTP/AVP 32
         a=mid:bar
         b=AS:1000
         a=rtpmap:32 MPV/90000
</pre><div>
</div></span><font class="Apple-style-span" face="Helvetica"><div><div style="white-space: normal;">Example 2:</div><div style="white-space: normal;"><pre>   Answer, from an entity that understands grouping, but does not
   understand TOGETHER

         v=0
         o=bob 2808844564 2808844564 IN IP4 <a moz-do-not-send="true" href="http://host.biloxi.example.com">host.biloxi.example.com</a>
         s=
         c=IN IP4 <a moz-do-not-send="true" href="http://host.biloxi.example.com">host.biloxi.example.com</a>
         t=0 0
         m=audio 49174 RTP/AVP 0
         a=mid:foo
         a=rtpmap:0 PCMU/8000
         b=AS:200
         m=video 49170 RTP/AVP 32
         a=mid:bar
         a=rtpmap:32 MPV/90000
         b=AS:1000
</pre><div>Should be (I think):</div></div></div></font><span class="Apple-style-span" style="font-family: Helvetica; white-space: normal;"><pre>         ...
         m=audio 49170 RTP/AVP 0
         a=mid:foo
         a=rtpmap:0 PCMU/8000
         b=AS:200
         m=video 51372 RTP/AVP 32
         a=mid:bar
         a=rtpmap:32 MPV/90000
         b=AS:1000</pre><div>
</div></span></pre>
      </div>
      <br>
      <div>
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            letter-spacing: normal; line-height: normal; orphans: 2;
            text-indent: 0px; text-transform: none; white-space: normal;
            widows: 2; word-spacing: 0px; font-size: medium;">Ross
            Finlayson<br>
            Live Networks, Inc.<br>
            <a moz-do-not-send="true" href="http://www.live555.com/">http://www.live555.com/</a></span></span>
      </div>
      <br>
      <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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From harald@alvestrand.no  Fri Aug 12 04:30:43 2011
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From: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org, Phil Zimmermann <prz@mit.edu>
Subject: [rtcweb] Exchange of trusted identities in server-based conferencing (Re: Use case change request: Identity in multiuser calls)
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[[ Reminder to all: If you discover that you're talking about something 
far away from the original topic of the thread, please CHANGE THE 
SUBJECT LINE! ]]

The normal method of authentication for teleconferences today is that 
people prove they have the conference number (by calling in), and 
recognize each others' voices.
We have security procedures at Google that depend on people recognizing 
each others' faces (sometimes based on prior meetings, sometimes based 
on matching with pictures).

The digital realm doesn't necessarily have to provide a steel vault 
where the analog realm is satisfied with plywood - or at least, I think 
it is not the best for finishing RTCWEB in a reasonable time if we 
insist that steel vaults be part of the requirements.

                   Harald

On 08/12/11 05:30, Paul Kyzivat wrote:
> On 8/11/11 7:01 PM, Igor Faynberg wrote:
>> Yes, definitely, this is a good course of action--to discuss it with the
>> authors.
>>
>> To this end, I will piggy-back another question.
>>
>> Do I understand it right that an endpoint would need to share a secret
>> with every other endpoint it would need to authenticate or be
>> authenticated by? If not, how could DH exchange be authenticated (to
>> avoid MiTM)?
>
> Related question:
>
> If the trusted server is a conference focus, and there is a single 
> media stream between each participant and the focus, how is this 
> secret exchange managed? each thing you send over that media stream is 
> presumable mixed or otherwise distributed to all the other participants.
>
>     Thanks,
>     Paul
>
>> Igor
>>
>> On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
>>> > So, if we end up with a "trusted MITM," why not have a trusted
>>> central server in the first place?
>>>
>>> draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support
>>> various other scenarios, but how I understand it, no PBX or any
>>> intermediary is required.
>>> The default scenario is e2e.
>>>
>>> Maybe the authors copied here can clarify.
>>>
>>> Thanks, Henry
>>>
>>>
>>>
>>> On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com>
>>> wrote:
>>>
>>>     Well...
>>>
>>>     (I thought that the lawful intercept issue that came up several
>>>     times is pertinent here.) But note that this was a parenthetical
>>>     remark!
>>>
>>>     A non-parenthetical one is based on the quote from the cited draft:
>>>
>>>
>>>     "In the development of ZRTP, it was realized that there are 
>>> scenarios
>>>     in which the media can not be encrypted end-to-end. For example,
>>>     when a client has a trusted server or PBX which provides media
>>>     services in the path. For these cases, ZRTP developed mechanisms 
>>> for
>>>     handling a "trusted MiTM" which can terminate than reoriginate the
>>>     SRTP encryption."
>>>
>>>
>>>     So, if we end up with a "trusted MITM," why not have a trusted
>>>     central server in the first place?
>>>
>>>     Igor
>>>
>>>     P.S.: I have nothing against ZRTP, which has done a thorough and
>>>     honest job. I question its applicability to the agreed use cases.
>>>     Personally I think we need an all-encompassing solution.
>>>
>>>
>>>     On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>>>
>>>
>>>
>>>             Henry - what sort of mechanism did you have in mind for
>>>             these anonymous
>>>             participants to identify themselves to one another? Are
>>>             you talking
>>>             about the exchange of credentials using the server as an
>>>             intermediary in
>>>             the exchange? Or did you have in mind an exchange outside
>>>             the server.
>>>             (E.g. Bob establishes a secure connection to Alice and
>>>             over it tells her
>>>             "I'm anon2 in conference foo right now"?
>>>
>>>
>>>
>>>         The most credible e2e identification that comes to my mind 
>>> is Phil
>>>         Zimmermann's Zfone technology:
>>>
>>>         
>>> http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>>>         http://zfoneproject.com/
>>>
>>>         Definitely not relying on any "trusted" server :-)
>>>
>>>         Thanks, Henry
>>>
>>>         On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>>> <mailto:pkyzivat@alum.mit.edu> wrote:
>>>
>>>
>>>
>>>             On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>>
>>>
>>>
>>>                     The participants are identified to
>>>                     each other by the central server
>>>
>>>
>>>
>>>                 There may be no contradiction, but just to make sure
>>>                 the scenario, such as a
>>>                 confidential business meeting is included (not
>>>                 necessarily a social network
>>>                 scenario).
>>>
>>>                 A confidential conference MUST assure that all
>>>                 participants can also be
>>>                 safely identified first to each other, not necessarily
>>>                 by the server, end to
>>>                 end, to the full satisfaction of all the participants,
>>>                 who would otherwise
>>>                 refuse to continue to speak up or continue to attend
>>>                 at all.
>>>                 Who provides for the e2e identification? A 3rd party?
>>>                 Inside the app?
>>>
>>>                 Did I understand correctly there is no contradiction?
>>>                 Any problem in clarifying this?
>>>
>>>
>>>
>>>             This is an example of the sort of thing I was speaking of
>>>             in my earlier
>>>             reply - that there are differing levels of identity that
>>>             may be desired.
>>>
>>>             In some cases, participants may want to be fully
>>>             anonymous. But even
>>>             then, it will typically be important that one participant
>>>             be able to
>>>             identify *which* of the other (anonymous) participants it
>>>             is receiving
>>>             media from. (E.g. by noting which participant in the
>>>             roster is the
>>>             current speaker - anon1, anon2, ...)
>>>
>>>             In other cases the participants *true* identities are
>>>             hidden, but
>>>             handles/nicknames may be used to identify them in the
>>>             roster and in
>>>             chats. ("snoopy" and "red barron" rather than anon1 and
>>>             anon2.)
>>>
>>>             But in that case there are issues of how the handles are
>>>             associated with
>>>             participants. It could be that handles are permanently
>>>             assigned to real
>>>             users by the server. Then you can be confident that when
>>>             talking to
>>>             "snoopy" it is always someone well defined - same as the
>>>             last time -
>>>             without awareness of who that person *actually* is.
>>>
>>>             Alternatively, the server might allow handles to be used
>>>             on a fcfs basis
>>>             per conference. Or it might allow handles to be
>>>             non-unique, so that you
>>>             could have two "snoopy"s at the same time, bound to
>>>             unrelated users.
>>>
>>>             Henry - what sort of mechanism did you have in mind for
>>>             these anonymous
>>>             participants to identify themselves to one another? Are
>>>             you talking
>>>             about the exchange of credentials using the server as an
>>>             intermediary in
>>>             the exchange? Or did you have in mind an exchange outside
>>>             the server.
>>>             (E.g. Bob establishes a secure connection to Alice and
>>>             over it tells her
>>>             "I'm anon2 in conference foo right now"?
>>>
>>>             Thanks,
>>>             Paul
>>>
>>>
>>>
>>>                 Thanks, Henry
>>>
>>>
>>>                 On 8/10/11 9:16 AM, "Harald
>>>                 Alvestrand"<harald@alvestrand.no>
>>> <mailto:harald@alvestrand.no> wrote:
>>>
>>>
>>>
>>>                     In draft-ietf-rtcweb-use-cases-and-requirements, I
>>>                     would like to extend
>>>                     one part of the scenario "4.3.3 Video conferencing
>>>                     system with central
>>>                     server".
>>>
>>>                     I would like to add one more paragraph:
>>>
>>>                     "All participant are authenticated by the central
>>>                     server, and authorized
>>>                     to connect to the central server. The participants
>>>                     are identified to
>>>                     each other by the central server, and the
>>>                     participants do not have
>>>                     access to each others' credentials such as e-mail
>>>                     addresses or login IDs".
>>>
>>>                     This is necessary in order to drive use cases that
>>>                     resemble Google
>>>                     Hangout, where it is a requirement that people are
>>>                     able to participate
>>>                     without disclosing their Google login IDs to each
>>>                     other.
>>>                     (in the particular case of Hangout, the display
>>>                     name on their profile
>>>                     *is* disclosed ... but that's a different matter)
>>>
>>>                     The reason I think this is important is that it
>>>                     feeds directly into the
>>>                     discussion of what WebRTC needs to authorize: The
>>>                     final source or
>>>                     destination of media, or the identity of the
>>>                     handler at the first hop.
>>>                     In at least the case of Hangouts, the requirement
>>>                     is to *not* authorize
>>>                     the final source or destination.
>>>
>>>                     Not sure yet how to formulate that as a
>>>                     requirement, and not sure yet if
>>>                     it applies to the cases without a central server,
>>>                     such as 4.2.6. We may
>>>                     have to decide.
>>>
>>>                     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
>>>
>>>
>>>
>>>
>>>             _______________________________________________
>>>             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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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Harald Alvestrand" <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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A few comments:
=20
1) Section 4 has this:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
session, and the parameters for the other sections MUST be ignored.
=20
I think it needs to be:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
and RCP session, and the parameters for the other sections MUST be=20
ignored.
=20
2) When the RTCP port is not the next higher (odd) port number following
the RTP port described in the m-line, the "a=3Drtcp" attribute (defined =
in
RFC-3605) is used to signal it, as in:
m=3Daudio 49170 RTP/AVP 0
a=3Drtcp:53020 IN IP4 126.16.64.4
=20
With the grouping semantics described in the draft, it seems this RTCP
port should be taken only from the first section. So, this could be
could be added to section 4.
=20
3) RTCP could be multiplexed with RTP on a single port and signaled
using the "a=3Drtcp-mux" attribute (defined in RFC 5761). When it is =
used
together with the grouping semantics described in the draft, it seems
all RTCP sessions of that group need to be multiplexed on the RTP port
from the first section.
=20
Muthu
=20
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Harald Alvestrand
Sent: Thursday, August 11, 2011 5:02 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
I have taken some of the information I learned from the discussions in
Quebec City about the issues of putting video and audio into the same
RTP session and created a draft from them outlining a solution to the
problem of signalling this configuration using SDP.

Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or
MMUSIC.

                      Harald

-------- Original Message --------=20
Subject:=20
I-D Action: draft-alvestrand-one-rtp-00.txt
Date:=20
Thu, 11 Aug 2011 04:28:09 -0700
From:=20
internet-drafts@ietf.org
Reply-To:=20
internet-drafts@ietf.org
To:=20
i-d-announce@ietf.org
=20
A New Internet-Draft is available from the on-line Internet-Drafts
directories.
=20
       Title           : SDP Grouping for Single RTP Sessions
       Author(s)       : Harald Tveit Alvestrand
       Filename        : draft-alvestrand-one-rtp-00.txt
       Pages           : 8
       Date            : 2011-08-11
=20
   This document describes an extension to the Session Description
   Protocol (SDP) to describe RTP sessions where media of multiple top
   level types, for example audio and video, are carried in the same RTP
   session.
=20
   This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
   consideration.
=20
=20
=20
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
=20
Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/
=20
This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
_______________________________________________
I-D-Announce mailing list
I-D-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/i-d-announce
Internet-Draft directories: http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
=20

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</head>

<body bgcolor=3Dwhite lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:.5in'>

<div class=3DWordSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>A
few comments:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>1)
Section 4 has this:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>If
the responder understands the semantics of the TOGETHER =
extension,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>the
parameters of the first section MUST be used to establish the <span
class=3DSpellE>RTP</span><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>session,
and the parameters for the other sections MUST be =
ignored.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>I
think it needs to be:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>If
the responder understands the semantics of the TOGETHER =
extension,<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>the
parameters of the first section MUST be used to establish the <span
class=3DSpellE>RTP</span><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>and
<span class=3DSpellE>RCP</span> session, and the parameters for the =
other
sections MUST be <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>ignored.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>2)
When the <span class=3DSpellE>RTCP</span> port is not the next higher =
(odd) port
number following the <span class=3DSpellE>RTP</span> port described in =
the m-line,
the &quot;a=3D<span class=3DSpellE>rtcp</span>&quot; attribute (defined =
in <span
class=3DSpellE>RFC</span>-3605) is used to signal it, as =
in:<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>m=3Daudio
49170 <span class=3DSpellE>RTP</span>/<span class=3DSpellE>AVP</span> =
0<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>a=3D<span
class=3DSpellE>rtcp:53020</span> IN <span class=3DSpellE>IP4</span> =
126.16.64.4<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>With
the grouping semantics described in the draft, it seems this <span
class=3DSpellE>RTCP</span> port should be taken only from the first =
section. So,
this could be could be added to section 4.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>3)
<span class=3DSpellE>RTCP</span> could be multiplexed with <span =
class=3DSpellE>RTP</span>
on a single port and signaled using the &quot;a=3D<span =
class=3DSpellE>rtcp-mux</span>&quot;
attribute (defined in <span class=3DSpellE>RFC</span> 5761). When it is =
used
together with the grouping semantics described in the draft, it seems =
all <span
class=3DSpellE>RTCP</span> sessions of that group need to be multiplexed =
on the <span
class=3DSpellE>RTP</span> port from the first =
section.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>Muthu<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><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=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
mso-fareast-font-family:"Times New =
Roman";color:windowtext'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";mso-fareast-f=
ont-family:
"Times New Roman";color:windowtext'> rtcweb-bounces@ietf.org
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Harald =
Alvestrand<br>
<b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>I
have taken some of the information I learned from the discussions in =
Quebec
City about the issues of putting video and audio into the same RTP =
session and
created a draft from them outlining a solution to the problem of =
signalling
this configuration using SDP.<br>
<br>
Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or =
MMUSIC.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Harald<br>
<br>
-------- Original Message -------- <o:p></o:p></span></p>

<table class=3DMsoNormalTable border=3D0 cellspacing=3D0 cellpadding=3D0
 style=3D'mso-cellspacing:0in;mso-yfti-tbllook:1184;mso-padding-alt:0in =
0in 0in 0in'>
 <tr style=3D'mso-yfti-irow:0;mso-yfti-firstrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b><span
  style=3D'mso-fareast-font-family:"Times New Roman"'>Subject: =
<o:p></o:p></span></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>I-D
  Action: draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:1'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b><span
  style=3D'mso-fareast-font-family:"Times New Roman"'>Date: =
<o:p></o:p></span></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>Thu,
  11 Aug 2011 04:28:09 -0700<o:p></o:p></span></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:2'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b><span
  style=3D'mso-fareast-font-family:"Times New Roman"'>From: =
<o:p></o:p></span></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'><a
  =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></span></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:3'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b><span
  style=3D'mso-fareast-font-family:"Times New Roman"'>Reply-To: =
<o:p></o:p></span></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'><a
  =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></span></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:4;mso-yfti-lastrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b><span
  style=3D'mso-fareast-font-family:"Times New Roman"'>To: =
<o:p></o:p></span></b></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'><a
  =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p=
></span></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><span =
style=3D'mso-fareast-font-family:
"Times New Roman"'><o:p>&nbsp;</o:p></span></p>

<pre>A New Internet-Draft is available from the on-line Internet-Drafts =
directories.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><span
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Title<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; </span>: SDP Grouping for Single RTP =
Sessions<o:p></o:p></pre><pre><span
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Author(s)<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: =
Harald Tveit Alvestrand<o:p></o:p></pre><pre><span
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Filename<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>: draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre><pre><span
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Pages<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; </span>: 8<o:p></o:p></pre><pre><span
style=3D'mso-tab-count:1'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span>Date<span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; </span>: =
2011-08-11<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>This document describes =
an extension to the Session Description<o:p></o:p></pre><pre><span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>Protocol (SDP) to =
describe RTP sessions where media of multiple =
top<o:p></o:p></pre><pre><span style=3D'mso-spacerun:yes'>&nbsp;&nbsp; =
</span>level types, for example audio and video, are carried in the same =
RTP<o:p></o:p></pre><pre><span style=3D'mso-spacerun:yes'>&nbsp;&nbsp; =
</span>session.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><span =
style=3D'mso-spacerun:yes'>&nbsp;&nbsp; </span>This document is =
presented to the RTCWEB, AVTCORE and MMUSIC WGs =
for<o:p></o:p></pre><pre><span style=3D'mso-spacerun:yes'>&nbsp;&nbsp; =
</span>consideration.<o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre><o=
:p>&nbsp;</o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>A URL for this =
Internet-Draft is:<o:p></o:p></pre><pre><a
href=3D"http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.t=
xt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</=
a><o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>Internet-Drafts are =
also available by anonymous FTP at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-=
drafts/</a><o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre><pre>This =
Internet-Draft can be retrieved at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.tx=
t">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a>=
<o:p></o:p></pre><pre>_______________________________________________<o:p=
></o:p></pre><pre>I-D-Announce mailing list<o:p></o:p></pre><pre><a
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p=
></pre><pre><a
href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.i=
etf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre><pre>Internet-D=
raft directories: <a
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/a><o:p></o:p></pre><pre>or <a
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</a><o:p></o:p></pre><pre><o:p>&nbsp;</o:p></pre></div=
>

</div>

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Date: Fri, 12 Aug 2011 15:18:37 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> A few comments:
>
> 1) Section 4 has this:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> session, and the parameters for the other sections MUST be ignored.
>
> I think it needs to be:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> and RCP session, and the parameters for the other sections MUST be
>
> ignored.
>
Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port 
multiplexing (RFC 5761), so I did not think of it.
In general, I think we should keep extensions as orthogonal as possible, 
so the normative language here should just say "establish RTCP sessions 
normally for the first section, and don't do it for other sections".
>
> 2) When the RTCP port is not the next higher (odd) port number 
> following the RTP port described in the m-line, the "a=rtcp" attribute 
> (defined in RFC-3605) is used to signal it, as in:
>
> m=audio 49170 RTP/AVP 0
>
> a=rtcp:53020 IN IP4 126.16.64.4
>
> With the grouping semantics described in the draft, it seems this RTCP 
> port should be taken only from the first section. So, this could be 
> could be added to section 4.
>
> 3) RTCP could be multiplexed with RTP on a single port and signaled 
> using the "a=rtcp-mux" attribute (defined in RFC 5761). When it is 
> used together with the grouping semantics described in the draft, it 
> seems all RTCP sessions of that group need to be multiplexed on the 
> RTP port from the first section.
>
There should be only one RTCP session established for all the TOGETHER 
m= lines, yes.
>
> Muthu
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Harald Alvestrand
> *Sent:* Thursday, August 11, 2011 5:02 PM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> I have taken some of the information I learned from the discussions in 
> Quebec City about the issues of putting video and audio into the same 
> RTP session and created a draft from them outlining a solution to the 
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the 
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.
>
>                       Harald
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> I-D Action: draft-alvestrand-one-rtp-00.txt
>
> *Date: *
>
> 	
>
> Thu, 11 Aug 2011 04:28:09 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *Reply-To: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   
>         Title            : SDP Grouping for Single RTP Sessions
>         Author(s)        : Harald Tveit Alvestrand
>         Filename         : draft-alvestrand-one-rtp-00.txt
>         Pages            : 8
>         Date             : 2011-08-11
>   
>     This document describes an extension to the Session Description
>     Protocol (SDP) to describe RTP sessions where media of multiple top
>     level types, for example audio and video, are carried in the same RTP
>     session.
>   
>     This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
>     consideration.
>   
>   
>   
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org  <mailto:I-D-Announce@ietf.org>
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:http://www.ietf.org/shadow.html
> orftp://ftp.ietf.org/ietf/1shadow-sites.txt
>   


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    On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">A
            few comments:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">1)
            Section 4 has this:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the responder understands the semantics of the TOGETHER
            extension,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">the
            parameters of the first section MUST be used to establish
            the <span class="SpellE">RTP</span><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">session,
            and the parameters for the other sections MUST be ignored.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">I
            think it needs to be:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the responder understands the semantics of the TOGETHER
            extension,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">the
            parameters of the first section MUST be used to establish
            the <span class="SpellE">RTP</span><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">and
            <span class="SpellE">RCP</span> session, and the parameters
            for the other
            sections MUST be <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">ignored.</span></p>
      </div>
    </blockquote>
    Good point. Will fix. Normally, I think RTCWEB apps will use RTCP
    port multiplexing (RFC 5761), so I did not think of it.<br>
    In general, I think we should keep extensions as orthogonal as
    possible, so the normative language here should just say "establish
    RTCP sessions normally for the first section, and don't do it for
    other sections".<br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625F795@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">2)
            When the <span class="SpellE">RTCP</span> port is not the
            next higher (odd) port
            number following the <span class="SpellE">RTP</span> port
            described in the m-line,
            the "a=<span class="SpellE">rtcp</span>" attribute (defined
            in <span class="SpellE">RFC</span>-3605) is used to signal
            it, as in:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">m=audio
            49170 <span class="SpellE">RTP</span>/<span class="SpellE">AVP</span>
            0<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">a=<span
              class="SpellE">rtcp:53020</span> IN <span class="SpellE">IP4</span>
            126.16.64.4<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">With
            the grouping semantics described in the draft, it seems this
            <span class="SpellE">RTCP</span> port should be taken only
            from the first section. So,
            this could be could be added to section 4.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">3)
            <span class="SpellE">RTCP</span> could be multiplexed with <span
              class="SpellE">RTP</span>
            on a single port and signaled using the "a=<span
              class="SpellE">rtcp-mux</span>"
            attribute (defined in <span class="SpellE">RFC</span>
            5761). When it is used
            together with the grouping semantics described in the draft,
            it seems all <span class="SpellE">RTCP</span> sessions of
            that group need to be multiplexed on the <span
              class="SpellE">RTP</span> port from the first section.</span></p>
      </div>
    </blockquote>
    There should be only one RTCP session established for all the
    TOGETHER m= lines, yes.<br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625F795@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Muthu<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <div style="border-width: medium medium medium 1.5pt;
          border-style: none none none solid; border-color:
          -moz-use-text-color -moz-use-text-color -moz-use-text-color
          blue; padding: 0in 0in 0in 4pt;">
          <div>
            <div style="border-right: medium none; border-width: 1pt
              medium medium; border-style: solid none none;
              border-color: rgb(181, 196, 223) -moz-use-text-color
              -moz-use-text-color; padding: 3pt 0in 0in;">
              <p class="MsoNormal"><b><span style="font-size: 10pt;
                    font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] <b>On Behalf Of </b>Harald
                  Alvestrand<br>
                  <b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
                  <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> [rtcweb] Fwd: I-D Action:
                  draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><span style="">I
              have taken some of the information I learned from the
              discussions in Quebec
              City about the issues of putting video and audio into the
              same RTP session and
              created a draft from them outlining a solution to the
              problem of signalling
              this configuration using SDP.<br>
              <br>
              Comments welcome, including the recommended context for
              processing the
              document; in a few days I'll send the same note to AVTCORE
              and/or MMUSIC.<br>
              <br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              Harald<br>
              <br>
              -------- Original Message -------- <o:p></o:p></span></p>
          <table class="MsoNormalTable" style="" border="0"
            cellpadding="0" cellspacing="0">
            <tbody>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Subject: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style="">I-D Action:
                      draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Date: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style="">Thu, 11 Aug 2011
                      04:28:09 -0700<o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">From: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Reply-To: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">To: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p></span></p>
                </td>
              </tr>
            </tbody>
          </table>
          <p class="MsoNormal" style="margin-bottom: 12pt;"><span
              style=""><o:p>&nbsp;</o:p></span></p>
          <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Title<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: SDP Grouping for Single RTP Sessions<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Author(s)<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: Harald Tveit Alvestrand<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Filename<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Pages<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: 8<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Date<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: 2011-08-11<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>This document describes an extension to the Session Description<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>Protocol (SDP) to describe RTP sessions where media of multiple top<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>level types, for example audio and video, are carried in the same RTP<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>session.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>consideration.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>A URL for this Internet-Draft is:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>This Internet-Draft can be retrieved at:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
          <pre>_______________________________________________<o:p></o:p></pre>
          <pre>I-D-Announce mailing list<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre>
          <pre>Internet-Draft directories: <a moz-do-not-send="true" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a><o:p></o:p></pre>
          <pre>or <a moz-do-not-send="true" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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Thread-Topic: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Harald Alvestrand" <harald@alvestrand.no>
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Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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|In general, I think we should keep extensions as orthogonal
|as possible, so the normative language here should just say
|"establish RTCP sessions normally for the first section, and=20
|don't do it for other sections".
=20
Looks good to me. Should it also say "Within a TOGETHER group it is
RECOMMENDED to multiplex RTCP on the same RTP port from the first
section" or something stronger?
=20
Muthu
=20
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Friday, August 12, 2011 6:49 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:=20
A few comments:
=20
1) Section 4 has this:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
session, and the parameters for the other sections MUST be ignored.
=20
I think it needs to be:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
and RCP session, and the parameters for the other sections MUST be=20
ignored.
Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port
multiplexing (RFC 5761), so I did not think of it.
In general, I think we should keep extensions as orthogonal as possible,
so the normative language here should just say "establish RTCP sessions
normally for the first section, and don't do it for other sections".


=20
2) When the RTCP port is not the next higher (odd) port number following
the RTP port described in the m-line, the "a=3Drtcp" attribute (defined =
in
RFC-3605) is used to signal it, as in:
m=3Daudio 49170 RTP/AVP 0
a=3Drtcp:53020 IN IP4 126.16.64.4
=20
With the grouping semantics described in the draft, it seems this RTCP
port should be taken only from the first section. So, this could be
could be added to section 4.
=20
3) RTCP could be multiplexed with RTP on a single port and signaled
using the "a=3Drtcp-mux" attribute (defined in RFC 5761). When it is =
used
together with the grouping semantics described in the draft, it seems
all RTCP sessions of that group need to be multiplexed on the RTP port
from the first section.
There should be only one RTCP session established for all the TOGETHER
m=3D lines, yes.


=20
Muthu
=20
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Harald Alvestrand
Sent: Thursday, August 11, 2011 5:02 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
I have taken some of the information I learned from the discussions in
Quebec City about the issues of putting video and audio into the same
RTP session and created a draft from them outlining a solution to the
problem of signalling this configuration using SDP.

Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or
MMUSIC.

                      Harald

-------- Original Message --------=20
Subject:=20
I-D Action: draft-alvestrand-one-rtp-00.txt
Date:=20
Thu, 11 Aug 2011 04:28:09 -0700
From:=20
internet-drafts@ietf.org
Reply-To:=20
internet-drafts@ietf.org
To:=20
i-d-announce@ietf.org
=20
A New Internet-Draft is available from the on-line Internet-Drafts
directories.
=20
       Title           : SDP Grouping for Single RTP Sessions
       Author(s)       : Harald Tveit Alvestrand
       Filename        : draft-alvestrand-one-rtp-00.txt
       Pages           : 8
       Date            : 2011-08-11
=20
   This document describes an extension to the Session Description
   Protocol (SDP) to describe RTP sessions where media of multiple top
   level types, for example audio and video, are carried in the same RTP
   session.
=20
   This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
   consideration.
=20
=20
=20
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
=20
Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/
=20
This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
_______________________________________________
I-D-Announce mailing list
I-D-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/i-d-announce
Internet-Draft directories: http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
=20
=20

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<body bgcolor=3Dwhite lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:.5in'>

<div class=3DWordSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|In
general, I think we should keep extensions as =
orthogonal<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|as
possible, so the normative language here should just =
say<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|&quot;establish
<span class=3DSpellE>RTCP</span> sessions normally for the first =
section, and <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|don't
do it for other sections&quot;.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>Looks
good to me. Should it also say &quot;Within a TOGETHER group it is =
RECOMMENDED to
multiplex <span class=3DSpellE>RTCP</span> on the same <span =
class=3DSpellE>RTP</span>
port from the first section&quot; or something =
stronger?<o:p></o:p></span></p>

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style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>Muthu<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

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<div>

<div style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt =
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<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
mso-fareast-font-family:"Times New =
Roman";color:windowtext'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";mso-fareast-f=
ont-family:
"Times New Roman";color:windowtext'> Harald Alvestrand
[mailto:harald@alvestrand.no] <br>
<b>Sent:</b> Friday, August 12, 2011 6:49 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>On
08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote: =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>A few comments:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>1) Section 4 has this:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>If the responder understands the semantics of the =
TOGETHER
extension,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>the parameters of the first section MUST be used to =
establish
the RTP</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>session, and the parameters for the other sections =
MUST be
ignored.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>I think it needs to be:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>If the responder understands the semantics of the =
TOGETHER extension,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>the parameters of the first section MUST be used to =
establish
the RTP</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>and RCP session, and the parameters for the other =
sections
MUST be </span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>ignored.</span><o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>Good
point. Will fix. Normally, I think RTCWEB apps will use RTCP port =
multiplexing
(RFC 5761), so I did not think of it.<br>
In general, I think we should keep extensions as orthogonal as possible, =
so the
normative language here should just say &quot;establish RTCP sessions =
normally
for the first section, and don't do it for other sections&quot;.<br
style=3D'mso-special-character:line-break'>
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<![endif]><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>2) When the RTCP port is not the next higher (odd) =
port
number following the RTP port described in the m-line, the =
&quot;a=3Drtcp&quot;
attribute (defined in RFC-3605) is used to signal it, as =
in:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>m=3Daudio 49170 RTP/AVP 0</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>a=3Drtcp:53020 IN IP4 =
126.16.64.4</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>With the grouping semantics described in the draft, it =
seems
this RTCP port should be taken only from the first section. So, this =
could be
could be added to section 4.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>3) RTCP could be multiplexed with RTP on a single port =
and
signaled using the &quot;a=3Drtcp-mux&quot; attribute (defined in RFC =
5761). When
it is used together with the grouping semantics described in the draft, =
it
seems all RTCP sessions of that group need to be multiplexed on the RTP =
port
from the first section.</span><o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>There
should be only one RTCP session established for all the TOGETHER m=3D =
lines, yes.<br
style=3D'mso-special-character:line-break'>
<![if !supportLineBreakNewLine]><br =
style=3D'mso-special-character:line-break'>
<![endif]><o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>Muthu</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<div style=3D'border:none;border-left:solid windowtext 1.5pt;padding:0in =
0in 0in 4.0pt;
border-color:-moz-use-text-color -moz-use-text-color =
-moz-use-text-color&#13;&#10;          blue'>

<div>

<div style=3D'border:none;border-top:solid windowtext =
1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color&#13;&#10;              =
-moz-use-text-color'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> <a
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] <b>On
Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>

</div>

</div>

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

<p class=3DMsoNormal>I have taken some of the information I learned from =
the
discussions in Quebec City about the issues of putting video and audio =
into the
same RTP session and created a draft from them outlining a solution to =
the
problem of signalling this configuration using SDP.<br>
<br>
Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or =
MMUSIC.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Harald<br>
<br>
-------- Original Message -------- <o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellspacing=3D0 cellpadding=3D0
 style=3D'mso-cellspacing:0in;mso-yfti-tbllook:1184;mso-padding-alt:0in =
5.4pt 0in 5.4pt'>
 <tr style=3D'mso-yfti-irow:0;mso-yfti-firstrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright =
style=3D'text-align:right'><b>Subject: </b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>I-D Action: =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:1'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>Date: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>Thu, 11 Aug 2011 04:28:09 -0700<o:p></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:2'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>From: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:3'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright =
style=3D'text-align:right'><b>Reply-To: </b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:4;mso-yfti-lastrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>To: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p=
></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>&nbsp;<o:p></o:p></p>

<pre>A New Internet-Draft is available from the on-line Internet-Drafts =
directories.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : SDP =
Grouping for Single RTP =
Sessions<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Harald Tveit =
Alvestrand<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
8<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2011-08-11<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp; =
This document describes an extension to the Session =
Description<o:p></o:p></pre><pre>&nbsp;&nbsp; Protocol (SDP) to describe =
RTP sessions where media of multiple =
top<o:p></o:p></pre><pre>&nbsp;&nbsp; level types, for example audio and =
video, are carried in the same RTP<o:p></o:p></pre><pre>&nbsp;&nbsp; =
session.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp; =
This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs =
for<o:p></o:p></pre><pre>&nbsp;&nbsp; =
consideration.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;<o:=
p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>A URL for this =
Internet-Draft is:<o:p></o:p></pre><pre><a
href=3D"http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.t=
xt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</=
a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>Internet-Drafts are =
also available by anonymous FTP at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-=
drafts/</a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>This =
Internet-Draft can be retrieved at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.tx=
t">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a>=
<o:p></o:p></pre><pre>_______________________________________________<o:p=
></o:p></pre><pre>I-D-Announce mailing list<o:p></o:p></pre><pre><a
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p=
></pre><pre><a
href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.i=
etf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre><pre>Internet-D=
raft directories: <a
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/a><o:p></o:p></pre><pre>or <a
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre></div=
>

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'><o:p>&nbsp;</o:p></span></p>

</div>

</div>

</body>

</html>

------_=_NextPart_001_01CC58F9.D1350FE2--

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From: Igor Faynberg <igor.faynberg@alcatel-lucent.com>
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References: <CA69AF47.1CBB0%henry.sinnreich@gmail.com>	<4E445F49.6080705@alcatel-lucent.com> <4E449E6B.8020205@alum.mit.edu> <4E450F00.9090908@alvestrand.no>
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Cc: rtcweb@ietf.org, Phil Zimmermann <prz@mit.edu>
Subject: Re: [rtcweb] Exchange of trusted identities in server-based conferencing (Re: Use case change request: Identity in multiuser calls)
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On 8/12/2011 7:31 AM, Harald Alvestrand wrote:
> ...
> The normal method of authentication for teleconferences today is that 
> people prove they have the conference number (by calling in), and 
> recognize each others' voices.
> ...

Is there not a conference code, too?  In all (five or six of them that I 
have used) there was one.  This code is, effectively, a shared secret. 
In the service we use, the conference host can create a new code for 
each meeting.  (One problem with relying on recognizing voices is 
passive intruding: I can join your conference and never say a word.)

Igor
>

From oej@edvina.net  Fri Aug 12 07:38:33 2011
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Coming in late to the discussion, I can't find any traces of a =
discussion about dual stacks. Are the issues related with this supposed =
to be solved with ICE/STUN/Turn - if so, shouldn't this be documented?

If not, what do we do? Happy RTC-eyeballs?

/O=

From pkyzivat@alum.mit.edu  Fri Aug 12 07:55:03 2011
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On 8/12/11 2:45 AM, Harald Alvestrand wrote:
> On 08/11/11 20:47, Paul Kyzivat wrote:
>> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>>> I have taken some of the information I learned from the discussions in
>>> Quebec City about the issues of putting video and audio into the same
>>> RTP session and created a draft from them outlining a solution to the
>>> problem of signalling this configuration using SDP.
>>>
>>> Comments welcome, including the recommended context for processing the
>>> document; in a few days I'll send the same note to AVTCORE and/or
>>> MMUSIC.
>>
>> Harald,
>>
>> A few questions/comments:
>>
>> 1) If ICE is being used, how do you expect it to be handled on the 2nd
>> m-line? Since one of the goals of multiplexing on a single rtp session
>> is to minimize ICE processing, one would hope only to do ICE on the
>> first of the grouped m-lines. But if the answerer doesn't support this
>> grouping, then the ICE will be needed on the others.
> I tried to be explicit that if negotiation is successful, only the ICE
> parameters from the first section listed in the A=group:TOGETHER would
> be used:

Sure. But if the other side *doesn't* support TOGETHER, then ICE will 
need to be negotiated independently for each m-line. And for that to 
work, the port for the 2nd m-line but be active, other ICE candidates 
identified and included for it, etc.

ISTM the implications of that need further explanation.
(I'm not an expert on ICE. If I were, maybe this would be obvious.)

> The following parameters are taken from the first section only:
>
> o Port number from the m= line
>
> o All media-level attributes defined in RFC 5245 section 15.1 - this
> includes "candidate", "remote-candidates", "ice-mismatch", "ice-
> ufrag", "ice-pwd"
>
> Suggestions for improving this language?
>
>> 2) The draft only shows this mechanism being used to combine one audio
>> and one video. Do you also imagine it being used to represent multiple
>> audio and/or multiple video streams that might otherwise be handled
>> independently? (I see no reason why it couldn't be used that way.)
>> This would allow description of different properties for each one.
> I certainly think that any number of m= sections could be combined this
> way, but in other discussions, I've seen people tend towards thinking of
> sections as "one m= section for all the audio channels, one m= section
> for all the video channels". Can you give an example of the usage you're
> thinking of?

If there was an intent to have multiple streams with different 
properties, then this would provide a way to describe those, that isn't 
otherwise available. For instance, if you wanted one hi-res video stream 
and one low-res video stream.

I'm also thinking ahead to what might be coming from CLUE 
(telepresence). It isn't far enough along to yet be able to predict how 
it will set up its multi-stream calls, but it will probably want to 
multiplex them over a lesser number of RTP streams. And so I'm shopping 
for potential mechanisms that might be exploited there.

> I'm worried that we may get too many special rules about how parameters
> from sections are to be combined - multiple codecs with different PT
> numbers are fine, but other parameters could be confusing, so I'd like
> to see examples.

Maybe its premature. But no matter what, rules will need to be written 
on how attributes are combined. So IMO its worthwhile to at least start 
thinking about how to write those in general, rather than writing a few 
special case hacks. It may turn out that the rules that are to be 
followed for combining one audio and one video will also "just work" for 
combining two audio or two video, etc.

	Thanks,
	Paul

>> 3) In the example, for an entity that doesn't understand TOGETHER, you
>> still show the a=mid lines. This is appropriate for an answerer that
>> *does* understand a=group. However an answerer that doesn't implement
>> rfc3388 would presumably omit the a=mid lines from the answer. So the
>> offerer had better be prepared for that as well.
> Yes, that's a reasonable point. I'll add a third example answer.
>>
>> Thanks,
>> Paul
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>


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On 08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> |In general, I think we should keep extensions as orthogonal
>
> |as possible, so the normative language here should just say
>
> |"establish RTCP sessions normally for the first section, and
>
> |don't do it for other sections".
>
> Looks good to me. Should it also say "Within a TOGETHER group it is 
> RECOMMENDED to multiplex RTCP on the same RTP port from the first 
> section" or something stronger?
>
I think it should be up to the offer generator whether he adds 
"a=rtcp-mux" to the first section or not.
Do we need to state that if the first section has it, all the other 
sections have to have it, or can we live with a mixture?
>
> Muthu
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Friday, August 12, 2011 6:49 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> A few comments:
>
> 1) Section 4 has this:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> session, and the parameters for the other sections MUST be ignored.
>
> I think it needs to be:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> and RCP session, and the parameters for the other sections MUST be
>
> ignored.
>
> Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port 
> multiplexing (RFC 5761), so I did not think of it.
> In general, I think we should keep extensions as orthogonal as 
> possible, so the normative language here should just say "establish 
> RTCP sessions normally for the first section, and don't do it for 
> other sections".
>
> 2) When the RTCP port is not the next higher (odd) port number 
> following the RTP port described in the m-line, the "a=rtcp" attribute 
> (defined in RFC-3605) is used to signal it, as in:
>
> m=audio 49170 RTP/AVP 0
>
> a=rtcp:53020 IN IP4 126.16.64.4
>
> With the grouping semantics described in the draft, it seems this RTCP 
> port should be taken only from the first section. So, this could be 
> could be added to section 4.
>
> 3) RTCP could be multiplexed with RTP on a single port and signaled 
> using the "a=rtcp-mux" attribute (defined in RFC 5761). When it is 
> used together with the grouping semantics described in the draft, it 
> seems all RTCP sessions of that group need to be multiplexed on the 
> RTP port from the first section.
>
> There should be only one RTCP session established for all the TOGETHER 
> m= lines, yes.
>
> Muthu
>
> *From:*rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org> 
> [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Harald Alvestrand
> *Sent:* Thursday, August 11, 2011 5:02 PM
> *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> I have taken some of the information I learned from the discussions in 
> Quebec City about the issues of putting video and audio into the same 
> RTP session and created a draft from them outlining a solution to the 
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the 
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.
>
>                       Harald
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> I-D Action: draft-alvestrand-one-rtp-00.txt
>
> *Date: *
>
> 	
>
> Thu, 11 Aug 2011 04:28:09 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *Reply-To: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   
>         Title           : SDP Grouping for Single RTP Sessions
>         Author(s)       : Harald Tveit Alvestrand
>         Filename        : draft-alvestrand-one-rtp-00.txt
>         Pages           : 8
>         Date            : 2011-08-11
>   
>     This document describes an extension to the Session Description
>     Protocol (SDP) to describe RTP sessions where media of multiple top
>     level types, for example audio and video, are carried in the same RTP
>     session.
>   
>     This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
>     consideration.
>   
>   
>   
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org  <mailto:I-D-Announce@ietf.org>
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:http://www.ietf.org/shadow.html
> orftp://ftp.ietf.org/ietf/1shadow-sites.txt
>   
>


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    On 08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|In
            general, I think we should keep extensions as orthogonal<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|as
            possible, so the normative language here should just say<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|"establish
            <span class="SpellE">RTCP</span> sessions normally for the
            first section, and <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|don't
            do it for other sections".<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
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        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Looks
            good to me. Should it also say "Within a TOGETHER group it
            is RECOMMENDED to
            multiplex <span class="SpellE">RTCP</span> on the same <span
              class="SpellE">RTP</span>
            port from the first section" or something stronger?</span></p>
      </div>
    </blockquote>
    I think it should be up to the offer generator whether he adds
    "a=rtcp-mux" to the first section or not.<br>
    Do we need to state that if the first section has it, all the other
    sections have to have it, or can we live with a mixture?<br>
    <span style="font-size: 10pt; font-family: &quot;Courier New&quot;;
      color: windowtext;"><o:p>&nbsp;</o:p></span>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625F7A6@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Muthu<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <div style="border-width: medium medium medium 1.5pt;
          border-style: none none none solid; border-color:
          -moz-use-text-color -moz-use-text-color -moz-use-text-color
          blue; padding: 0in 0in 0in 4pt;">
          <div>
            <div style="border-right: medium none; border-width: 1pt
              medium medium; border-style: solid none none;
              border-color: rgb(181, 196, 223) -moz-use-text-color
              -moz-use-text-color; padding: 3pt 0in 0in;">
              <p class="MsoNormal"><b><span style="font-size: 10pt;
                    font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> Harald Alvestrand
                  [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
                  <b>Sent:</b> Friday, August 12, 2011 6:49 PM<br>
                  <b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
                  <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> Re: [rtcweb] Fwd: I-D Action:
                  draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><span style="">On
              08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal)
              wrote: <o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">A
              few comments:</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">1)
              Section 4 has this:</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">If
              the responder understands the semantics of the TOGETHER
              extension,</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">the
              parameters of the first section MUST be used to establish
              the RTP</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">session,
              and the parameters for the other sections MUST be
              ignored.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">I
              think it needs to be:</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">If
              the responder understands the semantics of the TOGETHER
              extension,</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">the
              parameters of the first section MUST be used to establish
              the RTP</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">and
              RCP session, and the parameters for the other sections
              MUST be </span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">ignored.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="">Good
              point. Will fix. Normally, I think RTCWEB apps will use
              RTCP port multiplexing
              (RFC 5761), so I did not think of it.<br>
              In general, I think we should keep extensions as
              orthogonal as possible, so the
              normative language here should just say "establish RTCP
              sessions normally
              for the first section, and don't do it for other
              sections".<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">2)
              When the RTCP port is not the next higher (odd) port
              number following the RTP port described in the m-line, the
              "a=rtcp"
              attribute (defined in RFC-3605) is used to signal it, as
              in:</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">m=audio
              49170 RTP/AVP 0</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">a=rtcp:53020
              IN IP4 126.16.64.4</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">With
              the grouping semantics described in the draft, it seems
              this RTCP port should be taken only from the first
              section. So, this could be
              could be added to section 4.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">3)
              RTCP could be multiplexed with RTP on a single port and
              signaled using the "a=rtcp-mux" attribute (defined in RFC
              5761). When
              it is used together with the grouping semantics described
              in the draft, it
              seems all RTCP sessions of that group need to be
              multiplexed on the RTP port
              from the first section.</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="">There
              should be only one RTCP session established for all the
              TOGETHER m= lines, yes.<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">Muthu</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <div style="border-width: medium medium medium 1.5pt;
            border-style: none none none solid; padding: 0in 0in 0in
            4pt; border-color: -moz-use-text-color -moz-use-text-color
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            <div>
              <div style="border-right: medium none; border-width: 1pt
                medium medium; border-style: solid none none; padding:
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                <p class="MsoNormal"><b><span style="font-size: 10pt;
                      font-family:
                      &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                      windowtext;">From:</span></b><span
                    style="font-size: 10pt; font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;"> <a moz-do-not-send="true"
                      href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                    [<a moz-do-not-send="true"
                      href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                    <b>On
                      Behalf Of </b>Harald Alvestrand<br>
                    <b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
                    <b>To:</b> <a moz-do-not-send="true"
                      href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    <b>Subject:</b> [rtcweb] Fwd: I-D Action:
                    draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>
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            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
            <p class="MsoNormal">I have taken some of the information I
              learned from the
              discussions in Quebec City about the issues of putting
              video and audio into the
              same RTP session and created a draft from them outlining a
              solution to the
              problem of signalling this configuration using SDP.<br>
              <br>
              Comments welcome, including the recommended context for
              processing the
              document; in a few days I'll send the same note to AVTCORE
              and/or MMUSIC.<br>
              <br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              Harald<br>
              <br>
              -------- Original Message -------- <o:p></o:p></p>
            <table class="MsoNormalTable" style="" border="0"
              cellpadding="0" cellspacing="0">
              <tbody>
                <tr style="">
                  <td style="padding: 0in;" nowrap="nowrap" valign="top">
                    <p class="MsoNormal" style="text-align: right;"
                      align="right"><b>Subject: </b><o:p></o:p></p>
                  </td>
                  <td style="padding: 0in;">
                    <p class="MsoNormal">I-D Action:
                      draft-alvestrand-one-rtp-00.txt<o:p></o:p></p>
                  </td>
                </tr>
                <tr style="">
                  <td style="padding: 0in;" nowrap="nowrap" valign="top">
                    <p class="MsoNormal" style="text-align: right;"
                      align="right"><b>Date: </b><o:p></o:p></p>
                  </td>
                  <td style="padding: 0in;">
                    <p class="MsoNormal">Thu, 11 Aug 2011 04:28:09 -0700<o:p></o:p></p>
                  </td>
                </tr>
                <tr style="">
                  <td style="padding: 0in;" nowrap="nowrap" valign="top">
                    <p class="MsoNormal" style="text-align: right;"
                      align="right"><b>From: </b><o:p></o:p></p>
                  </td>
                  <td style="padding: 0in;">
                    <p class="MsoNormal"><a moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                  </td>
                </tr>
                <tr style="">
                  <td style="padding: 0in;" nowrap="nowrap" valign="top">
                    <p class="MsoNormal" style="text-align: right;"
                      align="right"><b>Reply-To: </b><o:p></o:p></p>
                  </td>
                  <td style="padding: 0in;">
                    <p class="MsoNormal"><a moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                  </td>
                </tr>
                <tr style="">
                  <td style="padding: 0in;" nowrap="nowrap" valign="top">
                    <p class="MsoNormal" style="text-align: right;"
                      align="right"><b>To: </b><o:p></o:p></p>
                  </td>
                  <td style="padding: 0in;">
                    <p class="MsoNormal"><a moz-do-not-send="true"
                        href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p></p>
                  </td>
                </tr>
              </tbody>
            </table>
            <p class="MsoNormal" style="margin-bottom: 12pt;">&nbsp;<o:p></o:p></p>
            <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : SDP Grouping for Single RTP Sessions<o:p></o:p></pre>
            <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Harald Tveit Alvestrand<o:p></o:p></pre>
            <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre>
            <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 8<o:p></o:p></pre>
            <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2011-08-11<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; This document describes an extension to the Session Description<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; Protocol (SDP) to describe RTP sessions where media of multiple top<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; level types, for example audio and video, are carried in the same RTP<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; session.<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for<o:p></o:p></pre>
            <pre>&nbsp;&nbsp; consideration.<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>A URL for this Internet-Draft is:<o:p></o:p></pre>
            <pre><a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></pre>
            <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
            <pre>This Internet-Draft can be retrieved at:<o:p></o:p></pre>
            <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
            <pre>_______________________________________________<o:p></o:p></pre>
            <pre>I-D-Announce mailing list<o:p></o:p></pre>
            <pre><a moz-do-not-send="true" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p></pre>
            <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre>
            <pre>Internet-Draft directories: <a moz-do-not-send="true" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a><o:p></o:p></pre>
            <pre>or <a moz-do-not-send="true" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><o:p></o:p></pre>
            <pre>&nbsp;<o:p></o:p></pre>
          </div>
          <p class="MsoNormal"><span style=""><o:p>&nbsp;</o:p></span></p>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------060808080707090503040200--

From harald@alvestrand.no  Fri Aug 12 08:20:07 2011
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Date: Fri, 12 Aug 2011 17:20:41 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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On 08/12/11 16:55, Paul Kyzivat wrote:
> On 8/12/11 2:45 AM, Harald Alvestrand wrote:
>> On 08/11/11 20:47, Paul Kyzivat wrote:
>>> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>>>> I have taken some of the information I learned from the discussions in
>>>> Quebec City about the issues of putting video and audio into the same
>>>> RTP session and created a draft from them outlining a solution to the
>>>> problem of signalling this configuration using SDP.
>>>>
>>>> Comments welcome, including the recommended context for processing the
>>>> document; in a few days I'll send the same note to AVTCORE and/or
>>>> MMUSIC.
>>>
>>> Harald,
>>>
>>> A few questions/comments:
>>>
>>> 1) If ICE is being used, how do you expect it to be handled on the 2nd
>>> m-line? Since one of the goals of multiplexing on a single rtp session
>>> is to minimize ICE processing, one would hope only to do ICE on the
>>> first of the grouped m-lines. But if the answerer doesn't support this
>>> grouping, then the ICE will be needed on the others.
>> I tried to be explicit that if negotiation is successful, only the ICE
>> parameters from the first section listed in the A=group:TOGETHER would
>> be used:
>
> Sure. But if the other side *doesn't* support TOGETHER, then ICE will 
> need to be negotiated independently for each m-line. And for that to 
> work, the port for the 2nd m-line but be active, other ICE candidates 
> identified and included for it, etc.
>
> ISTM the implications of that need further explanation.
> (I'm not an expert on ICE. If I were, maybe this would be obvious.)
There are 2 possible things to do for the offerer:
- Start ICE procedure on all ports when sending the offer
- Wait until the answer comes back, and either start ICE procedure on 
one port (if the responder understood TOGETHER), or start the ICE 
procedure on all ports (if the responder did not understand TOGETHER)

The first alternative is a number of milliseconds faster, but others 
should chime in on whether that's significant or not.

>
>> The following parameters are taken from the first section only:
>>
>> o Port number from the m= line
>>
>> o All media-level attributes defined in RFC 5245 section 15.1 - this
>> includes "candidate", "remote-candidates", "ice-mismatch", "ice-
>> ufrag", "ice-pwd"
>>
>> Suggestions for improving this language?
>>
>>> 2) The draft only shows this mechanism being used to combine one audio
>>> and one video. Do you also imagine it being used to represent multiple
>>> audio and/or multiple video streams that might otherwise be handled
>>> independently? (I see no reason why it couldn't be used that way.)
>>> This would allow description of different properties for each one.
>> I certainly think that any number of m= sections could be combined this
>> way, but in other discussions, I've seen people tend towards thinking of
>> sections as "one m= section for all the audio channels, one m= section
>> for all the video channels". Can you give an example of the usage you're
>> thinking of?
>
> If there was an intent to have multiple streams with different 
> properties, then this would provide a way to describe those, that 
> isn't otherwise available. For instance, if you wanted one hi-res 
> video stream and one low-res video stream.
>
> I'm also thinking ahead to what might be coming from CLUE 
> (telepresence). It isn't far enough along to yet be able to predict 
> how it will set up its multi-stream calls, but it will probably want 
> to multiplex them over a lesser number of RTP streams. And so I'm 
> shopping for potential mechanisms that might be exploited there.
>
>> I'm worried that we may get too many special rules about how parameters
>> from sections are to be combined - multiple codecs with different PT
>> numbers are fine, but other parameters could be confusing, so I'd like
>> to see examples.
>
> Maybe its premature. But no matter what, rules will need to be written 
> on how attributes are combined. So IMO its worthwhile to at least 
> start thinking about how to write those in general, rather than 
> writing a few special case hacks. It may turn out that the rules that 
> are to be followed for combining one audio and one video will also 
> "just work" for combining two audio or two video, etc.
>
>     Thanks,
>     Paul
>
>>> 3) In the example, for an entity that doesn't understand TOGETHER, you
>>> still show the a=mid lines. This is appropriate for an answerer that
>>> *does* understand a=group. However an answerer that doesn't implement
>>> rfc3388 would presumably omit the a=mid lines from the answer. So the
>>> offerer had better be prepared for that as well.
>> Yes, that's a reasonable point. I'll add a third example answer.
>>>
>>> Thanks,
>>> Paul
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>>
>
>


From pkyzivat@alum.mit.edu  Fri Aug 12 08:23:33 2011
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On 8/12/11 2:49 AM, Harald Alvestrand wrote:
> On 08/12/11 05:37, Ross Finlayson wrote:
>> Thanks, Harald, for this submission.
>>
>> A nit (I think). Unless I'm misunderstanding the purpose of the
>> "a=group:TOGETHER" attribute, the port numbers in the two example
>> 'answers' are wrong.
> I'm not sure. These examples came straight out of RFC 4317, including
> the port numbers; I noted the port number mismatch, and concluded
> (tentatively) that the port number of an offer means "I'll be using this
> port", while the port number of an answer means "I'll be using this
> port" - which means that they SHOULD be expected to be different.
>
> When using ICE, the port number is moot anyway (the "candidates"
> overrides them), so the only purpose of the port number is that the
> special value zero in an answer means "offer not accepted".

I had wondered about this too, and forgot to comment.
IMO the offer is right - you need two different ports in case TOGETHER 
isn't understood by answerer.

But in the answer, if TOGETHER *is* used, I wonder about the port. 
Putting a unique port there certainly implies to me that it will be 
used. (Though ICE may change things.) But using the *same* port in the 
answer may not be right either. In general, SDP doesn't allow you to use 
the same port in multiple m-lines. IIRC there is an exception to that, 
but I'll have to hunt to find what that is.

Putting zero as the port in the 2nd m-line of the answer might be a 
solution. It would certainly indicate that only the one port will be 
used. The issue is that when its zero, generally the rest of the stuff 
about that stream is ignored. However, no matter what, *some* rules have 
to be changed to make TOGETHER work.

I guess a question is: would it ever make sense for the answerer to 
indicate support for TOGETHER, and yet want to refuse one of the m-lines?

As long as there are never more than two m-lines combined with TOGETHER, 
refusing one of them can be done by *not* indicating support for 
TOGETHER, and then giving port=0 for the m-lines not desired. And that 
works even if only the 2nd m-line is desired.

If there were three or more m-lines combined with TOGETHER, it gets more 
complex. (To be concrete, say there was audio, video, and timed text. in 
the offer.) And suppose the answerer only wants audio and timed text. In 
that case I guess it does need a way to reject the video while still 
using TOGETHER. And port=0 for the video is the obvious way.

That brings up another peculiar case. Suppose the m-lines in question 
were, in order, video, audio, and timed text, and the answerer again 
only wanted to use audio and timed text. In that case, will it still 
want to use the port offered for video??? How would the answer be 
constructed, and where would the answering port be placed?

It occurs to me that there is another "special" port that might be 
co-opted for use here: port 9, the discard port. Perhaps port 9 could be 
used in the answer, when TOGETHER is used in the answer, to indicate 
m-lines that *are* to be combined with one of the other m-lines, while 
port 0 would be used to indicate m-lines that are being rejected 
entirely. Then there should only be one m-line in the group that has a 
port not equal to either 0 or 9, and it would be the one to use. (Or the 
one to negotiate ICE on.)

	Thanks,
	Paul

>> Example 1:
>>    Answer, from an entity that understands TOGETHER
>>
>>           v=0
>>           o=bob 2808844564 2808844564 IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>>           s=
>>           c=IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>>           t=0 0
>>           a=group:TOGETHER foo bar
>>           m=audio 49174 RTP/AVP 0
>>           a=mid:foo
>>           b=AS:200
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49170 RTP/AVP 32
>>           a=mid:bar
>>           b=AS:1000
>>           a=rtpmap:32 MPV/90000
>> Should be (I think):
>>           ...
>>           a=group:TOGETHER foo bar
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           b=AS:200
>>           a=rtpmap:0 PCMU/8000
>>           m=video 49170 RTP/AVP 32
>>           a=mid:bar
>>           b=AS:1000
>>           a=rtpmap:32 MPV/90000
>> Example 2:
>>     Answer, from an entity that understands grouping, but does not
>>     understand TOGETHER
>>
>>           v=0
>>           o=bob 2808844564 2808844564 IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>>           s=
>>           c=IN IP4host.biloxi.example.com  <http://host.biloxi.example.com>
>>           t=0 0
>>           m=audio 49174 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           b=AS:200
>>           m=video 49170 RTP/AVP 32
>>           a=mid:bar
>>           a=rtpmap:32 MPV/90000
>>           b=AS:1000
>> Should be (I think):
>>           ...
>>           m=audio 49170 RTP/AVP 0
>>           a=mid:foo
>>           a=rtpmap:0 PCMU/8000
>>           b=AS:200
>>           m=video 51372 RTP/AVP 32
>>           a=mid:bar
>>           a=rtpmap:32 MPV/90000
>>           b=AS:1000
>>
>> Ross Finlayson
>> Live Networks, Inc.
>> http://www.live555.com/
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From harald@alvestrand.no  Fri Aug 12 08:23:33 2011
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Cc: rtcweb@ietf.org
Subject: [rtcweb] Combining attributes (Re: Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt)
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On 08/12/11 16:55, Paul Kyzivat wrote:
> On 8/12/11 2:45 AM, Harald Alvestrand wrote:
>> On 08/11/11 20:47, Paul Kyzivat wrote:
>>> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>>>
>>
>>> 2) The draft only shows this mechanism being used to combine one audio
>>> and one video. Do you also imagine it being used to represent multiple
>>> audio and/or multiple video streams that might otherwise be handled
>>> independently? (I see no reason why it couldn't be used that way.)
>>> This would allow description of different properties for each one.
>> I certainly think that any number of m= sections could be combined this
>> way, but in other discussions, I've seen people tend towards thinking of
>> sections as "one m= section for all the audio channels, one m= section
>> for all the video channels". Can you give an example of the usage you're
>> thinking of?
>
> If there was an intent to have multiple streams with different 
> properties, then this would provide a way to describe those, that 
> isn't otherwise available. For instance, if you wanted one hi-res 
> video stream and one low-res video stream.
>
> I'm also thinking ahead to what might be coming from CLUE 
> (telepresence). It isn't far enough along to yet be able to predict 
> how it will set up its multi-stream calls, but it will probably want 
> to multiplex them over a lesser number of RTP streams. And so I'm 
> shopping for potential mechanisms that might be exploited there.
>
>> I'm worried that we may get too many special rules about how parameters
>> from sections are to be combined - multiple codecs with different PT
>> numbers are fine, but other parameters could be confusing, so I'd like
>> to see examples.
>
> Maybe its premature. But no matter what, rules will need to be written 
> on how attributes are combined. So IMO its worthwhile to at least 
> start thinking about how to write those in general, rather than 
> writing a few special case hacks. It may turn out that the rules that 
> are to be followed for combining one audio and one video will also 
> "just work" for combining two audio or two video, etc.
That's why I'm urging you to come up with concrete examples; I think 
it's easier to work out the general principles when we have worked 
through more than one example.

At the moment we have the "ICE way" (take the first), the "bandwidth 
way" (add the numbers), and the "codec way" (use them all, but require 
that they don't conflict).

If you have a good example-from-real-life of an SDP offer where you 
would like to use TOGETHER, let's see whether there are more cases.


From pkyzivat@alum.mit.edu  Fri Aug 12 08:43:19 2011
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Cc: rtcweb@ietf.org, Phil Zimmermann <prz@mit.edu>
Subject: Re: [rtcweb] Exchange of trusted identities in server-based conferencing (Re: Use case change request: Identity in multiuser calls)
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On 8/12/11 7:31 AM, Harald Alvestrand wrote:
> [[ Reminder to all: If you discover that you're talking about something
> far away from the original topic of the thread, please CHANGE THE
> SUBJECT LINE! ]]
>
> The normal method of authentication for teleconferences today is that
> people prove they have the conference number (by calling in), and
> recognize each others' voices.
> We have security procedures at Google that depend on people recognizing
> each others' faces (sometimes based on prior meetings, sometimes based
> on matching with pictures).
>
> The digital realm doesn't necessarily have to provide a steel vault
> where the analog realm is satisfied with plywood - or at least, I think
> it is not the best for finishing RTCWEB in a reasonable time if we
> insist that steel vaults be part of the requirements.

AFAIK this thread has evolved from Henry's original question about 
identification of participants in confidential conferences. And perhaps 
the proper answer to that is what you say above. While its an 
interesting theoretical issue, in practice I find even the existing 
minimal security mechanisms to be annoying and almost always 
unnecessary. (We flatter ourselves when we think somebody would be 
interested in hearing our discussions.)

There has been an assertion here that ZRTP is useful for solving this 
problem. I don't know if it is - I have never used ZRTP. But it appears 
that it might be difficult to make work well in a conference. (Or maybe 
this has already been worked out. If so, that's great.)

	Thanks,
	Paul

> Harald
>
> On 08/12/11 05:30, Paul Kyzivat wrote:
>> On 8/11/11 7:01 PM, Igor Faynberg wrote:
>>> Yes, definitely, this is a good course of action--to discuss it with the
>>> authors.
>>>
>>> To this end, I will piggy-back another question.
>>>
>>> Do I understand it right that an endpoint would need to share a secret
>>> with every other endpoint it would need to authenticate or be
>>> authenticated by? If not, how could DH exchange be authenticated (to
>>> avoid MiTM)?
>>
>> Related question:
>>
>> If the trusted server is a conference focus, and there is a single
>> media stream between each participant and the focus, how is this
>> secret exchange managed? each thing you send over that media stream is
>> presumable mixed or otherwise distributed to all the other participants.
>>
>> Thanks,
>> Paul
>>
>>> Igor
>>>
>>> On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
>>>> > So, if we end up with a "trusted MITM," why not have a trusted
>>>> central server in the first place?
>>>>
>>>> draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support
>>>> various other scenarios, but how I understand it, no PBX or any
>>>> intermediary is required.
>>>> The default scenario is e2e.
>>>>
>>>> Maybe the authors copied here can clarify.
>>>>
>>>> Thanks, Henry
>>>>
>>>>
>>>>
>>>> On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com>
>>>> wrote:
>>>>
>>>> Well...
>>>>
>>>> (I thought that the lawful intercept issue that came up several
>>>> times is pertinent here.) But note that this was a parenthetical
>>>> remark!
>>>>
>>>> A non-parenthetical one is based on the quote from the cited draft:
>>>>
>>>>
>>>> "In the development of ZRTP, it was realized that there are scenarios
>>>> in which the media can not be encrypted end-to-end. For example,
>>>> when a client has a trusted server or PBX which provides media
>>>> services in the path. For these cases, ZRTP developed mechanisms for
>>>> handling a "trusted MiTM" which can terminate than reoriginate the
>>>> SRTP encryption."
>>>>
>>>>
>>>> So, if we end up with a "trusted MITM," why not have a trusted
>>>> central server in the first place?
>>>>
>>>> Igor
>>>>
>>>> P.S.: I have nothing against ZRTP, which has done a thorough and
>>>> honest job. I question its applicability to the agreed use cases.
>>>> Personally I think we need an all-encompassing solution.
>>>>
>>>>
>>>> On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>>>>
>>>>
>>>>
>>>> Henry - what sort of mechanism did you have in mind for
>>>> these anonymous
>>>> participants to identify themselves to one another? Are
>>>> you talking
>>>> about the exchange of credentials using the server as an
>>>> intermediary in
>>>> the exchange? Or did you have in mind an exchange outside
>>>> the server.
>>>> (E.g. Bob establishes a secure connection to Alice and
>>>> over it tells her
>>>> "I'm anon2 in conference foo right now"?
>>>>
>>>>
>>>>
>>>> The most credible e2e identification that comes to my mind is Phil
>>>> Zimmermann's Zfone technology:
>>>>
>>>> http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>>>> http://zfoneproject.com/
>>>>
>>>> Definitely not relying on any "trusted" server :-)
>>>>
>>>> Thanks, Henry
>>>>
>>>> On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>>>> <mailto:pkyzivat@alum.mit.edu> wrote:
>>>>
>>>>
>>>>
>>>> On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>>>
>>>>
>>>>
>>>> The participants are identified to
>>>> each other by the central server
>>>>
>>>>
>>>>
>>>> There may be no contradiction, but just to make sure
>>>> the scenario, such as a
>>>> confidential business meeting is included (not
>>>> necessarily a social network
>>>> scenario).
>>>>
>>>> A confidential conference MUST assure that all
>>>> participants can also be
>>>> safely identified first to each other, not necessarily
>>>> by the server, end to
>>>> end, to the full satisfaction of all the participants,
>>>> who would otherwise
>>>> refuse to continue to speak up or continue to attend
>>>> at all.
>>>> Who provides for the e2e identification? A 3rd party?
>>>> Inside the app?
>>>>
>>>> Did I understand correctly there is no contradiction?
>>>> Any problem in clarifying this?
>>>>
>>>>
>>>>
>>>> This is an example of the sort of thing I was speaking of
>>>> in my earlier
>>>> reply - that there are differing levels of identity that
>>>> may be desired.
>>>>
>>>> In some cases, participants may want to be fully
>>>> anonymous. But even
>>>> then, it will typically be important that one participant
>>>> be able to
>>>> identify *which* of the other (anonymous) participants it
>>>> is receiving
>>>> media from. (E.g. by noting which participant in the
>>>> roster is the
>>>> current speaker - anon1, anon2, ...)
>>>>
>>>> In other cases the participants *true* identities are
>>>> hidden, but
>>>> handles/nicknames may be used to identify them in the
>>>> roster and in
>>>> chats. ("snoopy" and "red barron" rather than anon1 and
>>>> anon2.)
>>>>
>>>> But in that case there are issues of how the handles are
>>>> associated with
>>>> participants. It could be that handles are permanently
>>>> assigned to real
>>>> users by the server. Then you can be confident that when
>>>> talking to
>>>> "snoopy" it is always someone well defined - same as the
>>>> last time -
>>>> without awareness of who that person *actually* is.
>>>>
>>>> Alternatively, the server might allow handles to be used
>>>> on a fcfs basis
>>>> per conference. Or it might allow handles to be
>>>> non-unique, so that you
>>>> could have two "snoopy"s at the same time, bound to
>>>> unrelated users.
>>>>
>>>> Henry - what sort of mechanism did you have in mind for
>>>> these anonymous
>>>> participants to identify themselves to one another? Are
>>>> you talking
>>>> about the exchange of credentials using the server as an
>>>> intermediary in
>>>> the exchange? Or did you have in mind an exchange outside
>>>> the server.
>>>> (E.g. Bob establishes a secure connection to Alice and
>>>> over it tells her
>>>> "I'm anon2 in conference foo right now"?
>>>>
>>>> Thanks,
>>>> Paul
>>>>
>>>>
>>>>
>>>> Thanks, Henry
>>>>
>>>>
>>>> On 8/10/11 9:16 AM, "Harald
>>>> Alvestrand"<harald@alvestrand.no>
>>>> <mailto:harald@alvestrand.no> wrote:
>>>>
>>>>
>>>>
>>>> In draft-ietf-rtcweb-use-cases-and-requirements, I
>>>> would like to extend
>>>> one part of the scenario "4.3.3 Video conferencing
>>>> system with central
>>>> server".
>>>>
>>>> I would like to add one more paragraph:
>>>>
>>>> "All participant are authenticated by the central
>>>> server, and authorized
>>>> to connect to the central server. The participants
>>>> are identified to
>>>> each other by the central server, and the
>>>> participants do not have
>>>> access to each others' credentials such as e-mail
>>>> addresses or login IDs".
>>>>
>>>> This is necessary in order to drive use cases that
>>>> resemble Google
>>>> Hangout, where it is a requirement that people are
>>>> able to participate
>>>> without disclosing their Google login IDs to each
>>>> other.
>>>> (in the particular case of Hangout, the display
>>>> name on their profile
>>>> *is* disclosed ... but that's a different matter)
>>>>
>>>> The reason I think this is important is that it
>>>> feeds directly into the
>>>> discussion of what WebRTC needs to authorize: The
>>>> final source or
>>>> destination of media, or the identity of the
>>>> handler at the first hop.
>>>> In at least the case of Hangouts, the requirement
>>>> is to *not* authorize
>>>> the final source or destination.
>>>>
>>>> Not sure yet how to formulate that as a
>>>> requirement, and not sure yet if
>>>> it applies to the cases without a central server,
>>>> such as 4.2.6. We may
>>>> have to decide.
>>>>
>>>> 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
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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 pkyzivat@alum.mit.edu  Fri Aug 12 08:56:40 2011
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Subject: Re: [rtcweb] draft-alvestrand-one-rtp-00.txt ICE usage
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Trimming, and changing subject

On 8/12/11 11:20 AM, Harald Alvestrand wrote:
> On 08/12/11 16:55, Paul Kyzivat wrote:
>> On 8/12/11 2:45 AM, Harald Alvestrand wrote:
>>> On 08/11/11 20:47, Paul Kyzivat wrote:
>>>> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>>>>> I have taken some of the information I learned from the discussions in
>>>>> Quebec City about the issues of putting video and audio into the same
>>>>> RTP session and created a draft from them outlining a solution to the
>>>>> problem of signalling this configuration using SDP.
>>>>>
>>>>> Comments welcome, including the recommended context for processing the
>>>>> document; in a few days I'll send the same note to AVTCORE and/or
>>>>> MMUSIC.
>>>>
>>>> Harald,
>>>>
>>>> A few questions/comments:
>>>>
>>>> 1) If ICE is being used, how do you expect it to be handled on the 2nd
>>>> m-line? Since one of the goals of multiplexing on a single rtp session
>>>> is to minimize ICE processing, one would hope only to do ICE on the
>>>> first of the grouped m-lines. But if the answerer doesn't support this
>>>> grouping, then the ICE will be needed on the others.
>>> I tried to be explicit that if negotiation is successful, only the ICE
>>> parameters from the first section listed in the A=group:TOGETHER would
>>> be used:
>>
>> Sure. But if the other side *doesn't* support TOGETHER, then ICE will
>> need to be negotiated independently for each m-line. And for that to
>> work, the port for the 2nd m-line but be active, other ICE candidates
>> identified and included for it, etc.
>>
>> ISTM the implications of that need further explanation.
>> (I'm not an expert on ICE. If I were, maybe this would be obvious.)
> There are 2 possible things to do for the offerer:
> - Start ICE procedure on all ports when sending the offer
> - Wait until the answer comes back, and either start ICE procedure on
> one port (if the responder understood TOGETHER), or start the ICE
> procedure on all ports (if the responder did not understand TOGETHER)
>
> The first alternative is a number of milliseconds faster, but others
> should chime in on whether that's significant or not.

(I don't know much about ICE, so just call me on it if I say something 
stupid)

IIUC, there are some aspects of ICE that MUST be done before sending the 
offer. Specifically that involved gathering the candidate list, which 
may involve assignment of multiple local ports, interactions with a TURN 
server, etc. I don't think everything can be deferred until the answer 
is received.

(I suppose one approach would be for the initial offer to set the port 
to zero in all but the first m-line. Then, once the first answer is 
received, indicating whether TOGETHER is supported, another offer can be 
generated, using appropriate ICE based on whether using TOGETHER or not. 
But that does start to get complex.)

	Thanks,
	Paul

From alan.b.johnston@gmail.com  Fri Aug 12 08:57:14 2011
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References: <CA69AF47.1CBB0%henry.sinnreich@gmail.com> <4E445F49.6080705@alcatel-lucent.com> <4E449E6B.8020205@alum.mit.edu> <4E450F00.9090908@alvestrand.no> <4E454A35.1020207@alum.mit.edu>
Date: Fri, 12 Aug 2011 10:57:50 -0500
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Subject: Re: [rtcweb] Exchange of trusted identities in server-based conferencing (Re: Use case change request: Identity in multiuser calls)
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ZRTP is much more about media privacy than authentication.  The SRTP
key agreement can be authenticated by checking the SAS out of band,
but this doesn't really tell you who you are talking to, just that
there is no MiTM.  If a shared secret has been cached from a previous
call, then you can tell that it is the same person you talked to
previously, but again, it doesn't say who.  If an end-to-end integrity
protected signaling path was used, you can verify that you are talking
to the other signaling entity, whatever identity that provides.

For a conference call, if the media flows are full mesh
(peer-to-peer), then you could use ZRTP to authenticate each leg of
the conference, although this would be cumbersome.  For a centrally
mixed conference, each media stream is between a participant and the
mixer/focus, who is not a human, so the SAS approaches are out, and
each participant will have a different SAS, so comparing them isn't
possible.

- Alan -

On Fri, Aug 12, 2011 at 10:43 AM, Paul Kyzivat <pkyzivat@alum.mit.edu> wrot=
e:
> On 8/12/11 7:31 AM, Harald Alvestrand wrote:
>>
>> [[ Reminder to all: If you discover that you're talking about something
>> far away from the original topic of the thread, please CHANGE THE
>> SUBJECT LINE! ]]
>>
>> The normal method of authentication for teleconferences today is that
>> people prove they have the conference number (by calling in), and
>> recognize each others' voices.
>> We have security procedures at Google that depend on people recognizing
>> each others' faces (sometimes based on prior meetings, sometimes based
>> on matching with pictures).
>>
>> The digital realm doesn't necessarily have to provide a steel vault
>> where the analog realm is satisfied with plywood - or at least, I think
>> it is not the best for finishing RTCWEB in a reasonable time if we
>> insist that steel vaults be part of the requirements.
>
> AFAIK this thread has evolved from Henry's original question about
> identification of participants in confidential conferences. And perhaps t=
he
> proper answer to that is what you say above. While its an interesting
> theoretical issue, in practice I find even the existing minimal security
> mechanisms to be annoying and almost always unnecessary. (We flatter
> ourselves when we think somebody would be interested in hearing our
> discussions.)
>
> There has been an assertion here that ZRTP is useful for solving this
> problem. I don't know if it is - I have never used ZRTP. But it appears t=
hat
> it might be difficult to make work well in a conference. (Or maybe this h=
as
> already been worked out. If so, that's great.)
>
> =A0 =A0 =A0 =A0Thanks,
> =A0 =A0 =A0 =A0Paul
>
>> Harald
>>
>> On 08/12/11 05:30, Paul Kyzivat wrote:
>>>
>>> On 8/11/11 7:01 PM, Igor Faynberg wrote:
>>>>
>>>> Yes, definitely, this is a good course of action--to discuss it with t=
he
>>>> authors.
>>>>
>>>> To this end, I will piggy-back another question.
>>>>
>>>> Do I understand it right that an endpoint would need to share a secret
>>>> with every other endpoint it would need to authenticate or be
>>>> authenticated by? If not, how could DH exchange be authenticated (to
>>>> avoid MiTM)?
>>>
>>> Related question:
>>>
>>> If the trusted server is a conference focus, and there is a single
>>> media stream between each participant and the focus, how is this
>>> secret exchange managed? each thing you send over that media stream is
>>> presumable mixed or otherwise distributed to all the other participants=
.
>>>
>>> Thanks,
>>> Paul
>>>
>>>> Igor
>>>>
>>>> On 8/11/2011 5:09 PM, Henry Sinnreich wrote:
>>>>>
>>>>> > So, if we end up with a "trusted MITM," why not have a trusted
>>>>> central server in the first place?
>>>>>
>>>>> draft-johnston-rtcweb-media-privacy-00 says ZRTP can also support
>>>>> various other scenarios, but how I understand it, no PBX or any
>>>>> intermediary is required.
>>>>> The default scenario is e2e.
>>>>>
>>>>> Maybe the authors copied here can clarify.
>>>>>
>>>>> Thanks, Henry
>>>>>
>>>>>
>>>>>
>>>>> On 8/11/11 2:39 PM, "Igor Faynberg" <igor.faynberg@alcatel-lucent.com=
>
>>>>> wrote:
>>>>>
>>>>> Well...
>>>>>
>>>>> (I thought that the lawful intercept issue that came up several
>>>>> times is pertinent here.) But note that this was a parenthetical
>>>>> remark!
>>>>>
>>>>> A non-parenthetical one is based on the quote from the cited draft:
>>>>>
>>>>>
>>>>> "In the development of ZRTP, it was realized that there are scenarios
>>>>> in which the media can not be encrypted end-to-end. For example,
>>>>> when a client has a trusted server or PBX which provides media
>>>>> services in the path. For these cases, ZRTP developed mechanisms for
>>>>> handling a "trusted MiTM" which can terminate than reoriginate the
>>>>> SRTP encryption."
>>>>>
>>>>>
>>>>> So, if we end up with a "trusted MITM," why not have a trusted
>>>>> central server in the first place?
>>>>>
>>>>> Igor
>>>>>
>>>>> P.S.: I have nothing against ZRTP, which has done a thorough and
>>>>> honest job. I question its applicability to the agreed use cases.
>>>>> Personally I think we need an all-encompassing solution.
>>>>>
>>>>>
>>>>> On 8/11/2011 3:11 PM, Henry Sinnreich wrote:
>>>>>
>>>>>
>>>>>
>>>>> Henry - what sort of mechanism did you have in mind for
>>>>> these anonymous
>>>>> participants to identify themselves to one another? Are
>>>>> you talking
>>>>> about the exchange of credentials using the server as an
>>>>> intermediary in
>>>>> the exchange? Or did you have in mind an exchange outside
>>>>> the server.
>>>>> (E.g. Bob establishes a secure connection to Alice and
>>>>> over it tells her
>>>>> "I'm anon2 in conference foo right now"?
>>>>>
>>>>>
>>>>>
>>>>> The most credible e2e identification that comes to my mind is Phil
>>>>> Zimmermann's Zfone technology:
>>>>>
>>>>> http://www.ietf.org/id/draft-johnston-rtcweb-media-privacy-00.txt
>>>>> http://zfoneproject.com/
>>>>>
>>>>> Definitely not relying on any "trusted" server :-)
>>>>>
>>>>> Thanks, Henry
>>>>>
>>>>> On 8/11/11 11:50 AM, "Paul Kyzivat" <pkyzivat@alum.mit.edu>
>>>>> <mailto:pkyzivat@alum.mit.edu> wrote:
>>>>>
>>>>>
>>>>>
>>>>> On 8/11/11 12:07 PM, Henry Sinnreich wrote:
>>>>>
>>>>>
>>>>>
>>>>> The participants are identified to
>>>>> each other by the central server
>>>>>
>>>>>
>>>>>
>>>>> There may be no contradiction, but just to make sure
>>>>> the scenario, such as a
>>>>> confidential business meeting is included (not
>>>>> necessarily a social network
>>>>> scenario).
>>>>>
>>>>> A confidential conference MUST assure that all
>>>>> participants can also be
>>>>> safely identified first to each other, not necessarily
>>>>> by the server, end to
>>>>> end, to the full satisfaction of all the participants,
>>>>> who would otherwise
>>>>> refuse to continue to speak up or continue to attend
>>>>> at all.
>>>>> Who provides for the e2e identification? A 3rd party?
>>>>> Inside the app?
>>>>>
>>>>> Did I understand correctly there is no contradiction?
>>>>> Any problem in clarifying this?
>>>>>
>>>>>
>>>>>
>>>>> This is an example of the sort of thing I was speaking of
>>>>> in my earlier
>>>>> reply - that there are differing levels of identity that
>>>>> may be desired.
>>>>>
>>>>> In some cases, participants may want to be fully
>>>>> anonymous. But even
>>>>> then, it will typically be important that one participant
>>>>> be able to
>>>>> identify *which* of the other (anonymous) participants it
>>>>> is receiving
>>>>> media from. (E.g. by noting which participant in the
>>>>> roster is the
>>>>> current speaker - anon1, anon2, ...)
>>>>>
>>>>> In other cases the participants *true* identities are
>>>>> hidden, but
>>>>> handles/nicknames may be used to identify them in the
>>>>> roster and in
>>>>> chats. ("snoopy" and "red barron" rather than anon1 and
>>>>> anon2.)
>>>>>
>>>>> But in that case there are issues of how the handles are
>>>>> associated with
>>>>> participants. It could be that handles are permanently
>>>>> assigned to real
>>>>> users by the server. Then you can be confident that when
>>>>> talking to
>>>>> "snoopy" it is always someone well defined - same as the
>>>>> last time -
>>>>> without awareness of who that person *actually* is.
>>>>>
>>>>> Alternatively, the server might allow handles to be used
>>>>> on a fcfs basis
>>>>> per conference. Or it might allow handles to be
>>>>> non-unique, so that you
>>>>> could have two "snoopy"s at the same time, bound to
>>>>> unrelated users.
>>>>>
>>>>> Henry - what sort of mechanism did you have in mind for
>>>>> these anonymous
>>>>> participants to identify themselves to one another? Are
>>>>> you talking
>>>>> about the exchange of credentials using the server as an
>>>>> intermediary in
>>>>> the exchange? Or did you have in mind an exchange outside
>>>>> the server.
>>>>> (E.g. Bob establishes a secure connection to Alice and
>>>>> over it tells her
>>>>> "I'm anon2 in conference foo right now"?
>>>>>
>>>>> Thanks,
>>>>> Paul
>>>>>
>>>>>
>>>>>
>>>>> Thanks, Henry
>>>>>
>>>>>
>>>>> On 8/10/11 9:16 AM, "Harald
>>>>> Alvestrand"<harald@alvestrand.no>
>>>>> <mailto:harald@alvestrand.no> wrote:
>>>>>
>>>>>
>>>>>
>>>>> In draft-ietf-rtcweb-use-cases-and-requirements, I
>>>>> would like to extend
>>>>> one part of the scenario "4.3.3 Video conferencing
>>>>> system with central
>>>>> server".
>>>>>
>>>>> I would like to add one more paragraph:
>>>>>
>>>>> "All participant are authenticated by the central
>>>>> server, and authorized
>>>>> to connect to the central server. The participants
>>>>> are identified to
>>>>> each other by the central server, and the
>>>>> participants do not have
>>>>> access to each others' credentials such as e-mail
>>>>> addresses or login IDs".
>>>>>
>>>>> This is necessary in order to drive use cases that
>>>>> resemble Google
>>>>> Hangout, where it is a requirement that people are
>>>>> able to participate
>>>>> without disclosing their Google login IDs to each
>>>>> other.
>>>>> (in the particular case of Hangout, the display
>>>>> name on their profile
>>>>> *is* disclosed ... but that's a different matter)
>>>>>
>>>>> The reason I think this is important is that it
>>>>> feeds directly into the
>>>>> discussion of what WebRTC needs to authorize: The
>>>>> final source or
>>>>> destination of media, or the identity of the
>>>>> handler at the first hop.
>>>>> In at least the case of Hangouts, the requirement
>>>>> is to *not* authorize
>>>>> the final source or destination.
>>>>>
>>>>> Not sure yet how to formulate that as a
>>>>> requirement, and not sure yet if
>>>>> it applies to the cases without a central server,
>>>>> such as 4.2.6. We may
>>>>> have to decide.
>>>>>
>>>>> 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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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 randell-ietf@jesup.org  Fri Aug 12 10:27:20 2011
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On 8/12/2011 10:18 AM, Igor Faynberg wrote:

> On 8/12/2011 7:31 AM, Harald Alvestrand wrote:
>> ...
>> The normal method of authentication for teleconferences today is that
>> people prove they have the conference number (by calling in), and
>> recognize each others' voices.
>> ...
> Is there not a conference code, too?  In all (five or six of them that I
> have used) there was one.  This code is, effectively, a shared secret.
> In the service we use, the conference host can create a new code for
> each meeting.  (One problem with relying on recognizing voices is
> passive intruding: I can join your conference and never say a word.)

IMHO that particular level/type of security is not something to be decided by
rtcweb, that's up to the application and server to decide and handle.

At most, I'd suggest an API that lets the application supply a shared secret
(how it's obtained isn't our problem - from the user, from secure email, etc)
to use in generating/validating the keys and/or identities.


-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Fri Aug 12 10:36:07 2011
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On 8/12/2011 11:24 AM, Paul Kyzivat wrote:

> On 8/12/11 2:49 AM, Harald Alvestrand wrote:
>> On 08/12/11 05:37, Ross Finlayson wrote:
>>> Thanks, Harald, for this submission.
>>> A nit (I think). Unless I'm misunderstanding the purpose of the
>>> "a=group:TOGETHER" attribute, the port numbers in the two example
>>> 'answers' are wrong.
>> I'm not sure. These examples came straight out of RFC 4317, including
>> the port numbers; I noted the port number mismatch, and concluded
>> (tentatively) that the port number of an offer means "I'll be using this
>> port", while the port number of an answer means "I'll be using this
>> port" - which means that they SHOULD be expected to be different.
>> When using ICE, the port number is moot anyway (the "candidates"
>> overrides them), so the only purpose of the port number is that the
>> special value zero in an answer means "offer not accepted".
> I had wondered about this too, and forgot to comment.
> IMO the offer is right - you need two different ports in case TOGETHER
> isn't understood by answerer.
> But in the answer, if TOGETHER *is* used, I wonder about the port.
> Putting a unique port there certainly implies to me that it will be
> used. (Though ICE may change things.) But using the *same* port in the
> answer may not be right either. In general, SDP doesn't allow you to use
> the same port in multiple m-lines. IIRC there is an exception to that,
> but I'll have to hunt to find what that is.
> Putting zero as the port in the 2nd m-line of the answer might be a
> solution. It would certainly indicate that only the one port will be
> used. The issue is that when its zero, generally the rest of the stuff
> about that stream is ignored. However, no matter what, *some* rules have
> to be changed to make TOGETHER work.
> I guess a question is: would it ever make sense for the answerer to
> indicate support for TOGETHER, and yet want to refuse one of the m-lines?
Yes.  One m= line is audio, one is video, and I don't support video.  (Or I
do, but can't handle something else about it, or I support it but answered
with an "audio-only" button).

> As long as there are never more than two m-lines combined with TOGETHER,
> refusing one of them can be done by *not* indicating support for
> TOGETHER, and then giving port=0 for the m-lines not desired. And that
> works even if only the 2nd m-line is desired.
> If there were three or more m-lines combined with TOGETHER, it gets more
> complex. (To be concrete, say there was audio, video, and timed text. in
> the offer.) And suppose the answerer only wants audio and timed text. In
> that case I guess it does need a way to reject the video while still
> using TOGETHER. And port=0 for the video is the obvious way.

Agreed.


> That brings up another peculiar case. Suppose the m-lines in question
> were, in order, video, audio, and timed text, and the answerer again
> only wanted to use audio and timed text. In that case, will it still
> want to use the port offered for video??? How would the answer be
> constructed, and where would the answering port be placed?
> It occurs to me that there is another "special" port that might be
> co-opted for use here: port 9, the discard port. Perhaps port 9 could be
> used in the answer, when TOGETHER is used in the answer, to indicate
> m-lines that *are* to be combined with one of the other m-lines, while
> port 0 would be used to indicate m-lines that are being rejected
> entirely. Then there should only be one m-line in the group that has a
> port not equal to either 0 or 9, and it would be the one to use. (Or the
> one to negotiate ICE on.)

Well, since this is a response and we now know both sides support the new
mechanisms, we could special-case another port.  I can't say I love it, but
if there's no better alternative I'd be ok.  I suspect there's a better
alternative though.

-- 
Randell Jesup
randell-ietf@jesup.org


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On 8/12/2011 11:57 AM, Paul Kyzivat wrote:

> Trimming, and changing subject
> On 8/12/11 11:20 AM, Harald Alvestrand wrote:
>> There are 2 possible things to do for the offerer:
>> - Start ICE procedure on all ports when sending the offer
>> - Wait until the answer comes back, and either start ICE procedure on
>> one port (if the responder understood TOGETHER), or start the ICE
>> procedure on all ports (if the responder did not understand TOGETHER)
>> The first alternative is a number of milliseconds faster, but others
>> should chime in on whether that's significant or not.
1/2 RTT is significant especially on slower/longer-distance connections,
since I think all the other setup stuff we're talking about adds up.
We have to make sure that we don't have the "...lo?" problem when
answering an incoming connection.  (Setup delay of more than a couple
hundred ms causes first syllables to be lost.)  It might even be worse on a
computer if the person has a headset on or answers in speakerphone mode with
a button press - you don't have the natural delay of bringing a receiver to
face.

> (I don't know much about ICE, so just call me on it if I say something
> stupid)
> IIUC, there are some aspects of ICE that MUST be done before sending the
> offer. Specifically that involved gathering the candidate list, which
> may involve assignment of multiple local ports, interactions with a TURN
> server, etc. I don't think everything can be deferred until the answer
> is received.
Right; one assumes you've done them.

> (I suppose one approach would be for the initial offer to set the port
> to zero in all but the first m-line. Then, once the first answer is
> received, indicating whether TOGETHER is supported, another offer can be
> generated, using appropriate ICE based on whether using TOGETHER or not.
> But that does start to get complex.)
That forces a full extra RTT plus overhead...  Ouch.

-- 
Randell Jesup
randell-ietf@jesup.org


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On Fri, Aug 12, 2011 at 11:20 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

> On 08/12/11 16:55, Paul Kyzivat wrote:
>
>> On 8/12/11 2:45 AM, Harald Alvestrand wrote:
>>
>>> On 08/11/11 20:47, Paul Kyzivat wrote:
>>>
>>>> On 8/11/11 7:32 AM, Harald Alvestrand wrote:
>>>>
>>>>> I have taken some of the information I learned from the discussions in
>>>>> Quebec City about the issues of putting video and audio into the same
>>>>> RTP session and created a draft from them outlining a solution to the
>>>>> problem of signalling this configuration using SDP.
>>>>>
>>>>> Comments welcome, including the recommended context for processing the
>>>>> document; in a few days I'll send the same note to AVTCORE and/or
>>>>> MMUSIC.
>>>>>
>>>>
>>>> Harald,
>>>>
>>>> A few questions/comments:
>>>>
>>>> 1) If ICE is being used, how do you expect it to be handled on the 2nd
>>>> m-line? Since one of the goals of multiplexing on a single rtp session
>>>> is to minimize ICE processing, one would hope only to do ICE on the
>>>> first of the grouped m-lines. But if the answerer doesn't support this
>>>> grouping, then the ICE will be needed on the others.
>>>>
>>> I tried to be explicit that if negotiation is successful, only the ICE
>>> parameters from the first section listed in the A=group:TOGETHER would
>>> be used:
>>>
>>
>> Sure. But if the other side *doesn't* support TOGETHER, then ICE will need
>> to be negotiated independently for each m-line. And for that to work, the
>> port for the 2nd m-line but be active, other ICE candidates identified and
>> included for it, etc.
>>
>> ISTM the implications of that need further explanation.
>> (I'm not an expert on ICE. If I were, maybe this would be obvious.)
>>
> There are 2 possible things to do for the offerer:
> - Start ICE procedure on all ports when sending the offer
> - Wait until the answer comes back, and either start ICE procedure on one
> port (if the responder understood TOGETHER), or start the ICE procedure on
> all ports (if the responder did not understand TOGETHER)
>
> The first alternative is a number of milliseconds faster, but others should
> chime in on whether that's significant or not.


The former is what is recommended by the rtcp-mux RFC. If we don't do this
(and optimistically only obtain ICE candidates for one transport), we'd have
to do another exchange in the fallback case to provide the additional
candidates.

>
>
>
>>  The following parameters are taken from the first section only:
>>>
>>> o Port number from the m= line
>>>
>>> o All media-level attributes defined in RFC 5245 section 15.1 - this
>>> includes "candidate", "remote-candidates", "ice-mismatch", "ice-
>>> ufrag", "ice-pwd"
>>>
>>> Suggestions for improving this language?
>>>
>>>  2) The draft only shows this mechanism being used to combine one audio
>>>> and one video. Do you also imagine it being used to represent multiple
>>>> audio and/or multiple video streams that might otherwise be handled
>>>> independently? (I see no reason why it couldn't be used that way.)
>>>> This would allow description of different properties for each one.
>>>>
>>> I certainly think that any number of m= sections could be combined this
>>> way, but in other discussions, I've seen people tend towards thinking of
>>> sections as "one m= section for all the audio channels, one m= section
>>> for all the video channels". Can you give an example of the usage you're
>>> thinking of?
>>>
>>
>> If there was an intent to have multiple streams with different properties,
>> then this would provide a way to describe those, that isn't otherwise
>> available. For instance, if you wanted one hi-res video stream and one
>> low-res video stream.
>>
>> I'm also thinking ahead to what might be coming from CLUE (telepresence).
>> It isn't far enough along to yet be able to predict how it will set up its
>> multi-stream calls, but it will probably want to multiplex them over a
>> lesser number of RTP streams. And so I'm shopping for potential mechanisms
>> that might be exploited there.
>>
>>  I'm worried that we may get too many special rules about how parameters
>>> from sections are to be combined - multiple codecs with different PT
>>> numbers are fine, but other parameters could be confusing, so I'd like
>>> to see examples.
>>>
>>
>> Maybe its premature. But no matter what, rules will need to be written on
>> how attributes are combined. So IMO its worthwhile to at least start
>> thinking about how to write those in general, rather than writing a few
>> special case hacks. It may turn out that the rules that are to be followed
>> for combining one audio and one video will also "just work" for combining
>> two audio or two video, etc.
>>
>>    Thanks,
>>    Paul
>>
>>  3) In the example, for an entity that doesn't understand TOGETHER, you
>>>> still show the a=mid lines. This is appropriate for an answerer that
>>>> *does* understand a=group. However an answerer that doesn't implement
>>>> rfc3388 would presumably omit the a=mid lines from the answer. So the
>>>> offerer had better be prepared for that as well.
>>>>
>>> Yes, that's a reasonable point. I'll add a third example answer.
>>>
>>>>
>>>> Thanks,
>>>> Paul
>>>> ______________________________**_________________
>>>> rtcweb mailing list
>>>> rtcweb@ietf.org
>>>> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>>>>
>>>>
>>>
>>>
>>
>>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--0015177407d4e60fbc04aa527d5a
Content-Type: text/html; charset=UTF-8
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<br><br><div class=3D"gmail_quote">On Fri, Aug 12, 2011 at 11:20 AM, Harald=
 Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no">h=
arald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;=
">

<div class=3D"im">On 08/12/11 16:55, Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 8/12/11 2:45 AM, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 08/11/11 20:47, Paul Kyzivat wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 8/11/11 7:32 AM, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I have taken some of the information I learned from the discussions in<br>
Quebec City about the issues of putting video and audio into the same<br>
RTP session and created a draft from them outlining a solution to the<br>
problem of signalling this configuration using SDP.<br>
<br>
Comments welcome, including the recommended context for processing the<br>
document; in a few days I&#39;ll send the same note to AVTCORE and/or<br>
MMUSIC.<br>
</blockquote>
<br>
Harald,<br>
<br>
A few questions/comments:<br>
<br>
1) If ICE is being used, how do you expect it to be handled on the 2nd<br>
m-line? Since one of the goals of multiplexing on a single rtp session<br>
is to minimize ICE processing, one would hope only to do ICE on the<br>
first of the grouped m-lines. But if the answerer doesn&#39;t support this<=
br>
grouping, then the ICE will be needed on the others.<br>
</blockquote>
I tried to be explicit that if negotiation is successful, only the ICE<br>
parameters from the first section listed in the A=3Dgroup:TOGETHER would<br=
>
be used:<br>
</blockquote>
<br>
Sure. But if the other side *doesn&#39;t* support TOGETHER, then ICE will n=
eed to be negotiated independently for each m-line. And for that to work, t=
he port for the 2nd m-line but be active, other ICE candidates identified a=
nd included for it, etc.<br>


<br>
ISTM the implications of that need further explanation.<br>
(I&#39;m not an expert on ICE. If I were, maybe this would be obvious.)<br>
</blockquote></div>
There are 2 possible things to do for the offerer:<br>
- Start ICE procedure on all ports when sending the offer<br>
- Wait until the answer comes back, and either start ICE procedure on one p=
ort (if the responder understood TOGETHER), or start the ICE procedure on a=
ll ports (if the responder did not understand TOGETHER)<br>
<br>
The first alternative is a number of milliseconds faster, but others should=
 chime in on whether that&#39;s significant or not.</blockquote><div><br></=
div><div>The former is what is recommended by the rtcp-mux RFC. If we don&#=
39;t do this (and optimistically only obtain ICE candidates for one transpo=
rt), we&#39;d have to do another exchange in the fallback case to provide t=
he additional candidates.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div><div></div><div class=3D"h5"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
The following parameters are taken from the first section only:<br>
<br>
o Port number from the m=3D line<br>
<br>
o All media-level attributes defined in RFC 5245 section 15.1 - this<br>
includes &quot;candidate&quot;, &quot;remote-candidates&quot;, &quot;ice-mi=
smatch&quot;, &quot;ice-<br>
ufrag&quot;, &quot;ice-pwd&quot;<br>
<br>
Suggestions for improving this language?<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
2) The draft only shows this mechanism being used to combine one audio<br>
and one video. Do you also imagine it being used to represent multiple<br>
audio and/or multiple video streams that might otherwise be handled<br>
independently? (I see no reason why it couldn&#39;t be used that way.)<br>
This would allow description of different properties for each one.<br>
</blockquote>
I certainly think that any number of m=3D sections could be combined this<b=
r>
way, but in other discussions, I&#39;ve seen people tend towards thinking o=
f<br>
sections as &quot;one m=3D section for all the audio channels, one m=3D sec=
tion<br>
for all the video channels&quot;. Can you give an example of the usage you&=
#39;re<br>
thinking of?<br>
</blockquote>
<br>
If there was an intent to have multiple streams with different properties, =
then this would provide a way to describe those, that isn&#39;t otherwise a=
vailable. For instance, if you wanted one hi-res video stream and one low-r=
es video stream.<br>


<br>
I&#39;m also thinking ahead to what might be coming from CLUE (telepresence=
). It isn&#39;t far enough along to yet be able to predict how it will set =
up its multi-stream calls, but it will probably want to multiplex them over=
 a lesser number of RTP streams. And so I&#39;m shopping for potential mech=
anisms that might be exploited there.<br>


<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I&#39;m worried that we may get too many special rules about how parameters=
<br>
from sections are to be combined - multiple codecs with different PT<br>
numbers are fine, but other parameters could be confusing, so I&#39;d like<=
br>
to see examples.<br>
</blockquote>
<br>
Maybe its premature. But no matter what, rules will need to be written on h=
ow attributes are combined. So IMO its worthwhile to at least start thinkin=
g about how to write those in general, rather than writing a few special ca=
se hacks. It may turn out that the rules that are to be followed for combin=
ing one audio and one video will also &quot;just work&quot; for combining t=
wo audio or two video, etc.<br>


<br>
 =C2=A0 =C2=A0Thanks,<br>
 =C2=A0 =C2=A0Paul<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
3) In the example, for an entity that doesn&#39;t understand TOGETHER, you<=
br>
still show the a=3Dmid lines. This is appropriate for an answerer that<br>
*does* understand a=3Dgroup. However an answerer that doesn&#39;t implement=
<br>
rfc3388 would presumably omit the a=3Dmid lines from the answer. So the<br>
offerer had better be prepared for that as well.<br>
</blockquote>
Yes, that&#39;s a reasonable point. I&#39;ll add a third example answer.<br=
>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Thanks,<br>
Paul<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>
<br>
</blockquote>
<br>
<br>
</blockquote>
<br>
<br>
</blockquote>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

--0015177407d4e60fbc04aa527d5a--

From ted.ietf@gmail.com  Fri Aug 12 10:46:53 2011
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In one of the use cases we've discussed, there is an advertisement
placed on a page with something like "click here to chat with someone
now".  In some of the discussion in Quebec city, we circled around the
question of who did the job of connecting the user when there was a
click.  In a side discussion on that topic, Cullen, ekr, and I talked
about possibly making that explicit by using a uri structured along
these lines:

scheme: (e.g. rtw)
authority: the entity which will act as the signaling service for
setting up the session.
path-component-1: the entity to which you will be connected
path-component-2: a token used in the media path to assure you that
media is coming from someone with access to the uri

rtw://outsourced-video.example.com/custchat.carmaker.example;token=2342342433

I don't want to delve into the UI aspects of this too much, but the
presumption I was working under was that if someone clicked on such a
link, it would open a new context (e.g. new browser window), and start
a session. path-component one could be something meaningful only
within a specific context:

rtw://dating-site.example/prospective13344;token=444322111

Alternatively, it could be also meaningful in an external security
context, possibly because custchat.carmaker.example will appear in a
certificate in a known place during the media exchange.  It could also
be a reference to something like a presentity, where the presence
infrastructure provides an out-of-band mechanism for confirming
identity.

Do others think defining something like this would be useful?

If so, would the authority section be a host pointer or an xmpp-style service?
Does the format of path-compenent-1 need to be fixed, or can it be
service-dependent?
Would a token be a pawn-ticket style URI, or would the whole URI be
something that could be bookmarked and re-used?

regards,

Ted

From pkyzivat@alum.mit.edu  Fri Aug 12 10:56:21 2011
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Subject: Re: [rtcweb] draft-alvestrand-one-rtp-00.txt ICE usage
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On 8/12/11 1:41 PM, Randell Jesup wrote:
> On 8/12/2011 11:57 AM, Paul Kyzivat wrote:
>
>> Trimming, and changing subject
>> On 8/12/11 11:20 AM, Harald Alvestrand wrote:
>>> There are 2 possible things to do for the offerer:
>>> - Start ICE procedure on all ports when sending the offer
>>> - Wait until the answer comes back, and either start ICE procedure on
>>> one port (if the responder understood TOGETHER), or start the ICE
>>> procedure on all ports (if the responder did not understand TOGETHER)
>>> The first alternative is a number of milliseconds faster, but others
>>> should chime in on whether that's significant or not.
> 1/2 RTT is significant especially on slower/longer-distance connections,
> since I think all the other setup stuff we're talking about adds up.
> We have to make sure that we don't have the "...lo?" problem when
> answering an incoming connection. (Setup delay of more than a couple
> hundred ms causes first syllables to be lost.) It might even be worse on a
> computer if the person has a headset on or answers in speakerphone mode
> with
> a button press - you don't have the natural delay of bringing a receiver to
> face.

Note that this matters if the extra signaling happens after the callee 
answers. If it happens *before* the answer, then it just extends the 
"ring time" but doesn't cause clipping.

>> (I don't know much about ICE, so just call me on it if I say something
>> stupid)
>> IIUC, there are some aspects of ICE that MUST be done before sending the
>> offer. Specifically that involved gathering the candidate list, which
>> may involve assignment of multiple local ports, interactions with a TURN
>> server, etc. I don't think everything can be deferred until the answer
>> is received.
> Right; one assumes you've done them.
>
>> (I suppose one approach would be for the initial offer to set the port
>> to zero in all but the first m-line. Then, once the first answer is
>> received, indicating whether TOGETHER is supported, another offer can be
>> generated, using appropriate ICE based on whether using TOGETHER or not.
>> But that does start to get complex.)
> That forces a full extra RTT plus overhead... Ouch.

Again, if the extra round trip happens during "alerting" it may not be 
objectionable.

Also, if the first m-line is audio, it can get going sooner, with the 
video to follow with a slight lag.

But I agree this isn't the nicest of solutions. I'm just putting out 
some candidates to consider.

	Thanks,
	Paul



From matthew.kaufman@skype.net  Fri Aug 12 12:03:32 2011
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On 8/12/2011 10:47 AM, Ted Hardie wrote:
> Do others think defining something like this would be useful?

No. For the same reason I don't think that Gmail should be implemented 
by going to "imap:mail.google.com/inbox" and having the browser 
implement a mail reading user interface or it own (or even having a page 
that contains "<showIMAPinterface mailbox='inbox' 
server='mail.google.com'/>").

We should be providing components that allow for all manners for 
real-time communication applications to be built using HTML and 
JavaScript, not baking a phone into the browser.

Matthew Kaufman

From ted.ietf@gmail.com  Fri Aug 12 13:16:42 2011
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On Fri, Aug 12, 2011 at 12:03 PM, Matthew Kaufman
<matthew.kaufman@skype.net> wrote:
> On 8/12/2011 10:47 AM, Ted Hardie wrote:
>>
>> Do others think defining something like this would be useful?
>
> No. For the same reason I don't think that Gmail should be implemented by
> going to "imap:mail.google.com/inbox" and having the browser implement a
> mail reading user interface or it own (or even having a page that contains
> "<showIMAPinterface mailbox='inbox' server='mail.google.com'/>").
>

So, one of the aims here was to allow you to specify what the
signaling path was separate from whom you were calling.  In the case
where they were similar, you'd basically end up with:

rtw:///player23424232;token=234234234

(That is the authority/signaling path would be the same as the current origin).

I'm also not sure that having the ability to create a compact
representation of this equals baking a phone into the browser.  To
take a slightly different example, imagine that I'm using a gaming
site and I want to invite friends to come play the game with me.  A
URI that I could use to paste into a chat window or send in email
would be handy, as a simple way to give an out-of-band pointer to a
rtc-web context.  That would still fire up the full game context,
slotting the user into the game (rather than making a phone call).

Hope that explains the range of things I'd like to cover better.

regards,

Ted


> We should be providing components that allow for all manners for real-time
> communication applications to be built using HTML and JavaScript, not baking
> a phone into the browser.
>
> Matthew Kaufman
>

From christer.holmberg@ericsson.com  Fri Aug 12 15:26:07 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 13 Aug 2011 00:26:42 +0200
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Hi Harald,

When putting the draft together, did you consider intermediaires that do no=
t support the mechanism? Even if the endpoints support it, intermediaries w=
ould still see and expect separate streams.

Regards,

Christer

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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
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<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D448562322-12082011></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
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class=3D448562322-12082011>When putting the draft together, did you conside=
r=20
intermediaires that do not support the mechanism? Even if the endpoints sup=
port=20
it, intermediaries would still see and expect separate=20
streams.</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D448562322-12082011></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D448562322-12082011>Regards,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D448562322-12082011></SPAN></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
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class=3D448562322-12082011>Christer</SPAN></FONT></DIV></BODY></HTML>

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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 13 Aug 2011 00:29:47 +0200
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Hi Harald,

I would like to see some text on, when using the mechanism, how new streams=
 would be added, or existing ones would be removed, later during a session.

Regards,

Christer

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<BODY text=3D#000000 bgColor=3D#ffffff>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff size=3D2><SP=
AN=20
class=3D961132722-12082011>Hi Harald,</SPAN></FONT></DIV>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961132722-12082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>I would like to see some text on, when using the m=
echanism,=20
how new streams would be added, or existing ones would be&nbsp;removed, lat=
er=20
during&nbsp;a session.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961132722-12082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961132722-12082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961132722-12082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D961132722-12082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV></BODY></HTML>

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From harald@alvestrand.no  Sat Aug 13 13:11:03 2011
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On 08/13/11 00:26, Christer Holmberg wrote:
> Hi Harald,
> When putting the draft together, did you consider intermediaires that 
> do not support the mechanism? Even if the endpoints support it, 
> intermediaries would still see and expect separate streams.
I had not really considered this .... I am not convinced it is a problem.
Brief analysis:

Either they are in the media path or they are not in the media path.

If they are not in the media path, they will not see the difference, and 
will not care.

If they are in the media path.... we have a problem, but that same 
problem applies to any other SDP extension that affects the media, such 
as RTCP multiplexing; if that's successful, that's a hopeful sign that 
they either don't exist or aren't important.

(One special form of "intermediares" are wiretap units - they are in 
fact mentioned in the draft...)



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    On 08/13/11 00:26, Christer Holmberg wrote:
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05851DB61820E1@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <meta content="MSHTML 6.00.6000.17098" name="GENERATOR">
      <div dir="ltr" align="left"><font color="#0000ff" face="Arial"
          size="2"><span class="448562322-12082011">Hi Harald,</span></font></div>
      <div dir="ltr" align="left"><font color="#0000ff" face="Arial"
          size="2"><span class="448562322-12082011"></span></font>&nbsp;</div>
      <div dir="ltr" align="left"><font color="#0000ff" face="Arial"
          size="2"><span class="448562322-12082011">When putting the
            draft together, did you consider intermediaires that do not
            support the mechanism? Even if the endpoints support it,
            intermediaries would still see and expect separate streams.</span></font><br>
      </div>
    </blockquote>
    I had not really considered this .... I am not convinced it is a
    problem.<br>
    Brief analysis:<br>
    <br>
    Either they are in the media path or they are not in the media path.<br>
    <br>
    If they are not in the media path, they will not see the difference,
    and will not care.<br>
    <br>
    If they are in the media path.... we have a problem, but that same
    problem applies to any other SDP extension that affects the media,
    such as RTCP multiplexing; if that's successful, that's a hopeful
    sign that they either don't exist or aren't important.<br>
    <br>
    (One special form of "intermediares" are wiretap units - they are in
    fact mentioned in the draft...)<br>
    <br>
    <br>
  </body>
</html>

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Subject: Re: [rtcweb] URIs for rtcweb "calls"
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> rtc-web context.  That would still fire up the full game context,
> slotting the user into the game (rather than making a phone call).

By "full game context" do you mean it would somehow load an http webpage 
with HTML+CSS+JS to handle the signaling? If that's the case, what 
advantages does this offer over normal http[s] URLs (with a path to the 
necessary page and parameters, etc. needed to carry sufficient 
information to establish the session). That certainly seems to already 
cover the case of, "A URI that I could use to paste into a chat window." 
How does it handle all the things that an http[s] URL already provides 
(port, path, caching, proxies, all the associated services built around 
http (e.g. bit.ly), etc.)?

If that's not the case, what do you mean by "context"?

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Subject: Re: [rtcweb] URIs for rtcweb "calls"
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> rtc-web context.  That would still fire up the full game context,
> slotting the user into the game (rather than making a phone call).

By "full game context" do you mean it would somehow load an http webpage 
with HTML+CSS+JS to handle the signaling? If that's the case, what 
advantages does this offer over normal http[s] URLs (with a path to the 
necessary page and parameters, etc. needed to carry sufficient 
information to establish the session). That certainly seems to already 
cover the case of, "A URI that I could use to paste into a chat window." 
How does it handle all the things that an http[s] URL already provides 
(port, path, caching, proxies, all the associated services built around 
http (e.g. bit.ly), etc.)?

If that's not the case, what do you mean by "context"?

From christer.holmberg@ericsson.com  Sun Aug 14 12:12:13 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>
Date: Sun, 14 Aug 2011 21:12:53 +0200
Thread-Topic: draft-alvestrand-one-rtp-00.txt and intermediaires
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Subject: Re: [rtcweb] draft-alvestrand-one-rtp-00.txt and intermediaires
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Hi,

>>When putting the draft together, did you consider intermediaires that do =
not support the mechanism? Even if the >>endpoints support it, intermediari=
es would still see and expect separate streams.
>
>I had not really considered this .... I am not convinced it is a problem.
>Brief analysis:
>
>Either they are in the media path or they are not in the media path.
>
>If they are not in the media path, they will not see the difference, and w=
ill not care.
>
>If they are in the media path.... we have a problem, but that same problem=
 applies to any other SDP extension that >affects the media, such as RTCP m=
ultiplexing; if that's successful, that's a hopeful sign that they either d=
on't exist or >aren't important.

I guess the question is what the impacts are.

In this case, the SDP offer/answer as such may go fine, but the intermediar=
y might reserve bandwidth only for a single media type (the one which the S=
DP answer indicates will eventually be used for the multiplex) - even if th=
at media type will consist of multiple multiplexed media types.=20

Regards,

Christer=

From christer.holmberg@ericsson.com  Sun Aug 14 12:15:47 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: 'Harald Alvestrand' <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sun, 14 Aug 2011 21:13:16 +0200
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Hi Harald,

Section 3 of the draft says that the profile of the sections MUST be the sa=
me, but there is no justification.

Could you please explain the reason behind that MUST?

Regards,

Christer

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From: Harald Alvestrand <harald@alvestrand.no>
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To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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The profile affects the protocols available (such as tmmbr, or whether it is rtp or something completely different. Some parameters may be meaningless under a different profile. So making profile dependent on TOGETHER support looked strange to me. Do you see any case where different profiles would be critical?
-- 
Sent from my Android phone with K-9 Mail. Please excuse my brevity.

Christer Holmberg <christer.holmberg@ericsson.com> wrote:

Hi Harald,

Section 3 of the draft says that the profile of the sections MUST be the same, but there is no justification.

Could you please explain the reason behind that MUST?

Regards,

Christer


------6IMK0FPP6JL64QULE61KWQAKUC5GDB
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<html><head></head><body>The profile affects the protocols available (such as tmmbr, or whether it is rtp or something completely different. Some parameters may be meaningless under a different profile. So making profile dependent on TOGETHER support looked strange to me. Do you see any case where different profiles would be critical?<br>
-- <br>
Sent from my Android phone with K-9 Mail. Please excuse my brevity.<br><br><div class="gmail_quote">Christer Holmberg &lt;christer.holmberg@ericsson.com&gt; wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre style="white-space: pre-wrap; word-wrap:break-word; font-family: sans-serif">Hi Harald,<br /><br />Section 3 of the draft says that the profile of the sections MUST be the same, but there is no justification.<br /><br />Could you please explain the reason behind that MUST?<br /><br />Regards,<br /><br />Christer<br /><br /></pre></blockquote></div></body></html>
------6IMK0FPP6JL64QULE61KWQAKUC5GDB--


From harald@alvestrand.no  Sun Aug 14 23:42:02 2011
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Subject: Re: [rtcweb] Dual stack RTCweb
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On 08/12/2011 04:39 PM, Olle E. Johansson wrote:
> Coming in late to the discussion, I can't find any traces of a discussion about dual stacks. Are the issues related with this supposed to be solved with ICE/STUN/Turn - if so, shouldn't this be documented?
I think ICE/STUN is supposed to solve the issue of setting up a 
connection that is either IPv6 or IPv4. Bernard Aboba mentioned in 
Quebec that he had some experience in this area - Bernard, can you share?
> If not, what do we do? Happy RTC-eyeballs?
>
> /O
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From harald@alvestrand.no  Sun Aug 14 23:50:44 2011
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Date: Mon, 15 Aug 2011 08:51:26 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] URIs for rtcweb "calls"
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Two points on this:

1) An URI that is human-visible (for example in hover-over) creates the 
expectation that you can take this URI, plug it into another Web 
browser, and achieve a sensible result.

For many reasons, some of them involving branding of experience, which 
usually requires a ton of Javascript and CSS, I think the more likely 
URL is 
https://joes-outsourced-video.example.com/customerchat.html?callto=custchat.carmaker.example&token=234234234, 
which would pop up the UI that users of joes-outsourced-video has been 
led to expect. It would not contain any way of figuring out who you're 
calling, or rather the scheme would vary between providers - you have to 
trust in-call mechanisms (and Joe) for that.

2) An URI that is not human-visible (the blob: URI of the filesystem API 
spec comes to mind) may be an useful shorthand for a set of call 
parameters passed across internal APIs; I set out to create such a thing 
in draft-alvestrand-rtcweb-datagrams, but at the moment this seems to 
have been overtaken by events, and the smallest objects we're talking 
about for specifying such parameter sets is a SDP blob, which is a 
*little* big for fitting into the URI straitjacket.

My $0.02.

                         Harald

On 08/12/2011 07:47 PM, Ted Hardie wrote:
> In one of the use cases we've discussed, there is an advertisement
> placed on a page with something like "click here to chat with someone
> now".  In some of the discussion in Quebec city, we circled around the
> question of who did the job of connecting the user when there was a
> click.  In a side discussion on that topic, Cullen, ekr, and I talked
> about possibly making that explicit by using a uri structured along
> these lines:
>
> scheme: (e.g. rtw)
> authority: the entity which will act as the signaling service for
> setting up the session.
> path-component-1: the entity to which you will be connected
> path-component-2: a token used in the media path to assure you that
> media is coming from someone with access to the uri
>
> rtw://outsourced-video.example.com/custchat.carmaker.example;token=2342342433
>
> I don't want to delve into the UI aspects of this too much, but the
> presumption I was working under was that if someone clicked on such a
> link, it would open a new context (e.g. new browser window), and start
> a session. path-component one could be something meaningful only
> within a specific context:
>
> rtw://dating-site.example/prospective13344;token=444322111
>
> Alternatively, it could be also meaningful in an external security
> context, possibly because custchat.carmaker.example will appear in a
> certificate in a known place during the media exchange.  It could also
> be a reference to something like a presentity, where the presence
> infrastructure provides an out-of-band mechanism for confirming
> identity.
>
> Do others think defining something like this would be useful?
>
> If so, would the authority section be a host pointer or an xmpp-style service?
> Does the format of path-compenent-1 need to be fixed, or can it be
> service-dependent?
> Would a token be a pawn-ticket style URI, or would the whole URI be
> something that could be bookmarked and re-used?
>
> regards,
>
> Ted
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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I added this text in a non-normative appendix:

This document does not specify anything about RTCP sessions. Since a 
purpose of the specification is to minimize the number of transport 
connections, it's natural to use the "a=rtcp-mux" attribute defined in 
[RFC5761], but this document does not specify anything about that matter.

On 08/12/11 15:09, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> A few comments:
>
> 1) Section 4 has this:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> session, and the parameters for the other sections MUST be ignored.
>
> I think it needs to be:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> and RCP session, and the parameters for the other sections MUST be
>
> ignored.
>
> 2) When the RTCP port is not the next higher (odd) port number 
> following the RTP port described in the m-line, the "a=rtcp" attribute 
> (defined in RFC-3605) is used to signal it, as in:
>
> m=audio 49170 RTP/AVP 0
>
> a=rtcp:53020 IN IP4 126.16.64.4
>
> With the grouping semantics described in the draft, it seems this RTCP 
> port should be taken only from the first section. So, this could be 
> could be added to section 4.
>
> 3) RTCP could be multiplexed with RTP on a single port and signaled 
> using the "a=rtcp-mux" attribute (defined in RFC 5761). When it is 
> used together with the grouping semantics described in the draft, it 
> seems all RTCP sessions of that group need to be multiplexed on the 
> RTP port from the first section.
>
> Muthu
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Harald Alvestrand
> *Sent:* Thursday, August 11, 2011 5:02 PM
> *To:* rtcweb@ietf.org
> *Subject:* [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> I have taken some of the information I learned from the discussions in 
> Quebec City about the issues of putting video and audio into the same 
> RTP session and created a draft from them outlining a solution to the 
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the 
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.
>
>                       Harald
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> I-D Action: draft-alvestrand-one-rtp-00.txt
>
> *Date: *
>
> 	
>
> Thu, 11 Aug 2011 04:28:09 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *Reply-To: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   
>         Title            : SDP Grouping for Single RTP Sessions
>         Author(s)        : Harald Tveit Alvestrand
>         Filename         : draft-alvestrand-one-rtp-00.txt
>         Pages            : 8
>         Date             : 2011-08-11
>   
>     This document describes an extension to the Session Description
>     Protocol (SDP) to describe RTP sessions where media of multiple top
>     level types, for example audio and video, are carried in the same RTP
>     session.
>   
>     This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
>     consideration.
>   
>   
>   
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org  <mailto:I-D-Announce@ietf.org>
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:http://www.ietf.org/shadow.html
> orftp://ftp.ietf.org/ietf/1shadow-sites.txt
>   


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    I added this text in a non-normative appendix:<br>
    <br>
    This document does not specify anything about RTCP sessions. Since a
    purpose of the specification is to minimize the number of transport
    connections, it's natural to use the "a=rtcp-mux" attribute defined
    in [RFC5761], but this document does not specify anything about that
    matter.<br>
    <br>
    On 08/12/11 15:09, Muthu Arul Mozhi Perumal (mperumal) wrote:
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625F795@XMB-BGL-414.cisco.com"
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">A
            few comments:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">1)
            Section 4 has this:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the responder understands the semantics of the TOGETHER
            extension,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">the
            parameters of the first section MUST be used to establish
            the <span class="SpellE">RTP</span><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">session,
            and the parameters for the other sections MUST be ignored.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">I
            think it needs to be:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the responder understands the semantics of the TOGETHER
            extension,<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">the
            parameters of the first section MUST be used to establish
            the <span class="SpellE">RTP</span><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">and
            <span class="SpellE">RCP</span> session, and the parameters
            for the other
            sections MUST be <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">ignored.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">2)
            When the <span class="SpellE">RTCP</span> port is not the
            next higher (odd) port
            number following the <span class="SpellE">RTP</span> port
            described in the m-line,
            the "a=<span class="SpellE">rtcp</span>" attribute (defined
            in <span class="SpellE">RFC</span>-3605) is used to signal
            it, as in:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">m=audio
            49170 <span class="SpellE">RTP</span>/<span class="SpellE">AVP</span>
            0<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">a=<span
              class="SpellE">rtcp:53020</span> IN <span class="SpellE">IP4</span>
            126.16.64.4<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">With
            the grouping semantics described in the draft, it seems this
            <span class="SpellE">RTCP</span> port should be taken only
            from the first section. So,
            this could be could be added to section 4.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">3)
            <span class="SpellE">RTCP</span> could be multiplexed with <span
              class="SpellE">RTP</span>
            on a single port and signaled using the "a=<span
              class="SpellE">rtcp-mux</span>"
            attribute (defined in <span class="SpellE">RFC</span>
            5761). When it is used
            together with the grouping semantics described in the draft,
            it seems all <span class="SpellE">RTCP</span> sessions of
            that group need to be multiplexed on the <span
              class="SpellE">RTP</span> port from the first section.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Muthu<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <div style="border-width: medium medium medium 1.5pt;
          border-style: none none none solid; border-color:
          -moz-use-text-color -moz-use-text-color -moz-use-text-color
          blue; padding: 0in 0in 0in 4pt;">
          <div>
            <div style="border-right: medium none; border-width: 1pt
              medium medium; border-style: solid none none;
              border-color: rgb(181, 196, 223) -moz-use-text-color
              -moz-use-text-color; padding: 3pt 0in 0in;">
              <p class="MsoNormal"><b><span style="font-size: 10pt;
                    font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] <b>On Behalf Of </b>Harald
                  Alvestrand<br>
                  <b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
                  <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> [rtcweb] Fwd: I-D Action:
                  draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><span style="">I
              have taken some of the information I learned from the
              discussions in Quebec
              City about the issues of putting video and audio into the
              same RTP session and
              created a draft from them outlining a solution to the
              problem of signalling
              this configuration using SDP.<br>
              <br>
              Comments welcome, including the recommended context for
              processing the
              document; in a few days I'll send the same note to AVTCORE
              and/or MMUSIC.<br>
              <br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              Harald<br>
              <br>
              -------- Original Message -------- <o:p></o:p></span></p>
          <table class="MsoNormalTable" style="" border="0"
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              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Subject: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style="">I-D Action:
                      draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Date: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style="">Thu, 11 Aug 2011
                      04:28:09 -0700<o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">From: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">Reply-To: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></span></p>
                </td>
              </tr>
              <tr style="">
                <td style="padding: 0in;" nowrap="nowrap" valign="top">
                  <p class="MsoNormal" style="text-align: right;"
                    align="right"><b><span style="">To: <o:p></o:p></span></b></p>
                </td>
                <td style="padding: 0in;">
                  <p class="MsoNormal"><span style=""><a
                        moz-do-not-send="true"
                        href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p></span></p>
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          <p class="MsoNormal" style="margin-bottom: 12pt;"><span
              style=""><o:p>&nbsp;</o:p></span></p>
          <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Title<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: SDP Grouping for Single RTP Sessions<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Author(s)<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: Harald Tveit Alvestrand<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Filename<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Pages<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: 8<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Date<span style="">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>: 2011-08-11<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>This document describes an extension to the Session Description<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>Protocol (SDP) to describe RTP sessions where media of multiple top<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>level types, for example audio and video, are carried in the same RTP<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>session.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for<o:p></o:p></pre>
          <pre><span style="">&nbsp;&nbsp; </span>consideration.<o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>A URL for this Internet-Draft is:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
          <pre>This Internet-Draft can be retrieved at:<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
          <pre>_______________________________________________<o:p></o:p></pre>
          <pre>I-D-Announce mailing list<o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p></pre>
          <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre>
          <pre>Internet-Draft directories: <a moz-do-not-send="true" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a><o:p></o:p></pre>
          <pre>or <a moz-do-not-send="true" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><o:p></o:p></pre>
          <pre><o:p>&nbsp;</o:p></pre>
        </div>
      </div>
    </blockquote>
    <br>
  </body>
</html>

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From christer.holmberg@ericsson.com  Mon Aug 15 00:56:55 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 15 Aug 2011 09:57:36 +0200
Thread-Topic: draft-alvestrand-one-rtp-00.txt and profile
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Hi,

There are discussions in other SDOs about using AVPF for video and AVP for =
audio, within a session.

However, those discussions are not within the context of multiplex.

Regards,

Christer

________________________________
From: Harald Alvestrand [mailto:harald@alvestrand.no]
Sent: 14. elokuuta 2011 23:15
To: Christer Holmberg; rtcweb@ietf.org
Subject: RE: draft-alvestrand-one-rtp-00.txt and profile

The profile affects the protocols available (such as tmmbr, or whether it i=
s rtp or something completely different. Some parameters may be meaningless=
 under a different profile. So making profile dependent on TOGETHER support=
 looked strange to me. Do you see any case where different profiles would b=
e critical?
--
Sent from my Android phone with K-9 Mail. Please excuse my brevity.

Christer Holmberg <christer.holmberg@ericsson.com> wrote:

Hi Harald,

Section 3 of the draft says that the profile of the sections MUST be the sa=
me, but there is no justification.

Could you please explain the reason behind that MUST?

Regards,

Christer


--_000_7F2072F1E0DE894DA4B517B93C6A05852203CD99E6ESESSCMS0356e_
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.6002.18457" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><FONT face=3DArial color=3D#0000ff=20
size=3D2></FONT>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>There are discussions in other SDOs about using AV=
PF for=20
video and AVP for audio, within a session.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>However, those discussions are not within the cont=
ext of=20
multiplex.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D898335607-15082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Harald Alvestrand=20
  [mailto:harald@alvestrand.no] <BR><B>Sent:</B> 14. elokuuta 2011=20
  23:15<BR><B>To:</B> Christer Holmberg; rtcweb@ietf.org<BR><B>Subject:</B>=
 RE:=20
  draft-alvestrand-one-rtp-00.txt and profile<BR></FONT><BR></DIV>
  <DIV></DIV>The profile affects the protocols available (such as tmmbr, or=
=20
  whether it is rtp or something completely different. Some parameters may =
be=20
  meaningless under a different profile. So making profile dependent on TOG=
ETHER=20
  support looked strange to me. Do you see any case where different profile=
s=20
  would be critical?<BR>-- <BR>Sent from my Android phone with K-9 Mail. Pl=
ease=20
  excuse my brevity.<BR><BR>
  <DIV class=3Dgmail_quote>Christer Holmberg=20
  &lt;christer.holmberg@ericsson.com&gt; wrote:
  <BLOCKQUOTE class=3Dgmail_quote=20
  style=3D"PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(2=
04,204,204) 1px solid"><PRE style=3D"FONT-FAMILY: sans-serif; WORD-WRAP: br=
eak-word">Hi Harald,<BR><BR>Section 3 of the draft says that the profile of=
 the sections MUST be the same, but there is no justification.<BR><BR>Could=
 you please explain the reason behind that MUST?<BR><BR>Regards,<BR><BR>Chr=
ister<BR><BR></PRE></BLOCKQUOTE></DIV></BLOCKQUOTE></BODY></HTML>

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From harald@alvestrand.no  Mon Aug 15 01:19:43 2011
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Subject: Re: [rtcweb] I-D Action: draft-alvestrand-one-rtp-00.txt
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On 08/12/11 17:24, Paul Kyzivat wrote:
> On 8/12/11 2:49 AM, Harald Alvestrand wrote:
>> On 08/12/11 05:37, Ross Finlayson wrote:
>>> Thanks, Harald, for this submission.
>>>
>>> A nit (I think). Unless I'm misunderstanding the purpose of the
>>> "a=group:TOGETHER" attribute, the port numbers in the two example
>>> 'answers' are wrong.
>> I'm not sure. These examples came straight out of RFC 4317, including
>> the port numbers; I noted the port number mismatch, and concluded
>> (tentatively) that the port number of an offer means "I'll be using this
>> port", while the port number of an answer means "I'll be using this
>> port" - which means that they SHOULD be expected to be different.
>>
>> When using ICE, the port number is moot anyway (the "candidates"
>> overrides them), so the only purpose of the port number is that the
>> special value zero in an answer means "offer not accepted".
>
> I had wondered about this too, and forgot to comment.
> IMO the offer is right - you need two different ports in case TOGETHER 
> isn't understood by answerer.
>
> But in the answer, if TOGETHER *is* used, I wonder about the port. 
> Putting a unique port there certainly implies to me that it will be used.
I don't see such an implication. Numbers are cheap, and having an unique 
number avoids having to specify specific rules for this issue.
> (Though ICE may change things.) But using the *same* port in the 
> answer may not be right either. In general, SDP doesn't allow you to 
> use the same port in multiple m-lines. IIRC there is an exception to 
> that, but I'll have to hunt to find what that is.
The grouping RFC explicitly calls out the prohibition. So no luck there.
>
> Putting zero as the port in the 2nd m-line of the answer might be a 
> solution. It would certainly indicate that only the one port will be 
> used. The issue is that when its zero, generally the rest of the stuff 
> about that stream is ignored. However, no matter what, *some* rules 
> have to be changed to make TOGETHER work.
>
> I guess a question is: would it ever make sense for the answerer to 
> indicate support for TOGETHER, and yet want to refuse one of the m-lines?
>
> As long as there are never more than two m-lines combined with 
> TOGETHER, refusing one of them can be done by *not* indicating support 
> for TOGETHER, and then giving port=0 for the m-lines not desired. And 
> that works even if only the 2nd m-line is desired.
>
> If there were three or more m-lines combined with TOGETHER, it gets 
> more complex. (To be concrete, say there was audio, video, and timed 
> text. in the offer.) And suppose the answerer only wants audio and 
> timed text. In that case I guess it does need a way to reject the 
> video while still using TOGETHER. And port=0 for the video is the 
> obvious way.
Yep, that makes sense. Another way, which could be equally "obvious", 
would be to accept the section, but just not put any a=rtpmap: lines in 
it - this too "looks a bit odd", so I'm putting the "port zero" option 
in the draft.
>
> That brings up another peculiar case. Suppose the m-lines in question 
> were, in order, video, audio, and timed text, and the answerer again 
> only wanted to use audio and timed text. In that case, will it still 
> want to use the port offered for video??? How would the answer be 
> constructed, and where would the answering port be placed?
That is a strong argument in favour of "just don't give any codecs". I'm 
adding that too.
>
> It occurs to me that there is another "special" port that might be 
> co-opted for use here: port 9, the discard port. Perhaps port 9 could 
> be used in the answer, when TOGETHER is used in the answer, to 
> indicate m-lines that *are* to be combined with one of the other 
> m-lines, while port 0 would be used to indicate m-lines that are being 
> rejected entirely. Then there should only be one m-line in the group 
> that has a port not equal to either 0 or 9, and it would be the one to 
> use. (Or the one to negotiate ICE on.)
Magic numbers are bad for technological health. I don't think I'd like 
to see that.
>
>     Thanks,
>     Paul


From harald@alvestrand.no  Mon Aug 15 01:32:56 2011
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Subject: Re: [rtcweb] draft-alvestrand-one-rtp-00.txt and profile
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On 08/15/11 09:57, Christer Holmberg wrote:
> Hi,
> There are discussions in other SDOs about using AVPF for video and AVP 
> for audio, within a session.
> However, those discussions are not within the context of multiplex.
I added the following text:

The reason for the requirement for systematic proto is that there are 
many combinations that don't make sense (for instance "RTP/AVPF" in one 
section and "RTP/SAVP" in another would make encryption and availability 
of TMMBR depend on the outcome of negotiation, which seems strange). The 
cases where combinations make sense (RTP/AVPF with UDP/FEC for instance) 
also usually require that separate RTP sessions be used. [[QUESTION IN 
DRAFT: Are there sensible combinations?]]

Makes sense?
> Regards,
> Christer
>
>     ------------------------------------------------------------------------
>     *From:* Harald Alvestrand [mailto:harald@alvestrand.no]
>     *Sent:* 14. elokuuta 2011 23:15
>     *To:* Christer Holmberg; rtcweb@ietf.org
>     *Subject:* RE: draft-alvestrand-one-rtp-00.txt and profile
>
>     The profile affects the protocols available (such as tmmbr, or
>     whether it is rtp or something completely different. Some
>     parameters may be meaningless under a different profile. So making
>     profile dependent on TOGETHER support looked strange to me. Do you
>     see any case where different profiles would be critical?
>     -- 
>     Sent from my Android phone with K-9 Mail. Please excuse my brevity.
>
>     Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>
>         Hi Harald,
>
>         Section 3 of the draft says that the profile of the sections MUST be the same, but there is no justification.
>
>         Could you please explain the reason behind that MUST?
>
>         Regards,
>
>         Christer
>
>


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    On 08/15/11 09:57, Christer Holmberg wrote:
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05852203CD99E6@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=ISO-8859-1">
      <meta content="MSHTML 6.00.6002.18457" name="GENERATOR">
      <div dir="ltr" align="left">&nbsp;</div>
      <div dir="ltr" align="left"><span class="898335607-15082011"><font
            color="#0000ff" face="Arial" size="2">Hi,</font></span></div>
      <div dir="ltr" align="left"><span class="898335607-15082011"></span>&nbsp;</div>
      <div dir="ltr" align="left"><span class="898335607-15082011"><font
            color="#0000ff" face="Arial" size="2">There are discussions
            in other SDOs about using AVPF for video and AVP for audio,
            within a session.</font></span></div>
      <div dir="ltr" align="left"><span class="898335607-15082011"></span>&nbsp;</div>
      <div dir="ltr" align="left"><span class="898335607-15082011"><font
            color="#0000ff" face="Arial" size="2">However, those
            discussions are not within the context of multiplex.</font></span></div>
    </blockquote>
    I added the following text:<br>
    <br>
    The reason for the requirement for systematic proto is that there
    are many combinations that don't make sense (for instance "RTP/AVPF"
    in one section and "RTP/SAVP" in another would make encryption and
    availability of TMMBR depend on the outcome of negotiation, which
    seems strange). The cases where combinations make sense (RTP/AVPF
    with UDP/FEC for instance) also usually require that separate RTP
    sessions be used. [[QUESTION IN DRAFT: Are there sensible
    combinations?]]<br>
    <br>
    Makes sense?<br>
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05852203CD99E6@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
      <div dir="ltr" align="left"><span class="898335607-15082011"></span>&nbsp;</div>
      <div dir="ltr" align="left"><span class="898335607-15082011"><font
            color="#0000ff" face="Arial" size="2">Regards,</font></span></div>
      <div dir="ltr" align="left"><span class="898335607-15082011"></span>&nbsp;</div>
      <div dir="ltr" align="left"><span class="898335607-15082011"><font
            color="#0000ff" face="Arial" size="2">Christer</font></span></div>
      <br>
      <blockquote dir="ltr" style="padding-left: 5px; margin-left: 5px;
        border-left: 2px solid rgb(0, 0, 255); margin-right: 0px;">
        <div class="OutlookMessageHeader" dir="ltr" align="left"
          lang="en-us">
          <hr tabindex="-1"> <font face="Tahoma" size="2"><b>From:</b>
            Harald Alvestrand [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
            <b>Sent:</b> 14. elokuuta 2011 23:15<br>
            <b>To:</b> Christer Holmberg; <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <b>Subject:</b> RE: draft-alvestrand-one-rtp-00.txt and
            profile<br>
          </font><br>
        </div>
        The profile affects the protocols available (such as tmmbr, or
        whether it is rtp or something completely different. Some
        parameters may be meaningless under a different profile. So
        making profile dependent on TOGETHER support looked strange to
        me. Do you see any case where different profiles would be
        critical?<br>
        -- <br>
        Sent from my Android phone with K-9 Mail. Please excuse my
        brevity.<br>
        <br>
        <div class="gmail_quote">Christer Holmberg
          <a class="moz-txt-link-rfc2396E" href="mailto:christer.holmberg@ericsson.com">&lt;christer.holmberg@ericsson.com&gt;</a> wrote:
          <blockquote class="gmail_quote" style="padding-left: 1ex;
            margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204,
            204, 204);">
            <pre style="font-family: sans-serif; word-wrap: break-word;">Hi Harald,

Section 3 of the draft says that the profile of the sections MUST be the same, but there is no justification.

Could you please explain the reason behind that MUST?

Regards,

Christer

</pre>
          </blockquote>
        </div>
      </blockquote>
    </blockquote>
    <br>
  </body>
</html>

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From bernard_aboba@hotmail.com  Mon Aug 15 02:23:52 2011
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ICE is designed to handle dual stack operation.  It can even deal with NAT6=
4=2C if correctly implemented.=20

Since ICE tests pairs before using them=2C there is no "happy eyeballs" pro=
blem with ICE (e.g. if an IPv6 route doesn't exist=2C the test will fail).=
=20

That said=2C IPv6/IPv4 priorities may need some adjustment in some situatio=
ns (e.g. if the IPv6 routes are more circuitous than IPv4=2C it may not mak=
e sense to prefer IPv6 over IPv4). =20

That is one of the reasons that I am not enthusiastic about treating the SD=
P JSON blob as "opaque" (e.g. not subject to adjustment). Doing so without =
the ability
to adjust IPv6/IPv4 priorities will result in poor performance in dual stac=
k operation in some cases.=20


 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
ICE is designed to handle dual stack operation.&nbsp=3B It can even deal wi=
th NAT64=2C if correctly implemented. <br><br>Since ICE tests pairs before =
using them=2C there is no "happy eyeballs" problem with ICE (e.g. if an IPv=
6 route doesn't exist=2C the test will fail). <br><br>That said=2C IPv6/IPv=
4 priorities may need some adjustment in some situations (e.g. if the IPv6 =
routes are more circuitous than IPv4=2C it may not make sense to prefer IPv=
6 over IPv4).&nbsp=3B <br><br>That is one of the reasons that I am not enth=
usiastic about treating the SDP JSON blob as "opaque" (e.g. not subject to =
adjustment). Doing so without the ability<br>to adjust IPv6/IPv4 priorities=
 will result in poor performance in dual stack operation in some cases. <br=
><br><div><br></div> 		 	   		  </div></body>
</html>=

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On 8/15/2011 4:20 AM, Harald Alvestrand wrote:

> On 08/12/11 17:24, Paul Kyzivat wrote:
>> That brings up another peculiar case. Suppose the m-lines in question
>> were, in order, video, audio, and timed text, and the answerer again
>> only wanted to use audio and timed text. In that case, will it still
>> want to use the port offered for video??? How would the answer be
>> constructed, and where would the answering port be placed?
> That is a strong argument in favour of "just don't give any codecs". I'm
> adding that too.

If you're suggesting "m=video 4321 RTP/AVP" as an SDP line, it's illegal to
not have a payload listed, I'm afraid (though many implementations will 
accept
it).  That includes when you reject an m= line with port 0.

>> It occurs to me that there is another "special" port that might be
>> co-opted for use here: port 9, the discard port.
> Magic numbers are bad for technological health. I don't think I'd like
> to see that.

Agreed, though occasionally it's ok, even if not optimal (port 0 for 
reject for example).


-- 
Randell Jesup
randell-ietf@jesup.org


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On 08/15/11 16:52, Randell Jesup wrote:
> On 8/15/2011 4:20 AM, Harald Alvestrand wrote:
>
>> On 08/12/11 17:24, Paul Kyzivat wrote:
>>> That brings up another peculiar case. Suppose the m-lines in question
>>> were, in order, video, audio, and timed text, and the answerer again
>>> only wanted to use audio and timed text. In that case, will it still
>>> want to use the port offered for video??? How would the answer be
>>> constructed, and where would the answering port be placed?
>> That is a strong argument in favour of "just don't give any codecs". I'm
>> adding that too.
>
> If you're suggesting "m=video 4321 RTP/AVP" as an SDP line, it's 
> illegal to
> not have a payload listed, I'm afraid (though many implementations 
> will accept
> it).  That includes when you reject an m= line with port 0.
Hm. Yes, since a=rtpmap attributes are only required for dynamic 
numbers, leaving all the a=rtpmap: attributes out doesn't help, and the 
grammar of the m= line is given in RFC 4566 as

       m=<media> <port> <proto> <fmt> ...

  which seems like "one or more". So scratch that idea.
>
>>> It occurs to me that there is another "special" port that might be
>>> co-opted for use here: port 9, the discard port.
>> Magic numbers are bad for technological health. I don't think I'd like
>> to see that.
>
> Agreed, though occasionally it's ok, even if not optimal (port 0 for 
> reject for example).
>
>
Yup, zero and one are kind of OK to have magical.

From oej@edvina.net  Mon Aug 15 08:13:24 2011
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Subject: Re: [rtcweb] Dual stack RTCweb
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15 aug 2011 kl. 11:24 skrev Bernard Aboba:

> ICE is designed to handle dual stack operation.  It can even deal with =
NAT64, if correctly implemented.=20
>=20
> Since ICE tests pairs before using them, there is no "happy eyeballs" =
problem with ICE (e.g. if an IPv6 route doesn't exist, the test will =
fail).=20
>=20
> That said, IPv6/IPv4 priorities may need some adjustment in some =
situations (e.g. if the IPv6 routes are more circuitous than IPv4, it =
may not make sense to prefer IPv6 over IPv4). =20
>=20
> That is one of the reasons that I am not enthusiastic about treating =
the SDP JSON blob as "opaque" (e.g. not subject to adjustment). Doing so =
without the ability
> to adjust IPv6/IPv4 priorities will result in poor performance in dual =
stack operation in some cases.=20
>=20
>=20

Well, if we use SRV records, the receiving part may use SRV to indicate =
preference.

Now, if I have an IPv6 only client ICE won't help unless we force turn =
usage for IPv6 only clients so that they offer dual stack always (like =
the current recommendation for SIP).

/O=

From ted.ietf@gmail.com  Mon Aug 15 10:53:37 2011
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From: Ted Hardie <ted.ietf@gmail.com>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>
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Subject: Re: [rtcweb] URIs for rtcweb "calls"
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On Sun, Aug 14, 2011 at 4:45 AM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
> By "full game context" do you mean it would somehow load an http webpage
> with HTML+CSS+JS to handle the signaling? If that's the case, what
> advantages does this offer over normal http[s] URLs (with a path to the
> necessary page and parameters, etc. needed to carry sufficient information
> to establish the session). That certainly seems to already cover the case
> of, "A URI that I could use to paste into a chat window." How does it handle
> all the things that an http[s] URL already provides (port, path, caching,
> proxies, all the associated services built around http (e.g. bit.ly), etc.)?
>

As I mentioned in my response to Matthew, I'm thinking abut a range of
potential use cases. For the gaming site example where a full web
context is created, I agree that an HTTPS URI could do the same thing.
 You do get some minor advantages in using a distinct URI scheme,
primarily in early identification that the resulting context will be a
rtcweb context.  This might allow you apply permissions early, like a
parental permission against use of rtcweb, or to allow the device to
start setting up elements of its local context early.  It also allows
you to standardize how you to identify the signaling context and
target entity, which I doubt you could do in HTTPS URIs.  The
arguments for and against scheme proliferation have been going for
quite a while now, of course, so I understand that many people would
prefer that there be only HTTPS URIs here.

For the case where you are setting up something closer to a
web-chat-with-an-agent-for-the-ad-seen here, I think the amount of
page context will be very small, and that it will be closer to the
experience of the browser/app setting up its default widgets for this.

Just my personal opinion,

Ted






> If that's not the case, what do you mean by "context"?
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From prvs=201764c72=tterriberry@mozilla.com  Mon Aug 15 11:33:34 2011
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Subject: Re: [rtcweb] URIs for rtcweb "calls"
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> For the case where you are setting up something closer to a
> web-chat-with-an-agent-for-the-ad-seen here, I think the amount of
> page context will be very small, and that it will be closer to the
> experience of the browser/app setting up its default widgets for this.

I think over time there will be a desire to include more and more page 
context in such things. But that aside, it becomes a lot less clear to 
me what kind of default widgets would be. For the simple case of 1 audio 
+ 1 video stream, things are reasonably straightforward, but once you 
start talking about local preview, multiple video streams, layout and 
audio panning for conferencing, negotiating new streams in the middle of 
a call, and all the other things which people have been considering in 
the use cases here, there's a fair bit of complexity that a "default" 
widget set has to handle, compared to, say, creating a default <video> 
tag to wrap an URL that points directly at a WebM file. Web site authors 
are likely to have a much easier time handling this complexity, as they 
understand the context of what they're providing and can make 
simplifying assumptions, and present things in a coherent way (tying 
identity to video windows, etc.), but a generic widget set doesn't have 
that luxury, unless we explicitly limit the scope of what it is required 
to support.

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Subject: [rtcweb] RTCWEB WG Virtual Interim Meeting: September 8, 2011
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The RTCWEB working group plans to hold an interim meeting meeting. It
will be held on September 8, 2011 at 7:00 A.M. PDT. We're currently
allocating 4 hours for the meeting. Details of participation will
follow on the working group mailing list 
(http://www.ietf.org/mail-archive/web/rtcweb/).

- RTCWEB Chairs

From yang.yanzi@zte.com.cn  Mon Aug 15 19:03:34 2011
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Subject: [rtcweb] Looking for the RTCWeb minutes of IETF #81 meeting
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Dear All,

Where could I find the minutes of RTCWeb WG in IETF #81 meeting? It's 
greatly appreciated if a link of the minute could be provided.
Thanks in advance.
BR,
Yanzi


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<br><font size=2 face="sans-serif">Dear All,</font>
<br>
<br><font size=2 face="sans-serif">Where could I find the minutes of RTCWeb
WG in IETF #81 meeting? It's greatly appreciated if a link of the minute
could be provided.</font>
<br><font size=2 face="sans-serif">Thanks in advance.</font>
<br><font size=2 face="sans-serif">BR,</font>
<br><font size=2 face="sans-serif">Yanzi</font>
<br>
<br>
--=_alternative 000B564A482578EE_=--


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Subject: Re: [rtcweb] Looking for the RTCWeb minutes of IETF #81 meeting
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It is not uploaded yet.

Once it is available, it should be here:
https://datatracker.ietf.org/meeting/81/materials.html

Kepeng
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of yang.yanzi@zte.com.cn
Sent: Tuesday, August 16, 2011 10:04 AM
To: rtcweb@ietf.org
Subject: [rtcweb] Looking for the RTCWeb minutes of IETF #81 meeting


Dear All,

Where could I find the minutes of RTCWeb WG in IETF #81 meeting? It's greatly appreciated if a link of the minute could be provided.
Thanks in advance.
BR,
Yanzi

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<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D">It is not uploaded yet.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D">Once it is available, it should be here:<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><a href="https://datatracker.ietf.org/meeting/81/materials.html">https://datatracker.ietf.org/meeting/81/materials.html</a><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal"><span lang="EN-US" style="font-size:10.5pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;
color:#1F497D">Kepeng<o:p></o:p></span></p>
<div style="border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt;font-family:
&quot;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span lang="EN-US" style="font-size:10.0pt;
font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org]
<b>On Behalf Of </b>yang.yanzi@zte.com.cn<br>
<b>Sent:</b> Tuesday, August 16, 2011 10:04 AM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> [rtcweb] Looking for the RTCWeb minutes of IETF #81 meeting<o:p></o:p></span></p>
</div>
<p class="MsoNormal"><span lang="EN-US"><o:p>&nbsp;</o:p></span></p>
<p class="MsoNormal" style="margin-bottom:12.0pt"><span lang="EN-US"><br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Dear All,</span><span lang="EN-US">
<br>
<br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Where could I find the minutes of RTCWeb WG in IETF #81 meeting? It's greatly appreciated if a link of the minute could be provided.</span><span lang="EN-US">
<br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Thanks in advance.</span><span lang="EN-US">
<br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">BR,</span><span lang="EN-US">
<br>
</span><span lang="EN-US" style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;">Yanzi</span><span lang="EN-US">
<o:p></o:p></span></p>
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From harald@alvestrand.no  Tue Aug 16 01:31:35 2011
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Subject: Re: [rtcweb] Dual stack RTCweb
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On 08/15/11 17:14, Olle E. Johansson wrote:
> 15 aug 2011 kl. 11:24 skrev Bernard Aboba:
>
>> ICE is designed to handle dual stack operation.  It can even deal with NAT64, if correctly implemented.
>>
>> Since ICE tests pairs before using them, there is no "happy eyeballs" problem with ICE (e.g. if an IPv6 route doesn't exist, the test will fail).
>>
>> That said, IPv6/IPv4 priorities may need some adjustment in some situations (e.g. if the IPv6 routes are more circuitous than IPv4, it may not make sense to prefer IPv6 over IPv4).
>>
>> That is one of the reasons that I am not enthusiastic about treating the SDP JSON blob as "opaque" (e.g. not subject to adjustment). Doing so without the ability
>> to adjust IPv6/IPv4 priorities will result in poor performance in dual stack operation in some cases.
>>
>>
> Well, if we use SRV records, the receiving part may use SRV to indicate preference.
I think the ICE procedure for connecting doesn't do any DNS lookups.
In which phase of the negotiation are you suggesting we look up SRV records?

Not having any opinion on the idea, just trying to figure out what we're 
talking about.
> Now, if I have an IPv6 only client ICE won't help unless we force turn usage for IPv6 only clients so that they offer dual stack always (like the current recommendation for SIP).
we can't force anything, but I do see IPv6-only clients as a powerful 
argument for offering dual-stack TURN servers.

I'm not 100% sure .... if client A is IPv6 only, and has a dual-stack 
TURN server, while client B is IPv4 only and does not have (or does not 
want to use) a TURN server .... will the connection complete in both 
directions?

It's easier to imagine that dual-stack TURN servers will be deployed 
usefully if only the service provider that wants to support the 
IPv6-only client will have to do anything.




From oej@edvina.net  Tue Aug 16 01:42:42 2011
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16 aug 2011 kl. 10:32 skrev Harald Alvestrand:

> On 08/15/11 17:14, Olle E. Johansson wrote:
>> 15 aug 2011 kl. 11:24 skrev Bernard Aboba:
>>=20
>>> ICE is designed to handle dual stack operation.  It can even deal =
with NAT64, if correctly implemented.
>>>=20
>>> Since ICE tests pairs before using them, there is no "happy =
eyeballs" problem with ICE (e.g. if an IPv6 route doesn't exist, the =
test will fail).
>>>=20
>>> That said, IPv6/IPv4 priorities may need some adjustment in some =
situations (e.g. if the IPv6 routes are more circuitous than IPv4, it =
may not make sense to prefer IPv6 over IPv4).
>>>=20
>>> That is one of the reasons that I am not enthusiastic about treating =
the SDP JSON blob as "opaque" (e.g. not subject to adjustment). Doing so =
without the ability
>>> to adjust IPv6/IPv4 priorities will result in poor performance in =
dual stack operation in some cases.
>>>=20
>>>=20
>> Well, if we use SRV records, the receiving part may use SRV to =
indicate preference.
> I think the ICE procedure for connecting doesn't do any DNS lookups.
> In which phase of the negotiation are you suggesting we look up SRV =
records?
Sorry, me confused. In SIP we have both a signalling and a media issue. =
SRV is of course a signalling issue.

>=20
> Not having any opinion on the idea, just trying to figure out what =
we're talking about.
>> Now, if I have an IPv6 only client ICE won't help unless we force =
turn usage for IPv6 only clients so that they offer dual stack always =
(like the current recommendation for SIP).
> we can't force anything, but I do see IPv6-only clients as a powerful =
argument for offering dual-stack TURN servers.
>=20
> I'm not 100% sure .... if client A is IPv6 only, and has a dual-stack =
TURN server, while client B is IPv4 only and does not have (or does not =
want to use) a TURN server .... will the connection complete in both =
directions?
This is one of the not covered cases in SIP. Most documents assume that =
everyone has IPv4 and very few assume single-stack Ipv6, something I =
think we need to start thinking more about.=20

>=20
> It's easier to imagine that dual-stack TURN servers will be deployed =
usefully if only the service provider that wants to support the =
IPv6-only client will have to do anything.

Right. But regardless, we need to consider these scenarios in the media =
negotiation, SDP or no SDP.

/O=

From harald@alvestrand.no  Tue Aug 16 01:50:51 2011
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Subject: Re: [rtcweb] URIs for rtcweb "calls"
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On 08/15/11 19:54, Ted Hardie wrote:
> On Sun, Aug 14, 2011 at 4:45 AM, Timothy B. Terriberry
> <tterriberry@mozilla.com>  wrote:
>> By "full game context" do you mean it would somehow load an http webpage
>> with HTML+CSS+JS to handle the signaling? If that's the case, what
>> advantages does this offer over normal http[s] URLs (with a path to the
>> necessary page and parameters, etc. needed to carry sufficient information
>> to establish the session). That certainly seems to already cover the case
>> of, "A URI that I could use to paste into a chat window." How does it handle
>> all the things that an http[s] URL already provides (port, path, caching,
>> proxies, all the associated services built around http (e.g. bit.ly), etc.)?
>>
> As I mentioned in my response to Matthew, I'm thinking abut a range of
> potential use cases. For the gaming site example where a full web
> context is created, I agree that an HTTPS URI could do the same thing.
>   You do get some minor advantages in using a distinct URI scheme,
> primarily in early identification that the resulting context will be a
> rtcweb context.  This might allow you apply permissions early, like a
> parental permission against use of rtcweb, or to allow the device to
> start setting up elements of its local context early.  It also allows
> you to standardize how you to identify the signaling context and
> target entity, which I doubt you could do in HTTPS URIs.  The
> arguments for and against scheme proliferation have been going for
> quite a while now, of course, so I understand that many people would
> prefer that there be only HTTPS URIs here.
>
> For the case where you are setting up something closer to a
> web-chat-with-an-agent-for-the-ad-seen here, I think the amount of
> page context will be very small, and that it will be closer to the
> experience of the browser/app setting up its default widgets for this.
For the first version of the RTCWEB spec, I would like to not mandate 
that the browser *have* a default widget set.

Most browsers have an FTP agent, and most browsers know how to call out 
to an external app as their mailto: handler, but if RTCWEB support is 
built into the browser and nowhere else on the platform, external apps 
are not an option, and built-in clients will make sure we don't see 
uniform UIs across platforms.

My first target is to make sure you can build applications that work the 
same in any supporting browser. I don't see a need to have an rtcweb URI 
for that.

WRT the permissions .... any time I see a short, human readable string 
and hear the word "permission" in the same context, I go "phishing" .... 
anything not digitally signed is an attack surface ... and anything that 
includes a digital signature is not human friendly ....

                     Harald




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Harald said:=20

> > I think the ICE procedure for connecting doesn't do any DNS lookups.

[BA] Right.  The SRV RRs (and subsequent A/AAAA RRs) are used in signaling=
=2C not media and in any case=2C ICE is end-to-end. =20

Olle said:

> >> Now=2C if I have an IPv6 only client ICE won't help unless we force tu=
rn usage for IPv6 only clients so that they offer dual stack always (like t=
he current recommendation for SIP).
> > we can't force anything=2C but I do see IPv6-only clients as a powerful=
 argument for offering dual-stack TURN servers.

[BA] Even an IPv6-only client could need ICE if there is a NAT64 on their n=
etwork.   And yes=2C relay candidates from a dual-stack TURN server could e=
nd up being
selected.   Overall=2C the idea behind ICE is to be robust against network =
configurations=2C to the greatest degree possible.  So it's best not to say=
 "we think IPv6 networks will generally be built in the following ways=2C s=
o the following features of ICE are not needed".  =20

> > I'm not 100% sure .... if client A is IPv6 only=2C and has a dual-stack=
 TURN server=2C while client B is IPv4 only and does not have (or does not =
want to use) a TURN server .... will the connection complete in both direct=
ions?

[BA] There first question is if A and B can communicate via SIP.   There co=
uld be a dual-stack SIP proxy somewhere=2C or maybe a NAT64 on the network=
=2C so that IPv6-A and IPv4-B could signal each other.=20

If there is a way for media to flow between IPv6-A and IPv4-B=2C then ICE s=
hould find it.  This could be via NAT64 (server reflexive candidate) or via=
 a dual stack TURN server (relay candidate). =20

> This is one of the not covered cases in SIP. Most documents assume that e=
veryone has IPv4 and very few assume single-stack Ipv6=2C something I think=
 we need to start thinking more about.=20

[BA] The SIP (and ICE) standards do cover this.  However=2C operationally w=
e are still in the early stages=2C and so implementations may function well=
 until they hit a deployment they didn't expect.  =20

> > It's easier to imagine that dual-stack TURN servers will be deployed us=
efully if only the service provider that wants to support the IPv6-only cli=
ent will have to do anything.

[BA] Some service providers may opt for NAT64 so an ICE implementation need=
s to handle this.=20
 		 	   		  =

--_49e4d2a8-da50-4852-8018-0f35d790c44f_
Content-Type: text/html; charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
--></style>
</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
Harald said: <br><br><div>&gt=3B &gt=3B I think the ICE procedure for conne=
cting doesn't do any DNS lookups.<br><br>[BA] Right.&nbsp=3B The SRV RRs (a=
nd subsequent A/AAAA RRs) are used in signaling=2C not media and in any cas=
e=2C ICE is end-to-end.&nbsp=3B <br><br>Olle said:<br><br>&gt=3B &gt=3B&gt=
=3B Now=2C if I have an IPv6 only client ICE won't help unless we force tur=
n usage for IPv6 only clients so that they offer dual stack always (like th=
e current recommendation for SIP).<br>&gt=3B &gt=3B we can't force anything=
=2C but I do see IPv6-only clients as a powerful argument for offering dual=
-stack TURN servers.<br><br>[BA] Even an IPv6-only client could need ICE if=
 there is a NAT64 on their network.&nbsp=3B&nbsp=3B And yes=2C relay candid=
ates from a dual-stack TURN server could end up being<br>selected.&nbsp=3B&=
nbsp=3B Overall=2C the idea behind ICE is to be robust against network conf=
igurations=2C to the greatest degree possible.&nbsp=3B So it's best not to =
say "we think IPv6 networks will generally be built in the following ways=
=2C so the following features of ICE are not needed".&nbsp=3B&nbsp=3B <br><=
br>&gt=3B &gt=3B I'm not 100% sure .... if client A is IPv6 only=2C and has=
 a dual-stack TURN server=2C while client B is IPv4 only and does not have =
(or does not want to use) a TURN server .... will the connection complete i=
n both directions?<br><br>[BA] There first question is if A and B can commu=
nicate via SIP.&nbsp=3B&nbsp=3B There could be a dual-stack SIP proxy somew=
here=2C or maybe a NAT64 on the network=2C so that IPv6-A and IPv4-B could =
signal each other. <br><br>If there is a way for media to flow between IPv6=
-A and IPv4-B=2C then ICE should find it.&nbsp=3B This could be via NAT64 (=
server reflexive candidate) or via a dual stack TURN server (relay candidat=
e).&nbsp=3B <br><br>&gt=3B This is one of the not covered cases in SIP. Mos=
t documents assume that everyone has IPv4 and very few assume single-stack =
Ipv6=2C something I think we need to start thinking more about. <br><br>[BA=
] The SIP (and ICE) standards do cover this.&nbsp=3B However=2C operational=
ly we are still in the early stages=2C and so implementations may function =
well until they hit a deployment they didn't expect. &nbsp=3B <br><br>&gt=
=3B &gt=3B It's easier to imagine that dual-stack TURN servers will be depl=
oyed usefully if only the service provider that wants to support the IPv6-o=
nly client will have to do anything.<br><br>[BA] Some service providers may=
 opt for NAT64 so an ICE implementation needs to handle this. <br></div> 		=
 	   		  </div></body>
</html>=

--_49e4d2a8-da50-4852-8018-0f35d790c44f_--

From oej@edvina.net  Tue Aug 16 05:01:40 2011
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16 aug 2011 kl. 13:23 skrev Bernard Aboba:

> Harald said:=20
>=20
> > > I think the ICE procedure for connecting doesn't do any DNS =
lookups.
>=20
> [BA] Right.  The SRV RRs (and subsequent A/AAAA RRs) are used in =
signaling, not media and in any case, ICE is end-to-end. =20
>=20
> Olle said:
>=20
> > >> Now, if I have an IPv6 only client ICE won't help unless we force =
turn usage for IPv6 only clients so that they offer dual stack always =
(like the current recommendation for SIP).
> > > we can't force anything, but I do see IPv6-only clients as a =
powerful argument for offering dual-stack TURN servers.
>=20
> [BA] Even an IPv6-only client could need ICE if there is a NAT64 on =
their network.   And yes, relay candidates from a dual-stack TURN server =
could end up being
> selected.   Overall, the idea behind ICE is to be robust against =
network configurations, to the greatest degree possible.  So it's best =
not to say "we think IPv6 networks will generally be built in the =
following ways, so the following features of ICE are not needed".  =20
>=20
> > > I'm not 100% sure .... if client A is IPv6 only, and has a =
dual-stack TURN server, while client B is IPv4 only and does not have =
(or does not want to use) a TURN server .... will the connection =
complete in both directions?
>=20
> [BA] There first question is if A and B can communicate via SIP.   =
There could be a dual-stack SIP proxy somewhere, or maybe a NAT64 on the =
network, so that IPv6-A and IPv4-B could signal each other.=20
>=20
> If there is a way for media to flow between IPv6-A and IPv4-B, then =
ICE should find it.  This could be via NAT64 (server reflexive =
candidate) or via a dual stack TURN server (relay candidate). =20
>=20
> > This is one of the not covered cases in SIP. Most documents assume =
that everyone has IPv4 and very few assume single-stack Ipv6, something =
I think we need to start thinking more about.=20
>=20
> [BA] The SIP (and ICE) standards do cover this.  However, =
operationally we are still in the early stages, and so implementations =
may function well until they hit a deployment they didn't expect.  =20
I would not say that the SIP standards do cover dual stack and single =
stack use cases fully.  But that's an off-topic discussion here, I =
guess. We just need to make sure that we keep these issues in this =
discussion for media negotiation and possible signalling specs.

>=20
> > > It's easier to imagine that dual-stack TURN servers will be =
deployed usefully if only the service provider that wants to support the =
IPv6-only client will have to do anything.
>=20
> [BA] Some service providers may opt for NAT64 so an ICE implementation =
needs to handle this.=20

Well, when talking about apps in web browsers, we are talking about a =
large number of situations compared with where we have  "pure" sip =
deployments today (which is mostly managed networks).

/O=

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On 08/15/2011 06:35 PM, IESG Secretary wrote:
> The RTCWEB working group plans to hold an interim meeting meeting. It
> will be held on September 8, 2011 at 7:00 A.M. PDT. We're currently
> allocating 4 hours for the meeting. Details of participation will
> follow on the working group mailing list
> (http://www.ietf.org/mail-archive/web/rtcweb/).

If it would be useful, I believe I can provide a wideband-capable 
conference bridge  for this interim meeting. It would be SIP accessible 
with various wideband codecs, PSTN accessible, and Skype accessible 
(narrowband only).

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kfleming@digium.com | SIP: kpfleming@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Thread-Topic: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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From: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
To: "Harald Alvestrand" <harald@alvestrand.no>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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|I think it should be up to the offer generator whether
|he adds "a=3Drtcp-mux" to the first section or not.
=20
Since the goal is to minimize the number of transport connections I
think we should put something stronger to encourage the offerer and
answerer to support RTCP multiplexing. At the minimum, a normative
statement recommending the use of RTCP multiplexing with this grouping
semantics would go a long way (symmetric RTP for e.g is a
recommendation, but virtually all implementations support it).
=20
|Do we need to state that if the first section has it, all=20
|the other sections have to have it, or can we live with a=20
|mixture?
=20
If the first section has it, it can be applied to all other section
within that group, so it shouldn't matter whether the reset of the
sections have it or not. A mixture doesn't seem would serve any purpose.
=20
Muthu
=20
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Friday, August 12, 2011 8:29 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
On 08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:=20
|In general, I think we should keep extensions as orthogonal
|as possible, so the normative language here should just say
|"establish RTCP sessions normally for the first section, and=20
|don't do it for other sections".
=20
Looks good to me. Should it also say "Within a TOGETHER group it is
RECOMMENDED to multiplex RTCP on the same RTP port from the first
section" or something stronger?
I think it should be up to the offer generator whether he adds
"a=3Drtcp-mux" to the first section or not.
Do we need to state that if the first section has it, all the other
sections have to have it, or can we live with a mixture?
 =20
Muthu
=20
From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: Friday, August 12, 2011 6:49 PM
To: Muthu Arul Mozhi Perumal (mperumal)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:=20
A few comments:
=20
1) Section 4 has this:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
session, and the parameters for the other sections MUST be ignored.
=20
I think it needs to be:
If the responder understands the semantics of the TOGETHER extension,
the parameters of the first section MUST be used to establish the RTP
and RCP session, and the parameters for the other sections MUST be=20
ignored.
Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port
multiplexing (RFC 5761), so I did not think of it.
In general, I think we should keep extensions as orthogonal as possible,
so the normative language here should just say "establish RTCP sessions
normally for the first section, and don't do it for other sections".


=20
2) When the RTCP port is not the next higher (odd) port number following
the RTP port described in the m-line, the "a=3Drtcp" attribute (defined =
in
RFC-3605) is used to signal it, as in:
m=3Daudio 49170 RTP/AVP 0
a=3Drtcp:53020 IN IP4 126.16.64.4
=20
With the grouping semantics described in the draft, it seems this RTCP
port should be taken only from the first section. So, this could be
could be added to section 4.
=20
3) RTCP could be multiplexed with RTP on a single port and signaled
using the "a=3Drtcp-mux" attribute (defined in RFC 5761). When it is =
used
together with the grouping semantics described in the draft, it seems
all RTCP sessions of that group need to be multiplexed on the RTP port
from the first section.
There should be only one RTCP session established for all the TOGETHER
m=3D lines, yes.


=20
Muthu
=20
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Harald Alvestrand
Sent: Thursday, August 11, 2011 5:02 PM
To: rtcweb@ietf.org
Subject: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
=20
I have taken some of the information I learned from the discussions in
Quebec City about the issues of putting video and audio into the same
RTP session and created a draft from them outlining a solution to the
problem of signalling this configuration using SDP.

Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or
MMUSIC.

                      Harald

-------- Original Message --------=20
Subject:=20
I-D Action: draft-alvestrand-one-rtp-00.txt
Date:=20
Thu, 11 Aug 2011 04:28:09 -0700
From:=20
internet-drafts@ietf.org
Reply-To:=20
internet-drafts@ietf.org
To:=20
i-d-announce@ietf.org
=20
A New Internet-Draft is available from the on-line Internet-Drafts
directories.
=20
       Title           : SDP Grouping for Single RTP Sessions
       Author(s)       : Harald Tveit Alvestrand
       Filename        : draft-alvestrand-one-rtp-00.txt
       Pages           : 8
       Date            : 2011-08-11
=20
   This document describes an extension to the Session Description
   Protocol (SDP) to describe RTP sessions where media of multiple top
   level types, for example audio and video, are carried in the same RTP
   session.
=20
   This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
   consideration.
=20
=20
=20
A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
=20
Internet-Drafts are also available by anonymous FTP at:
ftp://ftp.ietf.org/internet-drafts/
=20
This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt
_______________________________________________
I-D-Announce mailing list
I-D-Announce@ietf.org
https://www.ietf.org/mailman/listinfo/i-d-announce
Internet-Draft directories: http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt
=20
=20
=20

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</head>

<body bgcolor=3Dwhite lang=3DEN-US link=3Dblue vlink=3Dpurple =
style=3D'tab-interval:.5in'>

<div class=3DWordSection1>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|I
think it should be up to the offer generator =
whether<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|he
adds &quot;a=3D<span class=3DSpellE>rtcp-mux</span>&quot; to the first =
section or
not.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>Since
the goal is to minimize the number of transport connections I think we =
should
put something stronger to encourage the <span =
class=3DSpellE>offerer</span> and answerer
to support <span class=3DSpellE>RTCP</span> multiplexing. At the =
minimum, a
normative statement recommending the use of <span =
class=3DSpellE>RTCP</span>
multiplexing with this grouping semantics would go a long way (symmetric =
<span
class=3DSpellE>RTP</span> for <span class=3DSpellE>e.g</span> is a =
recommendation,
but virtually all implementations support it).<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|Do
we need to state that if the first section has it, all =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|the
other sections have to have it, or can we live with a =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>|mixture?<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>If
the first section has it, it can be applied to all other section within =
that
group, so it shouldn't matter whether the reset of the sections have it =
or not.
A mixture doesn't seem would serve any purpose.<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><o:p>&nbsp;</o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'>Muthu<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;mso-bidi-font-size:11.0pt;
font-family:"Courier New";mso-bidi-font-family:"Times New =
Roman";color:windowtext'><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=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
mso-fareast-font-family:"Times New =
Roman";color:windowtext'>From:</span></b><span
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";mso-fareast-f=
ont-family:
"Times New Roman";color:windowtext'> Harald Alvestrand
[mailto:harald@alvestrand.no] <br>
<b>Sent:</b> Friday, August 12, 2011 8:29 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>On
08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote: =
<o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>|In general, I think we should keep extensions as =
orthogonal</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>|as possible, so the normative language here should =
just say</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>|&quot;establish RTCP sessions normally for the first
section, and </span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>|don't do it for other =
sections&quot;.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>Looks good to me. Should it also say &quot;Within a =
TOGETHER
group it is RECOMMENDED to multiplex RTCP on the same RTP port from the =
first
section&quot; or something stronger?</span><o:p></o:p></p>

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'>I
think it should be up to the offer generator whether he adds
&quot;a=3Drtcp-mux&quot; to the first section or not.<br>
Do we need to state that if the first section has it, all the other =
sections
have to have it, or can we live with a mixture?<br>
</span><span style=3D'font-size:10.0pt;font-family:"Courier =
New";mso-fareast-font-family:
"Times New Roman";color:windowtext'>&nbsp;</span><span =
style=3D'mso-fareast-font-family:
"Times New Roman"'> <o:p></o:p></span></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>Muthu</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<div style=3D'border:none;border-left:solid windowtext 1.5pt;padding:0in =
0in 0in 4.0pt;
border-color:-moz-use-text-color -moz-use-text-color =
-moz-use-text-color&#13;&#10;          blue'>

<div>

<div style=3D'border:none;border-top:solid windowtext =
1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color&#13;&#10;              =
-moz-use-text-color'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> Harald Alvestrand [<a
href=3D"mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] =
<br>
<b>Sent:</b> Friday, August 12, 2011 6:49 PM<br>
<b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
<b>Cc:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> Re: [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>

</div>

</div>

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

<p class=3DMsoNormal>On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal =
(mperumal)
wrote: <o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>A few comments:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>1) Section 4 has this:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>If the responder understands the semantics of the =
TOGETHER
extension,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>the parameters of the first section MUST be used to =
establish
the RTP</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>session, and the parameters for the other sections =
MUST be
ignored.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>I think it needs to be:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>If the responder understands the semantics of the =
TOGETHER
extension,</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>the parameters of the first section MUST be used to =
establish
the RTP</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>and RCP session, and the parameters for the other =
sections
MUST be </span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>ignored.</span><o:p></o:p></p>

<p class=3DMsoNormal>Good point. Will fix. Normally, I think RTCWEB apps =
will use
RTCP port multiplexing (RFC 5761), so I did not think of it.<br>
In general, I think we should keep extensions as orthogonal as possible, =
so the
normative language here should just say &quot;establish RTCP sessions =
normally
for the first section, and don't do it for other sections&quot;.<br
style=3D'mso-special-character:line-break'>
<![if !supportLineBreakNewLine]><br =
style=3D'mso-special-character:line-break'>
<![endif]><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>2) When the RTCP port is not the next higher (odd) =
port
number following the RTP port described in the m-line, the =
&quot;a=3Drtcp&quot;
attribute (defined in RFC-3605) is used to signal it, as =
in:</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>m=3Daudio 49170 RTP/AVP 0</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>a=3Drtcp:53020 IN IP4 =
126.16.64.4</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>With the grouping semantics described in the draft, it =
seems
this RTCP port should be taken only from the first section. So, this =
could be
could be added to section 4.</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>3) RTCP could be multiplexed with RTP on a single port =
and
signaled using the &quot;a=3Drtcp-mux&quot; attribute (defined in RFC =
5761). When
it is used together with the grouping semantics described in the draft, =
it
seems all RTCP sessions of that group need to be multiplexed on the RTP =
port
from the first section.</span><o:p></o:p></p>

<p class=3DMsoNormal>There should be only one RTCP session established =
for all
the TOGETHER m=3D lines, yes.<br =
style=3D'mso-special-character:line-break'>
<![if !supportLineBreakNewLine]><br =
style=3D'mso-special-character:line-break'>
<![endif]><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>Muthu</span><o:p></o:p></p>

<p class=3DMsoNormal><span =
style=3D'font-size:10.0pt;font-family:"Courier New";
color:windowtext'>&nbsp;</span><o:p></o:p></p>

<div style=3D'border:none;border-left:solid windowtext 1.5pt;padding:0in =
0in 0in 4.0pt;
border-color:-moz-use-text-color -moz-use-text-color&#13;&#10;           =
 -moz-use-text-color blue'>

<div>

<div style=3D'border:none;border-top:solid windowtext =
1.0pt;padding:3.0pt 0in 0in 0in;
border-color:-moz-use-text-color'>

<p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> <a
href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a> [<a
href=3D"mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a=
>] <b>On
Behalf Of </b>Harald Alvestrand<br>
<b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
<b>To:</b> <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<b>Subject:</b> [rtcweb] Fwd: I-D Action: =
draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>

</div>

</div>

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

<p class=3DMsoNormal>I have taken some of the information I learned from =
the
discussions in Quebec City about the issues of putting video and audio =
into the
same RTP session and created a draft from them outlining a solution to =
the
problem of signalling this configuration using SDP.<br>
<br>
Comments welcome, including the recommended context for processing the
document; in a few days I'll send the same note to AVTCORE and/or =
MMUSIC.<br>
<br>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
Harald<br>
<br>
-------- Original Message -------- <o:p></o:p></p>

<table class=3DMsoNormalTable border=3D0 cellspacing=3D0 cellpadding=3D0
 style=3D'mso-cellspacing:0in;mso-yfti-tbllook:1184;mso-padding-alt:0in =
5.4pt 0in 5.4pt'>
 <tr style=3D'mso-yfti-irow:0;mso-yfti-firstrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright =
style=3D'text-align:right'><b>Subject: </b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>I-D Action: =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:1'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>Date: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal>Thu, 11 Aug 2011 04:28:09 -0700<o:p></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:2'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>From: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:3'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright =
style=3D'text-align:right'><b>Reply-To: </b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p=
></o:p></p>
  </td>
 </tr>
 <tr style=3D'mso-yfti-irow:4;mso-yfti-lastrow:yes'>
  <td nowrap valign=3Dtop style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal align=3Dright style=3D'text-align:right'><b>To: =
</b><o:p></o:p></p>
  </td>
  <td style=3D'padding:0in 0in 0in 0in'>
  <p class=3DMsoNormal><a =
href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p=
></p>
  </td>
 </tr>
</table>

<p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>&nbsp;<o:p></o:p></p>

<pre>A New Internet-Draft is available from the on-line Internet-Drafts =
directories.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp; =
Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : SDP =
Grouping for Single RTP =
Sessions<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Harald Tveit =
Alvestrand<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp; =
Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
8<o:p></o:p></pre><pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : =
2011-08-11<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp; =
This document describes an extension to the Session =
Description<o:p></o:p></pre><pre>&nbsp;&nbsp; Protocol (SDP) to describe =
RTP sessions where media of multiple =
top<o:p></o:p></pre><pre>&nbsp;&nbsp; level types, for example audio and =
video, are carried in the same RTP<o:p></o:p></pre><pre>&nbsp;&nbsp; =
session.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;&nbsp; =
This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs =
for<o:p></o:p></pre><pre>&nbsp;&nbsp; =
consideration.<o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>&nbsp;<o:=
p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>A URL for this =
Internet-Draft is:<o:p></o:p></pre><pre><a
href=3D"http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.t=
xt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</=
a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>Internet-Drafts are =
also available by anonymous FTP at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-=
drafts/</a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre><pre>This =
Internet-Draft can be retrieved at:<o:p></o:p></pre><pre><a
href=3D"ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.tx=
t">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a>=
<o:p></o:p></pre><pre>_______________________________________________<o:p=
></o:p></pre><pre>I-D-Announce mailing list<o:p></o:p></pre><pre><a
href=3D"mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p=
></pre><pre><a
href=3D"https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.i=
etf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre><pre>Internet-D=
raft directories: <a
href=3D"http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html<=
/a><o:p></o:p></pre><pre>or <a
href=3D"ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/iet=
f/1shadow-sites.txt</a><o:p></o:p></pre><pre>&nbsp;<o:p></o:p></pre></div=
>

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

</div>

<p class=3DMsoNormal><span style=3D'mso-fareast-font-family:"Times New =
Roman"'><o:p>&nbsp;</o:p></span></p>

</div>

</div>

</body>

</html>

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To: "Muthu Arul Mozhi Perumal (mperumal)" <mperumal@cisco.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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On 08/17/11 11:27, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> |I think it should be up to the offer generator whether
>
> |he adds "a=rtcp-mux" to the first section or not.
>
> Since the goal is to minimize the number of transport connections I 
> think we should put something stronger to encourage the offerer and 
> answerer to support RTCP multiplexing. At the minimum, a normative 
> statement recommending the use of RTCP multiplexing with this grouping 
> semantics would go a long way (symmetric RTP for e.g is a 
> recommendation, but virtually all implementations support it).
>
I agree that this is critical in the RTCWEB context. It may not be 
critical for all contexts.
I think that should be a matter the (not yet existing) draft referenced 
here should say:

5.  Data framing and securing

    SRTP [RFC3550] is used for transport of all real-time media.

    The detailed considerations for usage of functions from RTP and SRTP,
    as well as for non-media real-time data, are given in <WORKING GROUP
    DRAFT "MEDIA TRANSPORTS">.

(this is from the -overview document)
>
> |Do we need to state that if the first section has it, all
>
> |the other sections have to have it, or can we live with a
>
> |mixture?
>
> If the first section has it, it can be applied to all other section 
> within that group, so it shouldn't matter whether the reset of the 
> sections have it or not. A mixture doesn't seem would serve any purpose.
>
When TOGETHER is applied, the other sections' values doesn't matter at all.
When TOGETHER does NOT apply (fallback), it may matter.
I don't see any purpose with a mixture either. But we'll see what 
further review brings.
>
> Muthu
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Friday, August 12, 2011 8:29 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> On 08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> |In general, I think we should keep extensions as orthogonal
>
> |as possible, so the normative language here should just say
>
> |"establish RTCP sessions normally for the first section, and
>
> |don't do it for other sections".
>
> Looks good to me. Should it also say "Within a TOGETHER group it is 
> RECOMMENDED to multiplex RTCP on the same RTP port from the first 
> section" or something stronger?
>
> I think it should be up to the offer generator whether he adds 
> "a=rtcp-mux" to the first section or not.
> Do we need to state that if the first section has it, all the other 
> sections have to have it, or can we live with a mixture?
>
> Muthu
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Friday, August 12, 2011 6:49 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> A few comments:
>
> 1) Section 4 has this:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> session, and the parameters for the other sections MUST be ignored.
>
> I think it needs to be:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> and RCP session, and the parameters for the other sections MUST be
>
> ignored.
>
> Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port 
> multiplexing (RFC 5761), so I did not think of it.
> In general, I think we should keep extensions as orthogonal as 
> possible, so the normative language here should just say "establish 
> RTCP sessions normally for the first section, and don't do it for 
> other sections".
>
> 2) When the RTCP port is not the next higher (odd) port number 
> following the RTP port described in the m-line, the "a=rtcp" attribute 
> (defined in RFC-3605) is used to signal it, as in:
>
> m=audio 49170 RTP/AVP 0
>
> a=rtcp:53020 IN IP4 126.16.64.4
>
> With the grouping semantics described in the draft, it seems this RTCP 
> port should be taken only from the first section. So, this could be 
> could be added to section 4.
>
> 3) RTCP could be multiplexed with RTP on a single port and signaled 
> using the "a=rtcp-mux" attribute (defined in RFC 5761). When it is 
> used together with the grouping semantics described in the draft, it 
> seems all RTCP sessions of that group need to be multiplexed on the 
> RTP port from the first section.
>
> There should be only one RTCP session established for all the TOGETHER 
> m= lines, yes.
>
> Muthu
>
> *From:*rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org> 
> [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Harald Alvestrand
> *Sent:* Thursday, August 11, 2011 5:02 PM
> *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> I have taken some of the information I learned from the discussions in 
> Quebec City about the issues of putting video and audio into the same 
> RTP session and created a draft from them outlining a solution to the 
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the 
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.
>
>                       Harald
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> I-D Action: draft-alvestrand-one-rtp-00.txt
>
> *Date: *
>
> 	
>
> Thu, 11 Aug 2011 04:28:09 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *Reply-To: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   
>         Title           : SDP Grouping for Single RTP Sessions
>         Author(s)       : Harald Tveit Alvestrand
>         Filename        : draft-alvestrand-one-rtp-00.txt
>         Pages           : 8
>         Date            : 2011-08-11
>   
>     This document describes an extension to the Session Description
>     Protocol (SDP) to describe RTP sessions where media of multiple top
>     level types, for example audio and video, are carried in the same RTP
>     session.
>   
>     This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
>     consideration.
>   
>   
>   
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org  <mailto:I-D-Announce@ietf.org>
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:http://www.ietf.org/shadow.html
> orftp://ftp.ietf.org/ietf/1shadow-sites.txt
>   
>


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    On 08/17/11 11:27, Muthu Arul Mozhi Perumal (mperumal) wrote:
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|I
            think it should be up to the offer generator whether<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|he
            adds "a=<span class="SpellE">rtcp-mux</span>" to the first
            section or
            not.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Since
            the goal is to minimize the number of transport connections
            I think we should
            put something stronger to encourage the <span
              class="SpellE">offerer</span> and answerer
            to support <span class="SpellE">RTCP</span> multiplexing.
            At the minimum, a
            normative statement recommending the use of <span
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              class="SpellE">e.g</span> is a recommendation,
            but virtually all implementations support it).</span></p>
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    </blockquote>
    I agree that this is critical in the RTCWEB context. It may not be
    critical for all contexts.<br>
    I think that should be a matter the (not yet existing) draft
    referenced here should say:<br>
    <br>
    5.&nbsp; Data framing and securing<br>
    <br>
    &nbsp;&nbsp; SRTP [RFC3550] is used for transport of all real-time media.<br>
    <br>
    &nbsp;&nbsp; The detailed considerations for usage of functions from RTP and
    SRTP,<br>
    &nbsp;&nbsp; as well as for non-media real-time data, are given in &lt;WORKING
    GROUP<br>
    &nbsp;&nbsp; DRAFT "MEDIA TRANSPORTS"&gt;.<br>
    <br>
    (this is from the -overview document)<br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625FDC5@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|Do
            we need to state that if the first section has it, all <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|the
            other sections have to have it, or can we live with a <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|mixture?<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the first section has it, it can be applied to all other
            section within that
            group, so it shouldn't matter whether the reset of the
            sections have it or not.
            A mixture doesn't seem would serve any purpose.</span></p>
      </div>
    </blockquote>
    When TOGETHER is applied, the other sections' values doesn't matter
    at all.<br>
    When TOGETHER does NOT apply (fallback), it may matter.<br>
    I don't see any purpose with a mixture either. But we'll see what
    further review brings. <br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625FDC5@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
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        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
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        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Muthu<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <div style="border-width: medium medium medium 1.5pt;
          border-style: none none none solid; border-color:
          -moz-use-text-color -moz-use-text-color -moz-use-text-color
          blue; padding: 0in 0in 0in 4pt;">
          <div>
            <div style="border-right: medium none; border-width: 1pt
              medium medium; border-style: solid none none;
              border-color: rgb(181, 196, 223) -moz-use-text-color
              -moz-use-text-color; padding: 3pt 0in 0in;">
              <p class="MsoNormal"><b><span style="font-size: 10pt;
                    font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> Harald Alvestrand
                  [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
                  <b>Sent:</b> Friday, August 12, 2011 8:29 PM<br>
                  <b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
                  <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> Re: [rtcweb] Fwd: I-D Action:
                  draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><span style="">On
              08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
              <o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|In
              general, I think we should keep extensions as orthogonal</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|as
              possible, so the normative language here should just say</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|"establish
              RTCP sessions normally for the first
              section, and </span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|don't
              do it for other sections".</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">Looks
              good to me. Should it also say "Within a TOGETHER
              group it is RECOMMENDED to multiplex RTCP on the same RTP
              port from the first
              section" or something stronger?</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="">I
              think it should be up to the offer generator whether he
              adds
              "a=rtcp-mux" to the first section or not.<br>
              Do we need to state that if the first section has it, all
              the other sections
              have to have it, or can we live with a mixture?<br>
            </span><span style="font-size: 10pt; font-family:
              &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><span
              style=""> <o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">Muthu</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <div style="border-width: medium medium medium 1.5pt;
            border-style: none none none solid; padding: 0in 0in 0in
            4pt; border-color: -moz-use-text-color -moz-use-text-color
            -moz-use-text-color blue;">
            <div>
              <div style="border-right: medium none; border-width: 1pt
                medium medium; border-style: solid none none; padding:
                3pt 0in 0in; border-color: -moz-use-text-color;">
                <p class="MsoNormal"><b><span style="font-size: 10pt;
                      font-family:
                      &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                      windowtext;">From:</span></b><span
                    style="font-size: 10pt; font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;"> Harald Alvestrand [<a
                      moz-do-not-send="true"
                      href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>]
                    <br>
                    <b>Sent:</b> Friday, August 12, 2011 6:49 PM<br>
                    <b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
                    <b>Cc:</b> <a moz-do-not-send="true"
                      href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    <b>Subject:</b> Re: [rtcweb] Fwd: I-D Action:
                    draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>
              </div>
            </div>
            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
            <p class="MsoNormal">On 08/12/2011 03:09 PM, Muthu Arul
              Mozhi Perumal (mperumal)
              wrote: <o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">A few comments:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">1) Section 4 has this:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">If the responder understands the semantics
                of the TOGETHER
                extension,</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">the parameters of the first section MUST be
                used to establish
                the RTP</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">session, and the parameters for the other
                sections MUST be
                ignored.</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">I think it needs to be:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">If the responder understands the semantics
                of the TOGETHER
                extension,</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">the parameters of the first section MUST be
                used to establish
                the RTP</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">and RCP session, and the parameters for the
                other sections
                MUST be </span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">ignored.</span><o:p></o:p></p>
            <p class="MsoNormal">Good point. Will fix. Normally, I think
              RTCWEB apps will use
              RTCP port multiplexing (RFC 5761), so I did not think of
              it.<br>
              In general, I think we should keep extensions as
              orthogonal as possible, so the
              normative language here should just say "establish RTCP
              sessions normally
              for the first section, and don't do it for other
              sections".<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">2) When the RTCP port is not the next
                higher (odd) port
                number following the RTP port described in the m-line,
                the "a=rtcp"
                attribute (defined in RFC-3605) is used to signal it, as
                in:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">m=audio 49170 RTP/AVP 0</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">a=rtcp:53020 IN IP4 126.16.64.4</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">With the grouping semantics described in
                the draft, it seems
                this RTCP port should be taken only from the first
                section. So, this could be
                could be added to section 4.</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">3) RTCP could be multiplexed with RTP on a
                single port and
                signaled using the "a=rtcp-mux" attribute (defined in
                RFC 5761). When
                it is used together with the grouping semantics
                described in the draft, it
                seems all RTCP sessions of that group need to be
                multiplexed on the RTP port
                from the first section.</span><o:p></o:p></p>
            <p class="MsoNormal">There should be only one RTCP session
              established for all
              the TOGETHER m= lines, yes.<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">Muthu</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <div style="border-width: medium medium medium 1.5pt;
              border-style: none none none solid; padding: 0in 0in 0in
              4pt; border-color: -moz-use-text-color -moz-use-text-color
              -moz-use-text-color blue;">
              <div>
                <div style="border-right: medium none; border-width: 1pt
                  medium medium; border-style: solid none none; padding:
                  3pt 0in 0in; border-color: -moz-use-text-color;">
                  <p class="MsoNormal"><b><span style="font-size: 10pt;
                        font-family:
                        &quot;Tahoma&quot;,&quot;sans-serif&quot;;
                        color: windowtext;">From:</span></b><span
                      style="font-size: 10pt; font-family:
                      &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                      windowtext;"> <a moz-do-not-send="true"
                        href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                      [<a moz-do-not-send="true"
                        href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                      <b>On
                        Behalf Of </b>Harald Alvestrand<br>
                      <b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
                      <b>To:</b> <a moz-do-not-send="true"
                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                      <b>Subject:</b> [rtcweb] Fwd: I-D Action:
                      draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>
                </div>
              </div>
              <p class="MsoNormal">&nbsp;<o:p></o:p></p>
              <p class="MsoNormal">I have taken some of the information
                I learned from the
                discussions in Quebec City about the issues of putting
                video and audio into the
                same RTP session and created a draft from them outlining
                a solution to the
                problem of signalling this configuration using SDP.<br>
                <br>
                Comments welcome, including the recommended context for
                processing the
                document; in a few days I'll send the same note to
                AVTCORE and/or MMUSIC.<br>
                <br>
                &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                Harald<br>
                <br>
                -------- Original Message -------- <o:p></o:p></p>
              <table class="MsoNormalTable" style="" border="0"
                cellpadding="0" cellspacing="0">
                <tbody>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Subject: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal">I-D Action:
                        draft-alvestrand-one-rtp-00.txt<o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Date: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal">Thu, 11 Aug 2011 04:28:09
                        -0700<o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>From: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Reply-To: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>To: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                </tbody>
              </table>
              <p class="MsoNormal" style="margin-bottom: 12pt;">&nbsp;<o:p></o:p></p>
              <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : SDP Grouping for Single RTP Sessions<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Harald Tveit Alvestrand<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 8<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2011-08-11<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; This document describes an extension to the Session Description<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; Protocol (SDP) to describe RTP sessions where media of multiple top<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; level types, for example audio and video, are carried in the same RTP<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; session.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; consideration.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>A URL for this Internet-Draft is:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>This Internet-Draft can be retrieved at:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
              <pre>_______________________________________________<o:p></o:p></pre>
              <pre>I-D-Announce mailing list<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre>
              <pre>Internet-Draft directories: <a moz-do-not-send="true" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a><o:p></o:p></pre>
              <pre>or <a moz-do-not-send="true" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
            </div>
            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
          </div>
          <p class="MsoNormal"><span style=""><o:p>&nbsp;</o:p></span></p>
        </div>
      </div>
    </blockquote>
    <br>
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Date: Wed, 17 Aug 2011 11:42:27 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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On 08/17/11 11:27, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> |I think it should be up to the offer generator whether
>
> |he adds "a=rtcp-mux" to the first section or not.
>
> Since the goal is to minimize the number of transport connections I 
> think we should put something stronger to encourage the offerer and 
> answerer to support RTCP multiplexing. At the minimum, a normative 
> statement recommending the use of RTCP multiplexing with this grouping 
> semantics would go a long way (symmetric RTP for e.g is a 
> recommendation, but virtually all implementations support it).
>
I agree that this is critical in the RTCWEB context. It may not be 
critical for all contexts.
I think that should be a matter the (not yet existing) draft referenced 
here should say:

5.  Data framing and securing

    SRTP [RFC3550] is used for transport of all real-time media.

    The detailed considerations for usage of functions from RTP and SRTP,
    as well as for non-media real-time data, are given in <WORKING GROUP
    DRAFT "MEDIA TRANSPORTS">.

(this is from the -overview document)
>
> |Do we need to state that if the first section has it, all
>
> |the other sections have to have it, or can we live with a
>
> |mixture?
>
> If the first section has it, it can be applied to all other section 
> within that group, so it shouldn't matter whether the reset of the 
> sections have it or not. A mixture doesn't seem would serve any purpose.
>
When TOGETHER is applied, the other sections' values doesn't matter at all.
When TOGETHER does NOT apply (fallback), it may matter.
I don't see any purpose with a mixture either. But we'll see what 
further review brings.
>
> Muthu
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Friday, August 12, 2011 8:29 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> On 08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> |In general, I think we should keep extensions as orthogonal
>
> |as possible, so the normative language here should just say
>
> |"establish RTCP sessions normally for the first section, and
>
> |don't do it for other sections".
>
> Looks good to me. Should it also say "Within a TOGETHER group it is 
> RECOMMENDED to multiplex RTCP on the same RTP port from the first 
> section" or something stronger?
>
> I think it should be up to the offer generator whether he adds 
> "a=rtcp-mux" to the first section or not.
> Do we need to state that if the first section has it, all the other 
> sections have to have it, or can we live with a mixture?
>
> Muthu
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* Friday, August 12, 2011 6:49 PM
> *To:* Muthu Arul Mozhi Perumal (mperumal)
> *Cc:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> On 08/12/2011 03:09 PM, Muthu Arul Mozhi Perumal (mperumal) wrote:
>
> A few comments:
>
> 1) Section 4 has this:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> session, and the parameters for the other sections MUST be ignored.
>
> I think it needs to be:
>
> If the responder understands the semantics of the TOGETHER extension,
>
> the parameters of the first section MUST be used to establish the RTP
>
> and RCP session, and the parameters for the other sections MUST be
>
> ignored.
>
> Good point. Will fix. Normally, I think RTCWEB apps will use RTCP port 
> multiplexing (RFC 5761), so I did not think of it.
> In general, I think we should keep extensions as orthogonal as 
> possible, so the normative language here should just say "establish 
> RTCP sessions normally for the first section, and don't do it for 
> other sections".
>
> 2) When the RTCP port is not the next higher (odd) port number 
> following the RTP port described in the m-line, the "a=rtcp" attribute 
> (defined in RFC-3605) is used to signal it, as in:
>
> m=audio 49170 RTP/AVP 0
>
> a=rtcp:53020 IN IP4 126.16.64.4
>
> With the grouping semantics described in the draft, it seems this RTCP 
> port should be taken only from the first section. So, this could be 
> could be added to section 4.
>
> 3) RTCP could be multiplexed with RTP on a single port and signaled 
> using the "a=rtcp-mux" attribute (defined in RFC 5761). When it is 
> used together with the grouping semantics described in the draft, it 
> seems all RTCP sessions of that group need to be multiplexed on the 
> RTP port from the first section.
>
> There should be only one RTCP session established for all the TOGETHER 
> m= lines, yes.
>
> Muthu
>
> *From:*rtcweb-bounces@ietf.org <mailto:rtcweb-bounces@ietf.org> 
> [mailto:rtcweb-bounces@ietf.org] *On Behalf Of *Harald Alvestrand
> *Sent:* Thursday, August 11, 2011 5:02 PM
> *To:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
>
> I have taken some of the information I learned from the discussions in 
> Quebec City about the issues of putting video and audio into the same 
> RTP session and created a draft from them outlining a solution to the 
> problem of signalling this configuration using SDP.
>
> Comments welcome, including the recommended context for processing the 
> document; in a few days I'll send the same note to AVTCORE and/or MMUSIC.
>
>                       Harald
>
> -------- Original Message --------
>
> *Subject: *
>
> 	
>
> I-D Action: draft-alvestrand-one-rtp-00.txt
>
> *Date: *
>
> 	
>
> Thu, 11 Aug 2011 04:28:09 -0700
>
> *From: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *Reply-To: *
>
> 	
>
> internet-drafts@ietf.org <mailto:internet-drafts@ietf.org>
>
> *To: *
>
> 	
>
> i-d-announce@ietf.org <mailto:i-d-announce@ietf.org>
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   
>         Title           : SDP Grouping for Single RTP Sessions
>         Author(s)       : Harald Tveit Alvestrand
>         Filename        : draft-alvestrand-one-rtp-00.txt
>         Pages           : 8
>         Date            : 2011-08-11
>   
>     This document describes an extension to the Session Description
>     Protocol (SDP) to describe RTP sessions where media of multiple top
>     level types, for example audio and video, are carried in the same RTP
>     session.
>   
>     This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
>     consideration.
>   
>   
>   
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.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-alvestrand-one-rtp-00.txt
> _______________________________________________
> I-D-Announce mailing list
> I-D-Announce@ietf.org  <mailto:I-D-Announce@ietf.org>
> https://www.ietf.org/mailman/listinfo/i-d-announce
> Internet-Draft directories:http://www.ietf.org/shadow.html
> orftp://ftp.ietf.org/ietf/1shadow-sites.txt
>   
>


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    On 08/17/11 11:27, Muthu Arul Mozhi Perumal (mperumal) wrote:
    <blockquote
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      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|I
            think it should be up to the offer generator whether<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|he
            adds "a=<span class="SpellE">rtcp-mux</span>" to the first
            section or
            not.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Since
            the goal is to minimize the number of transport connections
            I think we should
            put something stronger to encourage the <span
              class="SpellE">offerer</span> and answerer
            to support <span class="SpellE">RTCP</span> multiplexing.
            At the minimum, a
            normative statement recommending the use of <span
              class="SpellE">RTCP</span>
            multiplexing with this grouping semantics would go a long
            way (symmetric <span class="SpellE">RTP</span> for <span
              class="SpellE">e.g</span> is a recommendation,
            but virtually all implementations support it).</span></p>
      </div>
    </blockquote>
    I agree that this is critical in the RTCWEB context. It may not be
    critical for all contexts.<br>
    I think that should be a matter the (not yet existing) draft
    referenced here should say:<br>
    <br>
    5.&nbsp; Data framing and securing<br>
    <br>
    &nbsp;&nbsp; SRTP [RFC3550] is used for transport of all real-time media.<br>
    <br>
    &nbsp;&nbsp; The detailed considerations for usage of functions from RTP and
    SRTP,<br>
    &nbsp;&nbsp; as well as for non-media real-time data, are given in &lt;WORKING
    GROUP<br>
    &nbsp;&nbsp; DRAFT "MEDIA TRANSPORTS"&gt;.<br>
    <br>
    (this is from the -overview document)<br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625FDC5@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|Do
            we need to state that if the first section has it, all <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|the
            other sections have to have it, or can we live with a <o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">|mixture?<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">If
            the first section has it, it can be applied to all other
            section within that
            group, so it shouldn't matter whether the reset of the
            sections have it or not.
            A mixture doesn't seem would serve any purpose.</span></p>
      </div>
    </blockquote>
    When TOGETHER is applied, the other sections' values doesn't matter
    at all.<br>
    When TOGETHER does NOT apply (fallback), it may matter.<br>
    I don't see any purpose with a mixture either. But we'll see what
    further review brings. <br>
    <blockquote
cite="mid:1D062974A4845E4D8A343C65380492020625FDC5@XMB-BGL-414.cisco.com"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;">Muthu<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 10pt; font-family:
            &quot;Courier New&quot;; color: windowtext;"><o:p>&nbsp;</o:p></span></p>
        <div style="border-width: medium medium medium 1.5pt;
          border-style: none none none solid; border-color:
          -moz-use-text-color -moz-use-text-color -moz-use-text-color
          blue; padding: 0in 0in 0in 4pt;">
          <div>
            <div style="border-right: medium none; border-width: 1pt
              medium medium; border-style: solid none none;
              border-color: rgb(181, 196, 223) -moz-use-text-color
              -moz-use-text-color; padding: 3pt 0in 0in;">
              <p class="MsoNormal"><b><span style="font-size: 10pt;
                    font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> Harald Alvestrand
                  [<a class="moz-txt-link-freetext" href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] <br>
                  <b>Sent:</b> Friday, August 12, 2011 8:29 PM<br>
                  <b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
                  <b>Cc:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> Re: [rtcweb] Fwd: I-D Action:
                  draft-alvestrand-one-rtp-00.txt<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><span style="">On
              08/12/11 16:12, Muthu Arul Mozhi Perumal (mperumal) wrote:
              <o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|In
              general, I think we should keep extensions as orthogonal</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|as
              possible, so the normative language here should just say</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|"establish
              RTCP sessions normally for the first
              section, and </span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">|don't
              do it for other sections".</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">Looks
              good to me. Should it also say "Within a TOGETHER
              group it is RECOMMENDED to multiplex RTCP on the same RTP
              port from the first
              section" or something stronger?</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="">I
              think it should be up to the offer generator whether he
              adds
              "a=rtcp-mux" to the first section or not.<br>
              Do we need to state that if the first section has it, all
              the other sections
              have to have it, or can we live with a mixture?<br>
            </span><span style="font-size: 10pt; font-family:
              &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><span
              style=""> <o:p></o:p></span></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">Muthu</span><o:p></o:p></p>
          <p class="MsoNormal"><span style="font-size: 10pt;
              font-family: &quot;Courier New&quot;; color: windowtext;">&nbsp;</span><o:p></o:p></p>
          <div style="border-width: medium medium medium 1.5pt;
            border-style: none none none solid; padding: 0in 0in 0in
            4pt; border-color: -moz-use-text-color -moz-use-text-color
            -moz-use-text-color blue;">
            <div>
              <div style="border-right: medium none; border-width: 1pt
                medium medium; border-style: solid none none; padding:
                3pt 0in 0in; border-color: -moz-use-text-color;">
                <p class="MsoNormal"><b><span style="font-size: 10pt;
                      font-family:
                      &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                      windowtext;">From:</span></b><span
                    style="font-size: 10pt; font-family:
                    &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                    windowtext;"> Harald Alvestrand [<a
                      moz-do-not-send="true"
                      href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>]
                    <br>
                    <b>Sent:</b> Friday, August 12, 2011 6:49 PM<br>
                    <b>To:</b> Muthu Arul Mozhi Perumal (mperumal)<br>
                    <b>Cc:</b> <a moz-do-not-send="true"
                      href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                    <b>Subject:</b> Re: [rtcweb] Fwd: I-D Action:
                    draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>
              </div>
            </div>
            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
            <p class="MsoNormal">On 08/12/2011 03:09 PM, Muthu Arul
              Mozhi Perumal (mperumal)
              wrote: <o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">A few comments:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">1) Section 4 has this:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">If the responder understands the semantics
                of the TOGETHER
                extension,</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">the parameters of the first section MUST be
                used to establish
                the RTP</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">session, and the parameters for the other
                sections MUST be
                ignored.</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">I think it needs to be:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">If the responder understands the semantics
                of the TOGETHER
                extension,</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">the parameters of the first section MUST be
                used to establish
                the RTP</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">and RCP session, and the parameters for the
                other sections
                MUST be </span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">ignored.</span><o:p></o:p></p>
            <p class="MsoNormal">Good point. Will fix. Normally, I think
              RTCWEB apps will use
              RTCP port multiplexing (RFC 5761), so I did not think of
              it.<br>
              In general, I think we should keep extensions as
              orthogonal as possible, so the
              normative language here should just say "establish RTCP
              sessions normally
              for the first section, and don't do it for other
              sections".<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">2) When the RTCP port is not the next
                higher (odd) port
                number following the RTP port described in the m-line,
                the "a=rtcp"
                attribute (defined in RFC-3605) is used to signal it, as
                in:</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">m=audio 49170 RTP/AVP 0</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">a=rtcp:53020 IN IP4 126.16.64.4</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">With the grouping semantics described in
                the draft, it seems
                this RTCP port should be taken only from the first
                section. So, this could be
                could be added to section 4.</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">3) RTCP could be multiplexed with RTP on a
                single port and
                signaled using the "a=rtcp-mux" attribute (defined in
                RFC 5761). When
                it is used together with the grouping semantics
                described in the draft, it
                seems all RTCP sessions of that group need to be
                multiplexed on the RTP port
                from the first section.</span><o:p></o:p></p>
            <p class="MsoNormal">There should be only one RTCP session
              established for all
              the TOGETHER m= lines, yes.<br style="">
              <!--[if !supportLineBreakNewLine]--><br style="">
              <!--[endif]--><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">Muthu</span><o:p></o:p></p>
            <p class="MsoNormal"><span style="font-size: 10pt;
                font-family: &quot;Courier New&quot;; color:
                windowtext;">&nbsp;</span><o:p></o:p></p>
            <div style="border-width: medium medium medium 1.5pt;
              border-style: none none none solid; padding: 0in 0in 0in
              4pt; border-color: -moz-use-text-color -moz-use-text-color
              -moz-use-text-color blue;">
              <div>
                <div style="border-right: medium none; border-width: 1pt
                  medium medium; border-style: solid none none; padding:
                  3pt 0in 0in; border-color: -moz-use-text-color;">
                  <p class="MsoNormal"><b><span style="font-size: 10pt;
                        font-family:
                        &quot;Tahoma&quot;,&quot;sans-serif&quot;;
                        color: windowtext;">From:</span></b><span
                      style="font-size: 10pt; font-family:
                      &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                      windowtext;"> <a moz-do-not-send="true"
                        href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                      [<a moz-do-not-send="true"
                        href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>]
                      <b>On
                        Behalf Of </b>Harald Alvestrand<br>
                      <b>Sent:</b> Thursday, August 11, 2011 5:02 PM<br>
                      <b>To:</b> <a moz-do-not-send="true"
                        href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                      <b>Subject:</b> [rtcweb] Fwd: I-D Action:
                      draft-alvestrand-one-rtp-00.txt</span><o:p></o:p></p>
                </div>
              </div>
              <p class="MsoNormal">&nbsp;<o:p></o:p></p>
              <p class="MsoNormal">I have taken some of the information
                I learned from the
                discussions in Quebec City about the issues of putting
                video and audio into the
                same RTP session and created a draft from them outlining
                a solution to the
                problem of signalling this configuration using SDP.<br>
                <br>
                Comments welcome, including the recommended context for
                processing the
                document; in a few days I'll send the same note to
                AVTCORE and/or MMUSIC.<br>
                <br>
                &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                Harald<br>
                <br>
                -------- Original Message -------- <o:p></o:p></p>
              <table class="MsoNormalTable" style="" border="0"
                cellpadding="0" cellspacing="0">
                <tbody>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Subject: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal">I-D Action:
                        draft-alvestrand-one-rtp-00.txt<o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Date: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal">Thu, 11 Aug 2011 04:28:09
                        -0700<o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>From: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>Reply-To: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                  <tr style="">
                    <td style="padding: 0in;" nowrap="nowrap"
                      valign="top">
                      <p class="MsoNormal" style="text-align: right;"
                        align="right"><b>To: </b><o:p></o:p></p>
                    </td>
                    <td style="padding: 0in;">
                      <p class="MsoNormal"><a moz-do-not-send="true"
                          href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><o:p></o:p></p>
                    </td>
                  </tr>
                </tbody>
              </table>
              <p class="MsoNormal" style="margin-bottom: 12pt;">&nbsp;<o:p></o:p></p>
              <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Title&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : SDP Grouping for Single RTP Sessions<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Author(s)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : Harald Tveit Alvestrand<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Filename&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : draft-alvestrand-one-rtp-00.txt<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Pages&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 8<o:p></o:p></pre>
              <pre>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Date&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; : 2011-08-11<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; This document describes an extension to the Session Description<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; Protocol (SDP) to describe RTP sessions where media of multiple top<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; level types, for example audio and video, are carried in the same RTP<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; session.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for<o:p></o:p></pre>
              <pre>&nbsp;&nbsp; consideration.<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>A URL for this Internet-Draft is:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>Internet-Drafts are also available by anonymous FTP at:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
              <pre>This Internet-Draft can be retrieved at:<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-00.txt</a><o:p></o:p></pre>
              <pre>_______________________________________________<o:p></o:p></pre>
              <pre>I-D-Announce mailing list<o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="mailto:I-D-Announce@ietf.org">I-D-Announce@ietf.org</a><o:p></o:p></pre>
              <pre><a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/i-d-announce">https://www.ietf.org/mailman/listinfo/i-d-announce</a><o:p></o:p></pre>
              <pre>Internet-Draft directories: <a moz-do-not-send="true" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a><o:p></o:p></pre>
              <pre>or <a moz-do-not-send="true" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a><o:p></o:p></pre>
              <pre>&nbsp;<o:p></o:p></pre>
            </div>
            <p class="MsoNormal">&nbsp;<o:p></o:p></p>
          </div>
          <p class="MsoNormal"><span style=""><o:p>&nbsp;</o:p></span></p>
        </div>
      </div>
    </blockquote>
    <br>
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From harald@alvestrand.no  Wed Aug 17 02:44:09 2011
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Thanks for all the feedback! I've incorporated this as best I could.
Now to post this to AVTCORE and MMUSIC for comments...

                   Harald

-------- Original Message --------
Subject: 	I-D Action: draft-alvestrand-one-rtp-01.txt
Date: 	Wed, 17 Aug 2011 01:48:18 -0700
From: 	internet-drafts@ietf.org
Reply-To: 	internet-drafts@ietf.org
To: 	i-d-announce@ietf.org



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

	Title           : SDP Grouping for Single RTP Sessions
	Author(s)       : Harald Tveit Alvestrand
	Filename        : draft-alvestrand-one-rtp-01.txt
	Pages           : 12
	Date            : 2011-08-17

    This document describes an extension to the Session Description
    Protocol (SDP) to describe RTP sessions where media of multiple top
    level types, for example audio and video, are carried in the same RTP
    session.

    This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
    consideration.



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

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

This Internet-Draft can be retrieved at:
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    Thanks for all the feedback! I've incorporated this as best I could.<br>
    Now to post this to AVTCORE and MMUSIC for comments...<br>
    <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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>I-D Action: draft-alvestrand-one-rtp-01.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Wed, 17 Aug 2011 01:48:18 -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">Reply-To:
          </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:i-d-announce@ietf.org">i-d-announce@ietf.org</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A New Internet-Draft is available from the on-line Internet-Drafts directories.

	Title           : SDP Grouping for Single RTP Sessions
	Author(s)       : Harald Tveit Alvestrand
	Filename        : draft-alvestrand-one-rtp-01.txt
	Pages           : 12
	Date            : 2011-08-17

   This document describes an extension to the Session Description
   Protocol (SDP) to describe RTP sessions where media of multiple top
   level types, for example audio and video, are carried in the same RTP
   session.

   This document is presented to the RTCWEB, AVTCORE and MMUSIC WGs for
   consideration.



A URL for this Internet-Draft is:
<a class="moz-txt-link-freetext" href="http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-01.txt">http://www.ietf.org/internet-drafts/draft-alvestrand-one-rtp-01.txt</a>

Internet-Drafts are also available by anonymous FTP at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/">ftp://ftp.ietf.org/internet-drafts/</a>

This Internet-Draft can be retrieved at:
<a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-01.txt">ftp://ftp.ietf.org/internet-drafts/draft-alvestrand-one-rtp-01.txt</a>
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Internet-Draft directories: <a class="moz-txt-link-freetext" href="http://www.ietf.org/shadow.html">http://www.ietf.org/shadow.html</a>
or <a class="moz-txt-link-freetext" href="ftp://ftp.ietf.org/ietf/1shadow-sites.txt">ftp://ftp.ietf.org/ietf/1shadow-sites.txt</a>

</pre>
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From john.elwell@siemens-enterprise.com  Fri Aug 19 00:33:12 2011
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Subject: [rtcweb] Use cases - recording and voicemail
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Sorry for lateness catching up on this thread.

I did indeed propose recording use cases. I see that Randell has applied th=
at to mail (voicemail / videomail) use cases. I would like to try to provid=
e some separation here.

The recording use cases I had in mind were cases where audio/video are capt=
ured/rendered locally (through camera/mic/display/speaker) and IN ADDITION =
are recorded, either locally or remotely. I tend to think of voicemail/vide=
omail as being a replacement for local capture/rendering, or to put it anot=
her way, the storage device acts as the capture/rendering device. So in the=
 voicemail/videomail case, recording is INSTEAD OF conventional capture/ren=
dering.

Voicemail/videomail can be local or remote. In the remote case, the inbound=
 call is diverted to a remote mailbox device. This is a signalling plane ma=
tter, and outside the scope of RTC-Web, I believe.

So the interesting case is local voicemail/videomail. It simply means that =
a mailboxes or files replace the conventional capture rendering devices. So=
 received audio and video can be sent to files, and files can be used as th=
e source of prompts or other information to be fed back to a caller. The ba=
sic requirement seems to be to use files as sources and sinks of media sent=
/received over RTP. One additional requirement is some means by which the c=
aller can control the mailbox - this is conventionally done by DTMF or voic=
e recognition - in fact these are the only methods available when a call co=
mes from PSTN. So this could also lead to a requirement for receiving DTMF.

Local recording, on the other hand, means that media captured from local de=
vices and sent via RTP and media received via RTP and rendered at local dev=
ices are also copied to local files.

Remote recording means that media captured from local devices and sent via =
RTP and media received via RTP and rendered at local devices are also sent =
via RTP to a remote recording device.

Basically, all these use cases, if we choose to progress them, have a gener=
al impact on requirements - some sort of flexibility in terms of where medi=
a is sourced and sunk, including the ability to use a file as a source/sink=
 and the ability to have multiple sinks. It seems that a well chosen API ar=
chitecture could accommodate these, but leaving these requirements until la=
ter might mean that the chosen API architecture is not sufficiently flexibl=
e to accommodate these requirements later. DTMF reception, if we decide we =
need it, would be a separate requirement.

So I would welcome feedback on which of these use cases are interesting (I =
have already heard some support for recording use cases), and I could propo=
se text if necessary.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 04 August 2011 16:04
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases: summary of status
>=20
> On 8/4/2011 3:57 AM, Stefan H=E5kansson LK wrote:
>=20
> >
> > B. New use cases
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > I noted (as not immediately dismissed) the following proposals:
> >
> >      1. Distributed music band, discussed at the webrtc meeting
> >      2. Use cases regarding different situations when being=20
> invited to a "session", e.g. browser open, browser open but=20
> another tab active, browser open but active in session,=20
> browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
> >      3. Different TURN provider scenarios (Cullen=20
> Jennings); discussed at the webrtc meeting
> >      4. More challenging NAT case (Matthew Kaufman); UDP=20
> blocked=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
> >      5. E911 (Paul Beaumont)=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
> >      6. Local Recording (John Ewell) at webrtc meeting
>=20
> Would this cover all "voicemail" and "videomail" cases?  I.e.=20
> having a client
> accept connections in the background if the call is not=20
> answered, optionally
> play a message, and record incoming audio/video, and allow=20
> the remote user to
> interact with it.  Also remote playback of messages.  If not=20
> (and if it's not
> covered by the current document, we need to add this usecase.
>=20
> Note that there are two variants: one local (local client=20
> handles it, which
> has more security issues around camera access and=20
> pre-approval), and one for
> remote recording (after some time with no answer or if not=20
> "registered" call
> is forwarded to a message server that answers it).  Note that=20
> there are
> security identity issues here (key chains, etc).
>=20
>=20
> >
> >      7. Remote recording (John) =20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
> >      8. Emergency access for disabled (Bernard Aboba)=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
> >      9. Clue use cases (Roni Even)=20
> http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
> >      10. Rohan red cross (Cullen Jennings); proposed a bit=20
> earlier=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
> >
> > Probably I have missed a few.
>=20
> I'd add:
>=20
>=20
> 11. Remote assistance (ala VNC or RDP) - User is helping=20
> another user on
> their computer with either view-only or view-with-control,=20
> either of just
> the browser of the the entire screen.  Computer audio is=20
> optionally sent
> as well.  They are optionally talking and/or viewing each=20
> other while this
> is occurring.  They may be transferring files over a reliable data
> connection during this session.
>=20
>=20
> Commentary: How often have you talked to your=20
> father/mother/brother/etc
> and tried to debug their computer problems remotely?  And=20
> getting them to
> install a specific remote-assistance package, and to use it,=20
> and get it to
> work through firewalls, can be very painful.  (There are=20
> other uses of this
> ability to copilot as well, including classroom instruction,=20
> distance learning,
> etc.)  This use-case enables someone to build a JS app to provide this
> functionality.  (Note that for the W3 and browser vendors=20
> there will be
> significant security issues to resolve.)  I think this is=20
> actually a fairly
> important use-case.
>=20
>=20
> Additional resulting requirements:
>=20
> * Need to be able to select a "video" source other than a=20
> camera (window or
> screen) (W3 issue)
>=20
> * Need to be able to send multiple video and audio streams=20
> from different
> sources (probably already covered, mostly W3 issue)
>=20
> * Need to be able to use a reliable data connection (for=20
> mouse/keyboard/etc
> control, plus file transfer)
>=20
>=20
> 12. Security camera/baby monitor usage - use a persistent or temporary
> connection to provide basic security camera operation,=20
> including switching
> cameras or mics, or with the ability to remotely request=20
> review of recent
> data recorded.  Should be able to switch to 2-way audio=20
> and/or video remotely.
>=20
>=20
> Additional requirements:
>=20
> * Need to be able to select a "video" source other than a=20
> camera (file)
> (W3 issue)
>=20
> * Remote control of camera/mic selection and enabling (W3 issue) - may
> require reliable data connection
>=20
> * May need control from JS over codecs used, at least voice=20
> vs audio, etc,
> and max video framerate/resolution/bandwidth (W3 issue largely?)
>=20
>=20
> >
> >
> > E. Opinion as individual
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > My personal opinion is that we at this stage should focus=20
> on the basic use cases, and solve those in a good way. To me=20
> that would mean that we should add 3 (to allow different=20
> providers) and 4 (if you can't connect no use case can be=20
> supported). I think that these use cases can be added as=20
> twists of/extensions to the first use case (Simple Video=20
> Communication Service).
> >
> > In my view it also means that we should add text and=20
> requirements for 2 to make sure that the communication=20
> capabilites we are adding to the browser is usable in=20
> everyday scenarios. Presumable this would lead requirements=20
> (e.g. background execution capability) that are transferred=20
> to other W3C WGs - not to requirements in the webrtc/rtcweb space.
> >
> > > From a W3C perspective it could also make sense to add a=20
> recording use case (as it seems that other W3C WGs rely on=20
> webrtc to provide an API for recording).
> >
> > The rest of the new proposed use cases, IMO, could be=20
> looked into in a second stage when we've established the basics.
>=20
> I'd very much want to include the "remote assistance" case,=20
> and I think the
> voicemail cases could be very important in overall acceptance=20
> and utility, especially
> given the issues over whether someone's machine is running=20
> and accepting calls.
>=20
> Security cam/baby-monitor is less important, but highlights=20
> some controls that
> we may want to expose to the JS over maximum=20
> bitrate/framerate/resolution/etc,
> and of course a slew of security issues for W3 to think about.
>=20
> --=20
> Randell Jesup
> randell-ietf@jesup.org
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

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

	Title           : Web Real-Time Communication Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-02.txt
	Pages           : 17
	Date            : 2011-08-19

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


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-02.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-02.txt

From stefan.lk.hakansson@ericsson.com  Fri Aug 19 01:07:52 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 19 Aug 2011 10:03:56 +0200
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-use-cases-and-requirements-02.txt
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We've uploaded an updated version of  draft-ietf-rtcweb-use-cases-and-requi=
rement, trying to take the discussion in Quebec and on email afterwards int=
o account.

>From the "Change Log" Section:
"Changes from draft-ietf-rtcweb-ucreqs-01

   o  Changed Intended status to Information
   o  Changed "Ipr" to "trust200902"
   o  Added use case "Simple video communication service, NAT/FW that
      blocks UDP", and derived new req F26
   o  Added use case "Distributed Music Band" and derived new req A17
   o  Added F24 as requirement derived from use case "Simple video
      communication service with inter-operator calling"
   o  Added section "Additional use cases"
   o  Added text about ID handling to multiparty with central server use
      case
   o  Re-phrased A1 slightly"

All feedback is welcome.

Stefan
________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of intern=
et-drafts@ietf.org [internet-drafts@ietf.org]
Sent: Friday, August 19, 2011 10:02 AM
To: i-d-announce@ietf.org
Cc: rtcweb@ietf.org
Subject: [rtcweb] I-D Action:   draft-ietf-rtcweb-use-cases-and-requirement=
s-02.txt

A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

        Title           : Web Real-Time Communication Use-cases and Require=
ments
        Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
        Filename        : draft-ietf-rtcweb-use-cases-and-requirements-02.t=
xt
        Pages           : 17
        Date            : 2011-08-19

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


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-02.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-02.txt
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Fri Aug 19 01:10:11 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 10:08:59 +0200
Thread-Topic: Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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John,

I was just uploading an updated version of the use case doc when receiving =
this, so it is not taken into account in that version.

Stefan
________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Elwell=
, John [john.elwell@siemens-enterprise.com]
Sent: Friday, August 19, 2011 9:34 AM
To: rtcweb@ietf.org; public-webrtc@w3.org
Subject: [rtcweb] Use cases - recording and voicemail

Sorry for lateness catching up on this thread.

I did indeed propose recording use cases. I see that Randell has applied th=
at to mail (voicemail / videomail) use cases. I would like to try to provid=
e some separation here.

The recording use cases I had in mind were cases where audio/video are capt=
ured/rendered locally (through camera/mic/display/speaker) and IN ADDITION =
are recorded, either locally or remotely. I tend to think of voicemail/vide=
omail as being a replacement for local capture/rendering, or to put it anot=
her way, the storage device acts as the capture/rendering device. So in the=
 voicemail/videomail case, recording is INSTEAD OF conventional capture/ren=
dering.

Voicemail/videomail can be local or remote. In the remote case, the inbound=
 call is diverted to a remote mailbox device. This is a signalling plane ma=
tter, and outside the scope of RTC-Web, I believe.

So the interesting case is local voicemail/videomail. It simply means that =
a mailboxes or files replace the conventional capture rendering devices. So=
 received audio and video can be sent to files, and files can be used as th=
e source of prompts or other information to be fed back to a caller. The ba=
sic requirement seems to be to use files as sources and sinks of media sent=
/received over RTP. One additional requirement is some means by which the c=
aller can control the mailbox - this is conventionally done by DTMF or voic=
e recognition - in fact these are the only methods available when a call co=
mes from PSTN. So this could also lead to a requirement for receiving DTMF.

Local recording, on the other hand, means that media captured from local de=
vices and sent via RTP and media received via RTP and rendered at local dev=
ices are also copied to local files.

Remote recording means that media captured from local devices and sent via =
RTP and media received via RTP and rendered at local devices are also sent =
via RTP to a remote recording device.

Basically, all these use cases, if we choose to progress them, have a gener=
al impact on requirements - some sort of flexibility in terms of where medi=
a is sourced and sunk, including the ability to use a file as a source/sink=
 and the ability to have multiple sinks. It seems that a well chosen API ar=
chitecture could accommodate these, but leaving these requirements until la=
ter might mean that the chosen API architecture is not sufficiently flexibl=
e to accommodate these requirements later. DTMF reception, if we decide we =
need it, would be a separate requirement.

So I would welcome feedback on which of these use cases are interesting (I =
have already heard some support for recording use cases), and I could propo=
se text if necessary.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


> -----Original Message-----
> From: rtcweb-bounces@ietf.org
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 04 August 2011 16:04
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases: summary of status
>
> On 8/4/2011 3:57 AM, Stefan H=E5kansson LK wrote:
>
> >
> > B. New use cases
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > I noted (as not immediately dismissed) the following proposals:
> >
> >      1. Distributed music band, discussed at the webrtc meeting
> >      2. Use cases regarding different situations when being
> invited to a "session", e.g. browser open, browser open but
> another tab active, browser open but active in session,
> browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
> >      3. Different TURN provider scenarios (Cullen
> Jennings); discussed at the webrtc meeting
> >      4. More challenging NAT case (Matthew Kaufman); UDP
> blocked
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
> >      5. E911 (Paul Beaumont)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
> >      6. Local Recording (John Ewell) at webrtc meeting
>
> Would this cover all "voicemail" and "videomail" cases?  I.e.
> having a client
> accept connections in the background if the call is not
> answered, optionally
> play a message, and record incoming audio/video, and allow
> the remote user to
> interact with it.  Also remote playback of messages.  If not
> (and if it's not
> covered by the current document, we need to add this usecase.
>
> Note that there are two variants: one local (local client
> handles it, which
> has more security issues around camera access and
> pre-approval), and one for
> remote recording (after some time with no answer or if not
> "registered" call
> is forwarded to a message server that answers it).  Note that
> there are
> security identity issues here (key chains, etc).
>
>
> >
> >      7. Remote recording (John)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
> >      8. Emergency access for disabled (Bernard Aboba)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
> >      9. Clue use cases (Roni Even)
> http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
> >      10. Rohan red cross (Cullen Jennings); proposed a bit
> earlier
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
> >
> > Probably I have missed a few.
>
> I'd add:
>
>
> 11. Remote assistance (ala VNC or RDP) - User is helping
> another user on
> their computer with either view-only or view-with-control,
> either of just
> the browser of the the entire screen.  Computer audio is
> optionally sent
> as well.  They are optionally talking and/or viewing each
> other while this
> is occurring.  They may be transferring files over a reliable data
> connection during this session.
>
>
> Commentary: How often have you talked to your
> father/mother/brother/etc
> and tried to debug their computer problems remotely?  And
> getting them to
> install a specific remote-assistance package, and to use it,
> and get it to
> work through firewalls, can be very painful.  (There are
> other uses of this
> ability to copilot as well, including classroom instruction,
> distance learning,
> etc.)  This use-case enables someone to build a JS app to provide this
> functionality.  (Note that for the W3 and browser vendors
> there will be
> significant security issues to resolve.)  I think this is
> actually a fairly
> important use-case.
>
>
> Additional resulting requirements:
>
> * Need to be able to select a "video" source other than a
> camera (window or
> screen) (W3 issue)
>
> * Need to be able to send multiple video and audio streams
> from different
> sources (probably already covered, mostly W3 issue)
>
> * Need to be able to use a reliable data connection (for
> mouse/keyboard/etc
> control, plus file transfer)
>
>
> 12. Security camera/baby monitor usage - use a persistent or temporary
> connection to provide basic security camera operation,
> including switching
> cameras or mics, or with the ability to remotely request
> review of recent
> data recorded.  Should be able to switch to 2-way audio
> and/or video remotely.
>
>
> Additional requirements:
>
> * Need to be able to select a "video" source other than a
> camera (file)
> (W3 issue)
>
> * Remote control of camera/mic selection and enabling (W3 issue) - may
> require reliable data connection
>
> * May need control from JS over codecs used, at least voice
> vs audio, etc,
> and max video framerate/resolution/bandwidth (W3 issue largely?)
>
>
> >
> >
> > E. Opinion as individual
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > My personal opinion is that we at this stage should focus
> on the basic use cases, and solve those in a good way. To me
> that would mean that we should add 3 (to allow different
> providers) and 4 (if you can't connect no use case can be
> supported). I think that these use cases can be added as
> twists of/extensions to the first use case (Simple Video
> Communication Service).
> >
> > In my view it also means that we should add text and
> requirements for 2 to make sure that the communication
> capabilites we are adding to the browser is usable in
> everyday scenarios. Presumable this would lead requirements
> (e.g. background execution capability) that are transferred
> to other W3C WGs - not to requirements in the webrtc/rtcweb space.
> >
> > > From a W3C perspective it could also make sense to add a
> recording use case (as it seems that other W3C WGs rely on
> webrtc to provide an API for recording).
> >
> > The rest of the new proposed use cases, IMO, could be
> looked into in a second stage when we've established the basics.
>
> I'd very much want to include the "remote assistance" case,
> and I think the
> voicemail cases could be very important in overall acceptance
> and utility, especially
> given the issues over whether someone's machine is running
> and accepting calls.
>
> Security cam/baby-monitor is less important, but highlights
> some controls that
> we may want to expose to the JS over maximum
> bitrate/framerate/resolution/etc,
> and of course a slew of security issues for W3 to think about.
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> 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  Fri Aug 19 02:49:02 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 11:49:56 +0200
Thread-Topic: Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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John,

this was a good way sort things up.

I my view we should definitely support local recording of streams (regardle=
ss of if they are generated by local devices or received via RTP), and this=
 could be done in parallel to rendering them or not (up to the app).

The recorded media should also be possible to render locally (be the source=
 for a video element).

I'm less sure about that the recorded media should also be an RTP source - =
couldn't you just as well send the file over and then play it at the remote=
 end?

Stefan
________________________________________
From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Elwell=
, John [john.elwell@siemens-enterprise.com]
Sent: Friday, August 19, 2011 9:34 AM
To: rtcweb@ietf.org; public-webrtc@w3.org
Subject: [rtcweb] Use cases - recording and voicemail

Sorry for lateness catching up on this thread.

I did indeed propose recording use cases. I see that Randell has applied th=
at to mail (voicemail / videomail) use cases. I would like to try to provid=
e some separation here.

The recording use cases I had in mind were cases where audio/video are capt=
ured/rendered locally (through camera/mic/display/speaker) and IN ADDITION =
are recorded, either locally or remotely. I tend to think of voicemail/vide=
omail as being a replacement for local capture/rendering, or to put it anot=
her way, the storage device acts as the capture/rendering device. So in the=
 voicemail/videomail case, recording is INSTEAD OF conventional capture/ren=
dering.

Voicemail/videomail can be local or remote. In the remote case, the inbound=
 call is diverted to a remote mailbox device. This is a signalling plane ma=
tter, and outside the scope of RTC-Web, I believe.

So the interesting case is local voicemail/videomail. It simply means that =
a mailboxes or files replace the conventional capture rendering devices. So=
 received audio and video can be sent to files, and files can be used as th=
e source of prompts or other information to be fed back to a caller. The ba=
sic requirement seems to be to use files as sources and sinks of media sent=
/received over RTP. One additional requirement is some means by which the c=
aller can control the mailbox - this is conventionally done by DTMF or voic=
e recognition - in fact these are the only methods available when a call co=
mes from PSTN. So this could also lead to a requirement for receiving DTMF.

Local recording, on the other hand, means that media captured from local de=
vices and sent via RTP and media received via RTP and rendered at local dev=
ices are also copied to local files.

Remote recording means that media captured from local devices and sent via =
RTP and media received via RTP and rendered at local devices are also sent =
via RTP to a remote recording device.

Basically, all these use cases, if we choose to progress them, have a gener=
al impact on requirements - some sort of flexibility in terms of where medi=
a is sourced and sunk, including the ability to use a file as a source/sink=
 and the ability to have multiple sinks. It seems that a well chosen API ar=
chitecture could accommodate these, but leaving these requirements until la=
ter might mean that the chosen API architecture is not sufficiently flexibl=
e to accommodate these requirements later. DTMF reception, if we decide we =
need it, would be a separate requirement.

So I would welcome feedback on which of these use cases are interesting (I =
have already heard some support for recording use cases), and I could propo=
se text if necessary.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


> -----Original Message-----
> From: rtcweb-bounces@ietf.org
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 04 August 2011 16:04
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases: summary of status
>
> On 8/4/2011 3:57 AM, Stefan H=E5kansson LK wrote:
>
> >
> > B. New use cases
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > I noted (as not immediately dismissed) the following proposals:
> >
> >      1. Distributed music band, discussed at the webrtc meeting
> >      2. Use cases regarding different situations when being
> invited to a "session", e.g. browser open, browser open but
> another tab active, browser open but active in session,
> browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
> >      3. Different TURN provider scenarios (Cullen
> Jennings); discussed at the webrtc meeting
> >      4. More challenging NAT case (Matthew Kaufman); UDP
> blocked
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
> >      5. E911 (Paul Beaumont)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
> >      6. Local Recording (John Ewell) at webrtc meeting
>
> Would this cover all "voicemail" and "videomail" cases?  I.e.
> having a client
> accept connections in the background if the call is not
> answered, optionally
> play a message, and record incoming audio/video, and allow
> the remote user to
> interact with it.  Also remote playback of messages.  If not
> (and if it's not
> covered by the current document, we need to add this usecase.
>
> Note that there are two variants: one local (local client
> handles it, which
> has more security issues around camera access and
> pre-approval), and one for
> remote recording (after some time with no answer or if not
> "registered" call
> is forwarded to a message server that answers it).  Note that
> there are
> security identity issues here (key chains, etc).
>
>
> >
> >      7. Remote recording (John)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
> >      8. Emergency access for disabled (Bernard Aboba)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
> >      9. Clue use cases (Roni Even)
> http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
> >      10. Rohan red cross (Cullen Jennings); proposed a bit
> earlier
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
> >
> > Probably I have missed a few.
>
> I'd add:
>
>
> 11. Remote assistance (ala VNC or RDP) - User is helping
> another user on
> their computer with either view-only or view-with-control,
> either of just
> the browser of the the entire screen.  Computer audio is
> optionally sent
> as well.  They are optionally talking and/or viewing each
> other while this
> is occurring.  They may be transferring files over a reliable data
> connection during this session.
>
>
> Commentary: How often have you talked to your
> father/mother/brother/etc
> and tried to debug their computer problems remotely?  And
> getting them to
> install a specific remote-assistance package, and to use it,
> and get it to
> work through firewalls, can be very painful.  (There are
> other uses of this
> ability to copilot as well, including classroom instruction,
> distance learning,
> etc.)  This use-case enables someone to build a JS app to provide this
> functionality.  (Note that for the W3 and browser vendors
> there will be
> significant security issues to resolve.)  I think this is
> actually a fairly
> important use-case.
>
>
> Additional resulting requirements:
>
> * Need to be able to select a "video" source other than a
> camera (window or
> screen) (W3 issue)
>
> * Need to be able to send multiple video and audio streams
> from different
> sources (probably already covered, mostly W3 issue)
>
> * Need to be able to use a reliable data connection (for
> mouse/keyboard/etc
> control, plus file transfer)
>
>
> 12. Security camera/baby monitor usage - use a persistent or temporary
> connection to provide basic security camera operation,
> including switching
> cameras or mics, or with the ability to remotely request
> review of recent
> data recorded.  Should be able to switch to 2-way audio
> and/or video remotely.
>
>
> Additional requirements:
>
> * Need to be able to select a "video" source other than a
> camera (file)
> (W3 issue)
>
> * Remote control of camera/mic selection and enabling (W3 issue) - may
> require reliable data connection
>
> * May need control from JS over codecs used, at least voice
> vs audio, etc,
> and max video framerate/resolution/bandwidth (W3 issue largely?)
>
>
> >
> >
> > E. Opinion as individual
> > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > My personal opinion is that we at this stage should focus
> on the basic use cases, and solve those in a good way. To me
> that would mean that we should add 3 (to allow different
> providers) and 4 (if you can't connect no use case can be
> supported). I think that these use cases can be added as
> twists of/extensions to the first use case (Simple Video
> Communication Service).
> >
> > In my view it also means that we should add text and
> requirements for 2 to make sure that the communication
> capabilites we are adding to the browser is usable in
> everyday scenarios. Presumable this would lead requirements
> (e.g. background execution capability) that are transferred
> to other W3C WGs - not to requirements in the webrtc/rtcweb space.
> >
> > > From a W3C perspective it could also make sense to add a
> recording use case (as it seems that other W3C WGs rely on
> webrtc to provide an API for recording).
> >
> > The rest of the new proposed use cases, IMO, could be
> looked into in a second stage when we've established the basics.
>
> I'd very much want to include the "remote assistance" case,
> and I think the
> voicemail cases could be very important in overall acceptance
> and utility, especially
> given the issues over whether someone's machine is running
> and accepting calls.
>
> Security cam/baby-monitor is less important, but highlights
> some controls that
> we may want to expose to the JS over maximum
> bitrate/framerate/resolution/etc,
> and of course a slew of security issues for W3 to think about.
>
> --
> Randell Jesup
> randell-ietf@jesup.org
>
> _______________________________________________
> 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 john.elwell@siemens-enterprise.com  Fri Aug 19 05:19:13 2011
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To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 14:20:06 +0200
Thread-Topic: Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Stefan,

> -----Original Message-----
> From: Stefan H=E5kansson LK [mailto:stefan.lk.hakansson@ericsson.com]=20
> Sent: 19 August 2011 10:50
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: Use cases - recording and voicemail
>=20
> John,
>=20
> this was a good way sort things up.
>=20
> I my view we should definitely support local recording of=20
> streams (regardless of if they are generated by local devices=20
> or received via RTP), and this could be done in parallel to=20
> rendering them or not (up to the app).
>=20
> The recorded media should also be possible to render locally=20
> (be the source for a video element).
>=20
> I'm less sure about that the recorded media should also be an=20
> RTP source - couldn't you just as well send the file over and=20
> then play it at the remote end?
[JRE] Yes, the file could be streamed across, but there are folks who want =
it to get across more or less in real time, e.g., where the recorder is per=
forming real-time analytics, perhaps in a contact centre. This is the basis=
 for the work done in the IETF SIPREC WG. The same considerations that requ=
ire RTP browser-to-browser for RTC-Web also dictate RTP as the transport fo=
r sending media to a recorder.

In terms of what impact this would have on the solution, perhaps it would h=
ave very little impact. Perhaps it could be treated as two separate session=
s:
- one that exchanges audio/video between capture/rendering devices and the =
remote entity and copies to a local file; and
- one that takes audio/video from the local file and sends it to a remote r=
ecorder.
The main requirement then is to be able to read from the local file at the =
same time as writing to the local file.

John

>=20
> Stefan
> ________________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On=20
> Behalf Of Elwell, John [john.elwell@siemens-enterprise.com]
> Sent: Friday, August 19, 2011 9:34 AM
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: [rtcweb] Use cases - recording and voicemail
>=20
> Sorry for lateness catching up on this thread.
>=20
> I did indeed propose recording use cases. I see that Randell=20
> has applied that to mail (voicemail / videomail) use cases. I=20
> would like to try to provide some separation here.
>=20
> The recording use cases I had in mind were cases where=20
> audio/video are captured/rendered locally (through=20
> camera/mic/display/speaker) and IN ADDITION are recorded,=20
> either locally or remotely. I tend to think of=20
> voicemail/videomail as being a replacement for local=20
> capture/rendering, or to put it another way, the storage=20
> device acts as the capture/rendering device. So in the=20
> voicemail/videomail case, recording is INSTEAD OF=20
> conventional capture/rendering.
>=20
> Voicemail/videomail can be local or remote. In the remote=20
> case, the inbound call is diverted to a remote mailbox=20
> device. This is a signalling plane matter, and outside the=20
> scope of RTC-Web, I believe.
>=20
> So the interesting case is local voicemail/videomail. It=20
> simply means that a mailboxes or files replace the=20
> conventional capture rendering devices. So received audio and=20
> video can be sent to files, and files can be used as the=20
> source of prompts or other information to be fed back to a=20
> caller. The basic requirement seems to be to use files as=20
> sources and sinks of media sent/received over RTP. One=20
> additional requirement is some means by which the caller can=20
> control the mailbox - this is conventionally done by DTMF or=20
> voice recognition - in fact these are the only methods=20
> available when a call comes from PSTN. So this could also=20
> lead to a requirement for receiving DTMF.
>=20
> Local recording, on the other hand, means that media captured=20
> from local devices and sent via RTP and media received via=20
> RTP and rendered at local devices are also copied to local files.
>=20
> Remote recording means that media captured from local devices=20
> and sent via RTP and media received via RTP and rendered at=20
> local devices are also sent via RTP to a remote recording device.
>=20
> Basically, all these use cases, if we choose to progress=20
> them, have a general impact on requirements - some sort of=20
> flexibility in terms of where media is sourced and sunk,=20
> including the ability to use a file as a source/sink and the=20
> ability to have multiple sinks. It seems that a well chosen=20
> API architecture could accommodate these, but leaving these=20
> requirements until later might mean that the chosen API=20
> architecture is not sufficiently flexible to accommodate=20
> these requirements later. DTMF reception, if we decide we=20
> need it, would be a separate requirement.
>=20
> So I would welcome feedback on which of these use cases are=20
> interesting (I have already heard some support for recording=20
> use cases), and I could propose text if necessary.
>=20
> John
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of Siemens AG.
>=20
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org
> > [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> > Sent: 04 August 2011 16:04
> > To: rtcweb@ietf.org; public-webrtc@w3.org
> > Subject: Re: [rtcweb] Use cases: summary of status
> >
> > On 8/4/2011 3:57 AM, Stefan H=E5kansson LK wrote:
> >
> > >
> > > B. New use cases
> > > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > > I noted (as not immediately dismissed) the following proposals:
> > >
> > >      1. Distributed music band, discussed at the webrtc meeting
> > >      2. Use cases regarding different situations when being
> > invited to a "session", e.g. browser open, browser open but
> > another tab active, browser open but active in session,
> > browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
> > >      3. Different TURN provider scenarios (Cullen
> > Jennings); discussed at the webrtc meeting
> > >      4. More challenging NAT case (Matthew Kaufman); UDP
> > blocked
> > http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
> > >      5. E911 (Paul Beaumont)
> > http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
> > >      6. Local Recording (John Ewell) at webrtc meeting
> >
> > Would this cover all "voicemail" and "videomail" cases?  I.e.
> > having a client
> > accept connections in the background if the call is not
> > answered, optionally
> > play a message, and record incoming audio/video, and allow
> > the remote user to
> > interact with it.  Also remote playback of messages.  If not
> > (and if it's not
> > covered by the current document, we need to add this usecase.
> >
> > Note that there are two variants: one local (local client
> > handles it, which
> > has more security issues around camera access and
> > pre-approval), and one for
> > remote recording (after some time with no answer or if not
> > "registered" call
> > is forwarded to a message server that answers it).  Note that
> > there are
> > security identity issues here (key chains, etc).
> >
> >
> > >
> > >      7. Remote recording (John)
> > http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
> > >      8. Emergency access for disabled (Bernard Aboba)
> > http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
> > >      9. Clue use cases (Roni Even)
> > http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
> > >      10. Rohan red cross (Cullen Jennings); proposed a bit
> > earlier
> > http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
> > >
> > > Probably I have missed a few.
> >
> > I'd add:
> >
> >
> > 11. Remote assistance (ala VNC or RDP) - User is helping
> > another user on
> > their computer with either view-only or view-with-control,
> > either of just
> > the browser of the the entire screen.  Computer audio is
> > optionally sent
> > as well.  They are optionally talking and/or viewing each
> > other while this
> > is occurring.  They may be transferring files over a reliable data
> > connection during this session.
> >
> >
> > Commentary: How often have you talked to your
> > father/mother/brother/etc
> > and tried to debug their computer problems remotely?  And
> > getting them to
> > install a specific remote-assistance package, and to use it,
> > and get it to
> > work through firewalls, can be very painful.  (There are
> > other uses of this
> > ability to copilot as well, including classroom instruction,
> > distance learning,
> > etc.)  This use-case enables someone to build a JS app to=20
> provide this
> > functionality.  (Note that for the W3 and browser vendors
> > there will be
> > significant security issues to resolve.)  I think this is
> > actually a fairly
> > important use-case.
> >
> >
> > Additional resulting requirements:
> >
> > * Need to be able to select a "video" source other than a
> > camera (window or
> > screen) (W3 issue)
> >
> > * Need to be able to send multiple video and audio streams
> > from different
> > sources (probably already covered, mostly W3 issue)
> >
> > * Need to be able to use a reliable data connection (for
> > mouse/keyboard/etc
> > control, plus file transfer)
> >
> >
> > 12. Security camera/baby monitor usage - use a persistent=20
> or temporary
> > connection to provide basic security camera operation,
> > including switching
> > cameras or mics, or with the ability to remotely request
> > review of recent
> > data recorded.  Should be able to switch to 2-way audio
> > and/or video remotely.
> >
> >
> > Additional requirements:
> >
> > * Need to be able to select a "video" source other than a
> > camera (file)
> > (W3 issue)
> >
> > * Remote control of camera/mic selection and enabling (W3=20
> issue) - may
> > require reliable data connection
> >
> > * May need control from JS over codecs used, at least voice
> > vs audio, etc,
> > and max video framerate/resolution/bandwidth (W3 issue largely?)
> >
> >
> > >
> > >
> > > E. Opinion as individual
> > > =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> > > My personal opinion is that we at this stage should focus
> > on the basic use cases, and solve those in a good way. To me
> > that would mean that we should add 3 (to allow different
> > providers) and 4 (if you can't connect no use case can be
> > supported). I think that these use cases can be added as
> > twists of/extensions to the first use case (Simple Video
> > Communication Service).
> > >
> > > In my view it also means that we should add text and
> > requirements for 2 to make sure that the communication
> > capabilites we are adding to the browser is usable in
> > everyday scenarios. Presumable this would lead requirements
> > (e.g. background execution capability) that are transferred
> > to other W3C WGs - not to requirements in the webrtc/rtcweb space.
> > >
> > > > From a W3C perspective it could also make sense to add a
> > recording use case (as it seems that other W3C WGs rely on
> > webrtc to provide an API for recording).
> > >
> > > The rest of the new proposed use cases, IMO, could be
> > looked into in a second stage when we've established the basics.
> >
> > I'd very much want to include the "remote assistance" case,
> > and I think the
> > voicemail cases could be very important in overall acceptance
> > and utility, especially
> > given the issues over whether someone's machine is running
> > and accepting calls.
> >
> > Security cam/baby-monitor is less important, but highlights
> > some controls that
> > we may want to expose to the JS over maximum
> > bitrate/framerate/resolution/etc,
> > and of course a slew of security issues for W3 to think about.
> >
> > --
> > Randell Jesup
> > randell-ietf@jesup.org
> >
> > _______________________________________________
> > 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  Fri Aug 19 05:32:47 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 14:33:35 +0200
Thread-Topic: Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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John,
>> I'm less sure about that the recorded media should also be an
>> RTP source - couldn't you just as well send the file over and
>> then play it at the remote end?
>[JRE] Yes, the file could be streamed across, but there are folks who want=
 it to get across more or less in real time, e.g., where the recorder is pe=
rforming real-time analytics, perhaps in a >contact centre. This is the bas=
is for the work done in the IETF SIPREC WG. The same considerations that re=
quire RTP browser-to-browser for RTC-Web also dictate RTP as the transport =
for >sending media to a recorder.
[Stefan] If you want to do stuff remotely in real time, could you not just =
avoid recording locally? Regardless of if the analysis is to happen on a lo=
cally generated stream, or on a stream received via RTP, that stream can be=
 streamed to the analysing server (via a p2p-connection), right? (Probably =
I am missing something).

Stefan=

From pkyzivat@alum.mit.edu  Fri Aug 19 06:06:57 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/19/11 5:49 AM, Stefan Håkansson LK wrote:
> John,
>
> this was a good way sort things up.
>
> I my view we should definitely support local recording of streams (regardless of if they are generated by local devices or received via RTP), and this could be done in parallel to rendering them or not (up to the app).
>
> The recorded media should also be possible to render locally (be the source for a video element).
>
> I'm less sure about that the recorded media should also be an RTP source - couldn't you just as well send the file over and then play it at the remote end?

I think discussing this in terms of a "file" is an over simplification.
While it could be a file, it could also be synthesized, or assembled 
from pieces, interactively as a result of received media or signaling. 
In such cases you can't just "send the file".

Also, sending the file only works if the recipient can always deal with 
a file equally to receiving an RTP stream. That seems like a burden to 
levy on all possible recipients. For instance, it would mean that 
rtcweb:pstn gateways would all require the capability to buffer and 
playback audio/video files.

So perhaps what is needed is the concept of a software agent that can 
take on the role of a local source/sink of media streams.

	Thanks,
	Paul

> Stefan
> ________________________________________
> From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On Behalf Of Elwell, John [john.elwell@siemens-enterprise.com]
> Sent: Friday, August 19, 2011 9:34 AM
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: [rtcweb] Use cases - recording and voicemail
>
> Sorry for lateness catching up on this thread.
>
> I did indeed propose recording use cases. I see that Randell has applied that to mail (voicemail / videomail) use cases. I would like to try to provide some separation here.
>
> The recording use cases I had in mind were cases where audio/video are captured/rendered locally (through camera/mic/display/speaker) and IN ADDITION are recorded, either locally or remotely. I tend to think of voicemail/videomail as being a replacement for local capture/rendering, or to put it another way, the storage device acts as the capture/rendering device. So in the voicemail/videomail case, recording is INSTEAD OF conventional capture/rendering.
>
> Voicemail/videomail can be local or remote. In the remote case, the inbound call is diverted to a remote mailbox device. This is a signalling plane matter, and outside the scope of RTC-Web, I believe.
>
> So the interesting case is local voicemail/videomail. It simply means that a mailboxes or files replace the conventional capture rendering devices. So received audio and video can be sent to files, and files can be used as the source of prompts or other information to be fed back to a caller. The basic requirement seems to be to use files as sources and sinks of media sent/received over RTP. One additional requirement is some means by which the caller can control the mailbox - this is conventionally done by DTMF or voice recognition - in fact these are the only methods available when a call comes from PSTN. So this could also lead to a requirement for receiving DTMF.
>
> Local recording, on the other hand, means that media captured from local devices and sent via RTP and media received via RTP and rendered at local devices are also copied to local files.
>
> Remote recording means that media captured from local devices and sent via RTP and media received via RTP and rendered at local devices are also sent via RTP to a remote recording device.
>
> Basically, all these use cases, if we choose to progress them, have a general impact on requirements - some sort of flexibility in terms of where media is sourced and sunk, including the ability to use a file as a source/sink and the ability to have multiple sinks. It seems that a well chosen API architecture could accommodate these, but leaving these requirements until later might mean that the chosen API architecture is not sufficiently flexible to accommodate these requirements later. DTMF reception, if we decide we need it, would be a separate requirement.
>
> So I would welcome feedback on which of these use cases are interesting (I have already heard some support for recording use cases), and I could propose text if necessary.
>
> John
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
>> Sent: 04 August 2011 16:04
>> To: rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: Re: [rtcweb] Use cases: summary of status
>>
>> On 8/4/2011 3:57 AM, Stefan Håkansson LK wrote:
>>
>>>
>>> B. New use cases
>>> ===========
>>> I noted (as not immediately dismissed) the following proposals:
>>>
>>>       1. Distributed music band, discussed at the webrtc meeting
>>>       2. Use cases regarding different situations when being
>> invited to a "session", e.g. browser open, browser open but
>> another tab active, browser open but active in session,
>> browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
>>>       3. Different TURN provider scenarios (Cullen
>> Jennings); discussed at the webrtc meeting
>>>       4. More challenging NAT case (Matthew Kaufman); UDP
>> blocked
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
>>>       5. E911 (Paul Beaumont)
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
>>>       6. Local Recording (John Ewell) at webrtc meeting
>>
>> Would this cover all "voicemail" and "videomail" cases?  I.e.
>> having a client
>> accept connections in the background if the call is not
>> answered, optionally
>> play a message, and record incoming audio/video, and allow
>> the remote user to
>> interact with it.  Also remote playback of messages.  If not
>> (and if it's not
>> covered by the current document, we need to add this usecase.
>>
>> Note that there are two variants: one local (local client
>> handles it, which
>> has more security issues around camera access and
>> pre-approval), and one for
>> remote recording (after some time with no answer or if not
>> "registered" call
>> is forwarded to a message server that answers it).  Note that
>> there are
>> security identity issues here (key chains, etc).
>>
>>
>>>
>>>       7. Remote recording (John)
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
>>>       8. Emergency access for disabled (Bernard Aboba)
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
>>>       9. Clue use cases (Roni Even)
>> http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
>>>       10. Rohan red cross (Cullen Jennings); proposed a bit
>> earlier
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
>>>
>>> Probably I have missed a few.
>>
>> I'd add:
>>
>>
>> 11. Remote assistance (ala VNC or RDP) - User is helping
>> another user on
>> their computer with either view-only or view-with-control,
>> either of just
>> the browser of the the entire screen.  Computer audio is
>> optionally sent
>> as well.  They are optionally talking and/or viewing each
>> other while this
>> is occurring.  They may be transferring files over a reliable data
>> connection during this session.
>>
>>
>> Commentary: How often have you talked to your
>> father/mother/brother/etc
>> and tried to debug their computer problems remotely?  And
>> getting them to
>> install a specific remote-assistance package, and to use it,
>> and get it to
>> work through firewalls, can be very painful.  (There are
>> other uses of this
>> ability to copilot as well, including classroom instruction,
>> distance learning,
>> etc.)  This use-case enables someone to build a JS app to provide this
>> functionality.  (Note that for the W3 and browser vendors
>> there will be
>> significant security issues to resolve.)  I think this is
>> actually a fairly
>> important use-case.
>>
>>
>> Additional resulting requirements:
>>
>> * Need to be able to select a "video" source other than a
>> camera (window or
>> screen) (W3 issue)
>>
>> * Need to be able to send multiple video and audio streams
>> from different
>> sources (probably already covered, mostly W3 issue)
>>
>> * Need to be able to use a reliable data connection (for
>> mouse/keyboard/etc
>> control, plus file transfer)
>>
>>
>> 12. Security camera/baby monitor usage - use a persistent or temporary
>> connection to provide basic security camera operation,
>> including switching
>> cameras or mics, or with the ability to remotely request
>> review of recent
>> data recorded.  Should be able to switch to 2-way audio
>> and/or video remotely.
>>
>>
>> Additional requirements:
>>
>> * Need to be able to select a "video" source other than a
>> camera (file)
>> (W3 issue)
>>
>> * Remote control of camera/mic selection and enabling (W3 issue) - may
>> require reliable data connection
>>
>> * May need control from JS over codecs used, at least voice
>> vs audio, etc,
>> and max video framerate/resolution/bandwidth (W3 issue largely?)
>>
>>
>>>
>>>
>>> E. Opinion as individual
>>> ===============
>>> My personal opinion is that we at this stage should focus
>> on the basic use cases, and solve those in a good way. To me
>> that would mean that we should add 3 (to allow different
>> providers) and 4 (if you can't connect no use case can be
>> supported). I think that these use cases can be added as
>> twists of/extensions to the first use case (Simple Video
>> Communication Service).
>>>
>>> In my view it also means that we should add text and
>> requirements for 2 to make sure that the communication
>> capabilites we are adding to the browser is usable in
>> everyday scenarios. Presumable this would lead requirements
>> (e.g. background execution capability) that are transferred
>> to other W3C WGs - not to requirements in the webrtc/rtcweb space.
>>>
>>>>  From a W3C perspective it could also make sense to add a
>> recording use case (as it seems that other W3C WGs rely on
>> webrtc to provide an API for recording).
>>>
>>> The rest of the new proposed use cases, IMO, could be
>> looked into in a second stage when we've established the basics.
>>
>> I'd very much want to include the "remote assistance" case,
>> and I think the
>> voicemail cases could be very important in overall acceptance
>> and utility, especially
>> given the issues over whether someone's machine is running
>> and accepting calls.
>>
>> Security cam/baby-monitor is less important, but highlights
>> some controls that
>> we may want to expose to the JS over maximum
>> bitrate/framerate/resolution/etc,
>> and of course a slew of security issues for W3 to think about.
>>
>> --
>> Randell Jesup
>> randell-ietf@jesup.org
>>
>> _______________________________________________
>> 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 stefan.lk.hakansson@ericsson.com  Fri Aug 19 06:20:35 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 19 Aug 2011 15:21:30 +0200
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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>> I'm less sure about that the recorded media should also be an RTP source=
 - couldn't you just as well send the file over and then play it at the rem=
ote end?
>
>I think discussing this in terms of a "file" is an over simplification.
>While it could be a file, it could also be synthesized, or assembled
>from pieces, interactively as a result of received media or signaling.
>In such cases you can't just "send the file".
Maybe it was an over simplification, but synthesizing/assembling etc. sound=
 a bit overly complex to me, at least in a first version. Just my personal =
opinion. In any case we would have to specify more precisely what should be=
 possible.

From john.elwell@siemens-enterprise.com  Fri Aug 19 06:24:43 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 15:25:35 +0200
Thread-Topic: Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Stefan,

No, I am not suggesting both local recording and remote recording at the sa=
me time. Sometimes local recording will be required, sometimes remote recor=
ding.

However, I did suggest (in other text in my previous message) that one poss=
ible solution might be to record locally and use a second RTC-Web session t=
o transmit from the local file to the remote recorder. What I failed to say=
 was that in this case the local file would be a temporary repository - jus=
t a buffer between the two sessions.

John


> -----Original Message-----
> From: Stefan H=E5kansson LK [mailto:stefan.lk.hakansson@ericsson.com]=20
> Sent: 19 August 2011 13:34
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: Use cases - recording and voicemail
>=20
> John,
> >> I'm less sure about that the recorded media should also be an
> >> RTP source - couldn't you just as well send the file over and
> >> then play it at the remote end?
> >[JRE] Yes, the file could be streamed across, but there are=20
> folks who want it to get across more or less in real time,=20
> e.g., where the recorder is performing real-time analytics,=20
> perhaps in a >contact centre. This is the basis for the work=20
> done in the IETF SIPREC WG. The same considerations that=20
> require RTP browser-to-browser for RTC-Web also dictate RTP=20
> as the transport for >sending media to a recorder.
> [Stefan] If you want to do stuff remotely in real time, could=20
> you not just avoid recording locally? Regardless of if the=20
> analysis is to happen on a locally generated stream, or on a=20
> stream received via RTP, that stream can be streamed to the=20
> analysing server (via a p2p-connection), right? (Probably I=20
> am missing something).
>=20
> Stefan=

From john.elwell@siemens-enterprise.com  Fri Aug 19 06:29:57 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Fri, 19 Aug 2011 15:30:50 +0200
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Paul,

I agree file might not be the right term. What we need is a way of passing =
the stream, within the browser, between one session and a second session (i=
n SIPREC terms, between the communication session and the recording session=
).

John

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 19 August 2011 14:08
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>=20
> On 8/19/11 5:49 AM, Stefan H=E5kansson LK wrote:
> > John,
> >
> > this was a good way sort things up.
> >
> > I my view we should definitely support local recording of=20
> streams (regardless of if they are generated by local devices=20
> or received via RTP), and this could be done in parallel to=20
> rendering them or not (up to the app).
> >
> > The recorded media should also be possible to render=20
> locally (be the source for a video element).
> >
> > I'm less sure about that the recorded media should also be=20
> an RTP source - couldn't you just as well send the file over=20
> and then play it at the remote end?
>=20
> I think discussing this in terms of a "file" is an over=20
> simplification.
> While it could be a file, it could also be synthesized, or assembled=20
> from pieces, interactively as a result of received media or=20
> signaling.=20
> In such cases you can't just "send the file".
>=20
> Also, sending the file only works if the recipient can always=20
> deal with=20
> a file equally to receiving an RTP stream. That seems like a=20
> burden to=20
> levy on all possible recipients. For instance, it would mean that=20
> rtcweb:pstn gateways would all require the capability to buffer and=20
> playback audio/video files.
>=20
> So perhaps what is needed is the concept of a software agent that can=20
> take on the role of a local source/sink of media streams.
>=20
> 	Thanks,
> 	Paul
>=20
> > Stefan
> > ________________________________________
> > From: rtcweb-bounces@ietf.org [rtcweb-bounces@ietf.org] On=20
> Behalf Of Elwell, John [john.elwell@siemens-enterprise.com]
> > Sent: Friday, August 19, 2011 9:34 AM
> > To: rtcweb@ietf.org; public-webrtc@w3.org
> > Subject: [rtcweb] Use cases - recording and voicemail
> >
> > Sorry for lateness catching up on this thread.
> >
> > I did indeed propose recording use cases. I see that=20
> Randell has applied that to mail (voicemail / videomail) use=20
> cases. I would like to try to provide some separation here.
> >
> > The recording use cases I had in mind were cases where=20
> audio/video are captured/rendered locally (through=20
> camera/mic/display/speaker) and IN ADDITION are recorded,=20
> either locally or remotely. I tend to think of=20
> voicemail/videomail as being a replacement for local=20
> capture/rendering, or to put it another way, the storage=20
> device acts as the capture/rendering device. So in the=20
> voicemail/videomail case, recording is INSTEAD OF=20
> conventional capture/rendering.
> >
> > Voicemail/videomail can be local or remote. In the remote=20
> case, the inbound call is diverted to a remote mailbox=20
> device. This is a signalling plane matter, and outside the=20
> scope of RTC-Web, I believe.
> >
> > So the interesting case is local voicemail/videomail. It=20
> simply means that a mailboxes or files replace the=20
> conventional capture rendering devices. So received audio and=20
> video can be sent to files, and files can be used as the=20
> source of prompts or other information to be fed back to a=20
> caller. The basic requirement seems to be to use files as=20
> sources and sinks of media sent/received over RTP. One=20
> additional requirement is some means by which the caller can=20
> control the mailbox - this is conventionally done by DTMF or=20
> voice recognition - in fact these are the only methods=20
> available when a call comes from PSTN. So this could also=20
> lead to a requirement for receiving DTMF.
> >
> > Local recording, on the other hand, means that media=20
> captured from local devices and sent via RTP and media=20
> received via RTP and rendered at local devices are also=20
> copied to local files.
> >
> > Remote recording means that media captured from local=20
> devices and sent via RTP and media received via RTP and=20
> rendered at local devices are also sent via RTP to a remote=20
> recording device.
> >
> > Basically, all these use cases, if we choose to progress=20
> them, have a general impact on requirements - some sort of=20
> flexibility in terms of where media is sourced and sunk,=20
> including the ability to use a file as a source/sink and the=20
> ability to have multiple sinks. It seems that a well chosen=20
> API architecture could accommodate these, but leaving these=20
> requirements until later might mean that the chosen API=20
> architecture is not sufficiently flexible to accommodate=20
> these requirements later. DTMF reception, if we decide we=20
> need it, would be a separate requirement.
> >
> > So I would welcome feedback on which of these use cases are=20
> interesting (I have already heard some support for recording=20
> use cases), and I could propose text if necessary.
> >
> > John
> >
> > John Elwell
> > Tel: +44 1908 817801 (office and mobile)
> > Email: john.elwell@siemens-enterprise.com
> > http://www.siemens-enterprise.com/uk/
> >
> > Siemens Enterprise Communications Limited.
> > Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15 0DJ.
> > Registered No: 5903714, England.
> >
> > Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of Siemens AG.
> >
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org
> >> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> >> Sent: 04 August 2011 16:04
> >> To: rtcweb@ietf.org; public-webrtc@w3.org
> >> Subject: Re: [rtcweb] Use cases: summary of status
> >>
> >> On 8/4/2011 3:57 AM, Stefan H=E5kansson LK wrote:
> >>
> >>>
> >>> B. New use cases
> >>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>> I noted (as not immediately dismissed) the following proposals:
> >>>
> >>>       1. Distributed music band, discussed at the webrtc meeting
> >>>       2. Use cases regarding different situations when being
> >> invited to a "session", e.g. browser open, browser open but
> >> another tab active, browser open but active in session,
> >> browser closed, .. (Matthew Kaufman); discussed at webrtc meeting
> >>>       3. Different TURN provider scenarios (Cullen
> >> Jennings); discussed at the webrtc meeting
> >>>       4. More challenging NAT case (Matthew Kaufman); UDP
> >> blocked
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00468.html
> >>>       5. E911 (Paul Beaumont)
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00525.html
> >>>       6. Local Recording (John Ewell) at webrtc meeting
> >>
> >> Would this cover all "voicemail" and "videomail" cases?  I.e.
> >> having a client
> >> accept connections in the background if the call is not
> >> answered, optionally
> >> play a message, and record incoming audio/video, and allow
> >> the remote user to
> >> interact with it.  Also remote playback of messages.  If not
> >> (and if it's not
> >> covered by the current document, we need to add this usecase.
> >>
> >> Note that there are two variants: one local (local client
> >> handles it, which
> >> has more security issues around camera access and
> >> pre-approval), and one for
> >> remote recording (after some time with no answer or if not
> >> "registered" call
> >> is forwarded to a message server that answers it).  Note that
> >> there are
> >> security identity issues here (key chains, etc).
> >>
> >>
> >>>
> >>>       7. Remote recording (John)
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00472.html
> >>>       8. Emergency access for disabled (Bernard Aboba)
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00478.html
> >>>       9. Clue use cases (Roni Even)
> >>=20
> http://tools.ietf.org/html/draft-ietf-clue-telepresence-use-cases-01
> >>>       10. Rohan red cross (Cullen Jennings); proposed a bit
> >> earlier
> >> http://www.ietf.org/mail-archive/web/rtcweb/current/msg00323.html
> >>>
> >>> Probably I have missed a few.
> >>
> >> I'd add:
> >>
> >>
> >> 11. Remote assistance (ala VNC or RDP) - User is helping
> >> another user on
> >> their computer with either view-only or view-with-control,
> >> either of just
> >> the browser of the the entire screen.  Computer audio is
> >> optionally sent
> >> as well.  They are optionally talking and/or viewing each
> >> other while this
> >> is occurring.  They may be transferring files over a reliable data
> >> connection during this session.
> >>
> >>
> >> Commentary: How often have you talked to your
> >> father/mother/brother/etc
> >> and tried to debug their computer problems remotely?  And
> >> getting them to
> >> install a specific remote-assistance package, and to use it,
> >> and get it to
> >> work through firewalls, can be very painful.  (There are
> >> other uses of this
> >> ability to copilot as well, including classroom instruction,
> >> distance learning,
> >> etc.)  This use-case enables someone to build a JS app to=20
> provide this
> >> functionality.  (Note that for the W3 and browser vendors
> >> there will be
> >> significant security issues to resolve.)  I think this is
> >> actually a fairly
> >> important use-case.
> >>
> >>
> >> Additional resulting requirements:
> >>
> >> * Need to be able to select a "video" source other than a
> >> camera (window or
> >> screen) (W3 issue)
> >>
> >> * Need to be able to send multiple video and audio streams
> >> from different
> >> sources (probably already covered, mostly W3 issue)
> >>
> >> * Need to be able to use a reliable data connection (for
> >> mouse/keyboard/etc
> >> control, plus file transfer)
> >>
> >>
> >> 12. Security camera/baby monitor usage - use a persistent=20
> or temporary
> >> connection to provide basic security camera operation,
> >> including switching
> >> cameras or mics, or with the ability to remotely request
> >> review of recent
> >> data recorded.  Should be able to switch to 2-way audio
> >> and/or video remotely.
> >>
> >>
> >> Additional requirements:
> >>
> >> * Need to be able to select a "video" source other than a
> >> camera (file)
> >> (W3 issue)
> >>
> >> * Remote control of camera/mic selection and enabling (W3=20
> issue) - may
> >> require reliable data connection
> >>
> >> * May need control from JS over codecs used, at least voice
> >> vs audio, etc,
> >> and max video framerate/resolution/bandwidth (W3 issue largely?)
> >>
> >>
> >>>
> >>>
> >>> E. Opinion as individual
> >>> =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
> >>> My personal opinion is that we at this stage should focus
> >> on the basic use cases, and solve those in a good way. To me
> >> that would mean that we should add 3 (to allow different
> >> providers) and 4 (if you can't connect no use case can be
> >> supported). I think that these use cases can be added as
> >> twists of/extensions to the first use case (Simple Video
> >> Communication Service).
> >>>
> >>> In my view it also means that we should add text and
> >> requirements for 2 to make sure that the communication
> >> capabilites we are adding to the browser is usable in
> >> everyday scenarios. Presumable this would lead requirements
> >> (e.g. background execution capability) that are transferred
> >> to other W3C WGs - not to requirements in the webrtc/rtcweb space.
> >>>
> >>>>  From a W3C perspective it could also make sense to add a
> >> recording use case (as it seems that other W3C WGs rely on
> >> webrtc to provide an API for recording).
> >>>
> >>> The rest of the new proposed use cases, IMO, could be
> >> looked into in a second stage when we've established the basics.
> >>
> >> I'd very much want to include the "remote assistance" case,
> >> and I think the
> >> voicemail cases could be very important in overall acceptance
> >> and utility, especially
> >> given the issues over whether someone's machine is running
> >> and accepting calls.
> >>
> >> Security cam/baby-monitor is less important, but highlights
> >> some controls that
> >> we may want to expose to the JS over maximum
> >> bitrate/framerate/resolution/etc,
> >> and of course a slew of security issues for W3 to think about.
> >>
> >> --
> >> Randell Jesup
> >> randell-ietf@jesup.org
> >>
> >> _______________________________________________
> >> 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
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From stefan.lk.hakansson@ericsson.com  Fri Aug 19 07:59:23 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 16:55:40 +0200
Thread-Topic: Use cases - recording and voicemail
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>However, I did suggest (in other text in my previous message) that one pos=
sible solution might be to record locally and use a second RTC-Web session =
to transmit from the local file to the >remote recorder. What I failed to s=
ay was that in this case the local file would be a temporary repository - j=
ust a buffer between the two sessions.
This makes sense. Also, if you look at the API proposals available, it woul=
d be quite easy to forward (in real time) a stream being received to anothe=
r entity. There is no explicit recording, a stream being received (via RTP)=
 is just streamed to another entity (via a separate RTC-Web session). I thi=
nk this would solve this case.

Stefan=

From pravindran@sonusnet.com  Fri Aug 19 10:18:01 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "Elwell, John" <john.elwell@siemens-enterprise.com>, <rtcweb@ietf.org>, <public-webrtc@w3.org>
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Stefan,

In case recording similar to SIPREC, it is little bit more than spanning =
two media (RTP stream) alone because recording has to include some =
context data about recording apart from the media stream.

Thanks
Partha

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stefan H=E5kansson LK
> Sent: Friday, August 19, 2011 8:26 PM
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>=20
> >However, I did suggest (in other text in my previous message) that =
one
> possible solution might be to record locally and use a second RTC-Web
> session to transmit from the local file to the >remote recorder. What =
I
> failed to say was that in this case the local file would be a =
temporary
> repository - just a buffer between the two sessions.
> This makes sense. Also, if you look at the API proposals available, it
> would be quite easy to forward (in real time) a stream being received
> to another entity. There is no explicit recording, a stream being
> received (via RTP) is just streamed to another entity (via a separate
> RTC-Web session). I think this would solve this case.
>=20
> Stefan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From stefan.lk.hakansson@ericsson.com  Fri Aug 19 10:46:47 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Ravindran Parthasarathi <pravindran@sonusnet.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Fri, 19 Aug 2011 19:46:52 +0200
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Partha,

thanks for updating me.

Stefan
________________________________________
From: Ravindran Parthasarathi [pravindran@sonusnet.com]
Sent: Friday, August 19, 2011 7:18 PM
To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org; public-webrtc@w3.=
org
Subject: RE: [rtcweb] Use cases - recording and voicemail

Stefan,

In case recording similar to SIPREC, it is little bit more than spanning tw=
o media (RTP stream) alone because recording has to include some context da=
ta about recording apart from the media stream.

Thanks
Partha

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stefan H=E5kansson LK
> Sent: Friday, August 19, 2011 8:26 PM
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>
> >However, I did suggest (in other text in my previous message) that one
> possible solution might be to record locally and use a second RTC-Web
> session to transmit from the local file to the >remote recorder. What I
> failed to say was that in this case the local file would be a temporary
> repository - just a buffer between the two sessions.
> This makes sense. Also, if you look at the API proposals available, it
> would be quite easy to forward (in real time) a stream being received
> to another entity. There is no explicit recording, a stream being
> received (via RTP) is just streamed to another entity (via a separate
> RTC-Web session). I think this would solve this case.
>
> Stefan
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From randell-ietf@jesup.org  Fri Aug 19 14:52:54 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/19/2011 5:49 AM, Stefan Håkansson LK wrote:

> John,
>
> this was a good way sort things up.
>
> I my view we should definitely support local recording of streams (regardless of if they are generated by local devices or received via RTP), and this could be done in parallel to rendering them or not (up to the app).

Agreed.   Note that there are legal requirements in various locations around recording
conversations; that's up to the application IMHO -- however we'll want to make sure it's
reasonably easy for the application to do.  While I'm not an expert, recording someone
in many jurisdictions requires periodic beeps, etc.  They'd have to mix it into the outgoing
stream, but it would have to remain there even if the user "muted".  I want to make sure
we're providing something that won't be a hassle for the application developers.


>
> The recorded media should also be possible to render locally (be the source for a video element).
Yes.

>
> I'm less sure about that the recorded media should also be an RTP source - couldn't you just as well send the file over and then play it at the remote end?

That might not work for cases where the two users are talking through different
providers/apps, and it also would imply a much longer delay in many cases, plus
local storage requirements, etc.   Think a recorded greeting played to callers if
no one answers, for example.

Assuming that recording and playback are of encoded media:

There are issues with recording and playback having to do with error recovery.
For recording an incoming stream, it's less of an issue - you do normal error
recovery, and on playback it would look the same as it would have if the call had
been live.

For sending a pre-recorded stream (greeting, etc): I'd assume it was recorded
without loss.  However, the other side may experience loss in receiving it.
To deal with this, we can a) decode and re-encode the media, allowing us to react
to incoming loss reports, or b) include periodic IDRs or equivalent.  I would
lean to a) (decode&  re-encode), that also handles issues with codec parameters,
codec choice, etc.  So an input would be from a decoded stream.  This may use more
resources than b; the application could (at its discretion) not render incoming media
while playing back.


-- 
Randell Jesup
randell-ietf@jesup.org


From pkyzivat@alum.mit.edu  Fri Aug 19 16:51:12 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Randell,

Many of the issues you are bringing up are things we have been 
addressing in the siprec WG. RTCWEB can either reinvent all that, or we 
can find some way to reuse that work. At the moment, siprec assumes that 
the agent that is initiating the recording is in the signaling path of a 
sip call, and that it is capable of establishing another sip session to 
the recorder. Those assumptions would seem to be wrong for RTCWEB.

	Thanks,
	Paul

On 8/19/11 5:51 PM, Randell Jesup wrote:
> On 8/19/2011 5:49 AM, Stefan Håkansson LK wrote:
>
>> John,
>>
>> this was a good way sort things up.
>>
>> I my view we should definitely support local recording of streams
>> (regardless of if they are generated by local devices or received via
>> RTP), and this could be done in parallel to rendering them or not (up
>> to the app).
>
> Agreed. Note that there are legal requirements in various locations
> around recording
> conversations; that's up to the application IMHO -- however we'll want
> to make sure it's
> reasonably easy for the application to do. While I'm not an expert,
> recording someone
> in many jurisdictions requires periodic beeps, etc. They'd have to mix
> it into the outgoing
> stream, but it would have to remain there even if the user "muted". I
> want to make sure
> we're providing something that won't be a hassle for the application
> developers.
>
>
>>
>> The recorded media should also be possible to render locally (be the
>> source for a video element).
> Yes.
>
>>
>> I'm less sure about that the recorded media should also be an RTP
>> source - couldn't you just as well send the file over and then play it
>> at the remote end?
>
> That might not work for cases where the two users are talking through
> different
> providers/apps, and it also would imply a much longer delay in many
> cases, plus
> local storage requirements, etc. Think a recorded greeting played to
> callers if
> no one answers, for example.
>
> Assuming that recording and playback are of encoded media:
>
> There are issues with recording and playback having to do with error
> recovery.
> For recording an incoming stream, it's less of an issue - you do normal
> error
> recovery, and on playback it would look the same as it would have if the
> call had
> been live.
>
> For sending a pre-recorded stream (greeting, etc): I'd assume it was
> recorded
> without loss. However, the other side may experience loss in receiving it.
> To deal with this, we can a) decode and re-encode the media, allowing us
> to react
> to incoming loss reports, or b) include periodic IDRs or equivalent. I
> would
> lean to a) (decode& re-encode), that also handles issues with codec
> parameters,
> codec choice, etc. So an input would be from a decoded stream. This may
> use more
> resources than b; the application could (at its discretion) not render
> incoming media
> while playing back.
>
>


From harald@alvestrand.no  Sat Aug 20 08:48:35 2011
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On 08/20/11 01:52, Paul Kyzivat wrote:
> Randell,
>
> Many of the issues you are bringing up are things we have been 
> addressing in the siprec WG. RTCWEB can either reinvent all that, or 
> we can find some way to reuse that work. At the moment, siprec assumes 
> that the agent that is initiating the recording is in the signaling 
> path of a sip call, and that it is capable of establishing another sip 
> session to the recorder. Those assumptions would seem to be wrong for 
> RTCWEB.
Hm.

RTCWEB doesn't seem to going in the direction of mandating SIP, but 
still, I would think that it is reasonable to say something along the 
lines of "the remote recording case is handled by connecting to a 
SIPREC-capable recorder" (with the usual degree of gatewaying help from 
our signaling proxies).

That will, of course, require that the SIPREC recorder is capable of 
participiating in an RTCWEB session (that is, support ICE and the 
mandatory codecs), that the RTCWEB implementation be capable of copying 
incoming media streams to an outgoing interface, and that negotiation 
can down-negotiate to something that is supported by both call 
participants and the recording device. Does SIPREC envision establishing 
minimum requirements for codecs and profiles?

Once we can satisfy ourselves that we have all the pieces required to 
send media off to a remote API, we should see if we can do something 
very similar for sending media off to some kind of local recorder; it 
seems less likely that we'll get into trouble with locking ourselves 
into a wrong model if we do things in that order.

My $0.02.

               Harald



From john.elwell@siemens-enterprise.com  Sun Aug 21 23:27:28 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Ravindran Parthasarathi <pravindran@sonusnet.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
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Partha,

You are talking here about the metadata, I think. I assume the web page / J=
avaScript has to deal with that - not the browser.

John
=20

> -----Original Message-----
> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]=20
> Sent: 19 August 2011 18:19
> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;=20
> public-webrtc@w3.org
> Subject: RE: [rtcweb] Use cases - recording and voicemail
>=20
> Stefan,
>=20
> In case recording similar to SIPREC, it is little bit more=20
> than spanning two media (RTP stream) alone because recording=20
> has to include some context data about recording apart from=20
> the media stream.
>=20
> Thanks
> Partha
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Stefan H=E5kansson LK
> > Sent: Friday, August 19, 2011 8:26 PM
> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> > Subject: Re: [rtcweb] Use cases - recording and voicemail
> >=20
> > >However, I did suggest (in other text in my previous=20
> message) that one
> > possible solution might be to record locally and use a=20
> second RTC-Web
> > session to transmit from the local file to the >remote=20
> recorder. What I
> > failed to say was that in this case the local file would be=20
> a temporary
> > repository - just a buffer between the two sessions.
> > This makes sense. Also, if you look at the API proposals=20
> available, it
> > would be quite easy to forward (in real time) a stream=20
> being received
> > to another entity. There is no explicit recording, a stream being
> > received (via RTP) is just streamed to another entity (via=20
> a separate
> > RTC-Web session). I think this would solve this case.
> >=20
> > Stefan
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> =

From pravindran@sonusnet.com  Sun Aug 21 23:41:30 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  <rtcweb@ietf.org>, <public-webrtc@w3.org>
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John,

I agree with you. JavaScript API should have the provision to pass the =
metadata.

Thanks
Partha

>-----Original Message-----
>From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>Sent: Monday, August 22, 2011 11:59 AM
>To: Ravindran Parthasarathi; Stefan H=E5kansson LK; rtcweb@ietf.org;
>public-webrtc@w3.org
>Subject: RE: [rtcweb] Use cases - recording and voicemail
>
>Partha,
>
>You are talking here about the metadata, I think. I assume the web page
>/ JavaScript has to deal with that - not the browser.
>
>John
>
>
>> -----Original Message-----
>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>> Sent: 19 August 2011 18:19
>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>> public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Stefan,
>>
>> In case recording similar to SIPREC, it is little bit more
>> than spanning two media (RTP stream) alone because recording
>> has to include some context data about recording apart from
>> the media stream.
>>
>> Thanks
>> Partha
>>
>> > -----Original Message-----
>> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> > Behalf Of Stefan H=E5kansson LK
>> > Sent: Friday, August 19, 2011 8:26 PM
>> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>> > Subject: Re: [rtcweb] Use cases - recording and voicemail
>> >
>> > >However, I did suggest (in other text in my previous
>> message) that one
>> > possible solution might be to record locally and use a
>> second RTC-Web
>> > session to transmit from the local file to the >remote
>> recorder. What I
>> > failed to say was that in this case the local file would be
>> a temporary
>> > repository - just a buffer between the two sessions.
>> > This makes sense. Also, if you look at the API proposals
>> available, it
>> > would be quite easy to forward (in real time) a stream
>> being received
>> > to another entity. There is no explicit recording, a stream being
>> > received (via RTP) is just streamed to another entity (via
>> a separate
>> > RTC-Web session). I think this would solve this case.
>> >
>> > Stefan
>> > _______________________________________________
>> > rtcweb mailing list
>> > rtcweb@ietf.org
>> > https://www.ietf.org/mailman/listinfo/rtcweb
>>

From Ranjit.Avasarala@Polycom.com  Sun Aug 21 23:51:11 2011
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: Ravindran Parthasarathi <pravindran@sonusnet.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Mon, 22 Aug 2011 14:52:09 +0800
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Message-ID: <1F2A2C70609D9E41844A2126145FC09801F1550D@HKGMBOXPRD22.polycom.com>
References: <BBF498F2D030E84AB1179E24D1AC41D616C389F16D@ESESSCMS0362.eemea.ericsson.se><4E3AB4D4.4070308@jesup.org>, <A444A0F8084434499206E78C106220CA09BDB6A238@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F242@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A34F@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F245@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A397@MCHP058A.global-ad.net> <BBF498F2D030E84AB1179E24D1AC41D616C389F24A@ESESSCMS0362.eemea.ericsson.se> <2E239D6FCD033C4BAF15F386A979BF510640D0@sonusinmail02.sonusnet.com> <A444A0F8084434499206E78C106220CA0B00FDABAD@MCHP058A.global-ad.net> <2E239D6FCD033C4BAF15F386A979BF51064106@sonusinmail02.sonusnet.com>
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Hi

We could use websockets protocol to pass metadata information.=20

Regards
Ranjit

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Ravindran Parthasarathi
Sent: Monday, August 22, 2011 12:12 PM
To: Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.org; public-webrtc@w3.=
org
Subject: Re: [rtcweb] Use cases - recording and voicemail

John,

I agree with you. JavaScript API should have the provision to pass the meta=
data.

Thanks
Partha

>-----Original Message-----
>From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>Sent: Monday, August 22, 2011 11:59 AM
>To: Ravindran Parthasarathi; Stefan H=E5kansson LK; rtcweb@ietf.org;
>public-webrtc@w3.org
>Subject: RE: [rtcweb] Use cases - recording and voicemail
>
>Partha,
>
>You are talking here about the metadata, I think. I assume the web page
>/ JavaScript has to deal with that - not the browser.
>
>John
>
>
>> -----Original Message-----
>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>> Sent: 19 August 2011 18:19
>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>> public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Stefan,
>>
>> In case recording similar to SIPREC, it is little bit more
>> than spanning two media (RTP stream) alone because recording
>> has to include some context data about recording apart from
>> the media stream.
>>
>> Thanks
>> Partha
>>
>> > -----Original Message-----
>> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> > Behalf Of Stefan H=E5kansson LK
>> > Sent: Friday, August 19, 2011 8:26 PM
>> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>> > Subject: Re: [rtcweb] Use cases - recording and voicemail
>> >
>> > >However, I did suggest (in other text in my previous
>> message) that one
>> > possible solution might be to record locally and use a
>> second RTC-Web
>> > session to transmit from the local file to the >remote
>> recorder. What I
>> > failed to say was that in this case the local file would be
>> a temporary
>> > repository - just a buffer between the two sessions.
>> > This makes sense. Also, if you look at the API proposals
>> available, it
>> > would be quite easy to forward (in real time) a stream
>> being received
>> > to another entity. There is no explicit recording, a stream being
>> > received (via RTP) is just streamed to another entity (via
>> a separate
>> > RTC-Web session). I think this would solve this case.
>> >
>> > Stefan
>> > _______________________________________________
>> > 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  Mon Aug 22 01:15:33 2011
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References: <BBF498F2D030E84AB1179E24D1AC41D616C389F16D@ESESSCMS0362.eemea.ericsson.se><4E3AB4D4.4070308@jesup.org>, <A444A0F8084434499206E78C106220CA09BDB6A238@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F242@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A34F@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F245@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A397@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F24A@ESESSCMS0362.eemea.ericsson.se><2E239D6FCD033C4BAF15F386A979BF510640D0@sonusinmail02.sonusnet.com><A444A0F8084434499206E78C106220CA0B00FDABAD@MCHP058A.global-ad.net> <2E239D6FCD033C4BAF15F386A979BF51064106@sonusinmail02.sonusnet.com> <1F2A2C70609D9E41844A2126145FC09801F1550D@HKGMBOXPRD22.polycom.com>
From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  <rtcweb@ietf.org>, <public-webrtc@w3.org>
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Hi Ranjit,

I have mentioned as JavaScript API because there is no specific =
dependency on browser. Let us discuss about the exact mechanism later. =
Currently, let us have common understanding whether recording usecase =
has to be added in RTCWeb or not.

Thanks
Partha=20

>-----Original Message-----
>From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>Sent: Monday, August 22, 2011 12:22 PM
>To: Ravindran Parthasarathi; Elwell, John; Stefan H=E5kansson LK;
>rtcweb@ietf.org; public-webrtc@w3.org
>Subject: RE: [rtcweb] Use cases - recording and voicemail
>
>Hi
>
>We could use websockets protocol to pass metadata information.
>
>Regards
>Ranjit
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>Of Ravindran Parthasarathi
>Sent: Monday, August 22, 2011 12:12 PM
>To: Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.org; public-
>webrtc@w3.org
>Subject: Re: [rtcweb] Use cases - recording and voicemail
>
>John,
>
>I agree with you. JavaScript API should have the provision to pass the
>metadata.
>
>Thanks
>Partha
>
>>-----Original Message-----
>>From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>Sent: Monday, August 22, 2011 11:59 AM
>>To: Ravindran Parthasarathi; Stefan H=E5kansson LK; rtcweb@ietf.org;
>>public-webrtc@w3.org
>>Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>>Partha,
>>
>>You are talking here about the metadata, I think. I assume the web =
page
>>/ JavaScript has to deal with that - not the browser.
>>
>>John
>>
>>
>>> -----Original Message-----
>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>> Sent: 19 August 2011 18:19
>>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Stefan,
>>>
>>> In case recording similar to SIPREC, it is little bit more
>>> than spanning two media (RTP stream) alone because recording
>>> has to include some context data about recording apart from
>>> the media stream.
>>>
>>> Thanks
>>> Partha
>>>
>>> > -----Original Message-----
>>> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> > Behalf Of Stefan H=E5kansson LK
>>> > Sent: Friday, August 19, 2011 8:26 PM
>>> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>> > Subject: Re: [rtcweb] Use cases - recording and voicemail
>>> >
>>> > >However, I did suggest (in other text in my previous
>>> message) that one
>>> > possible solution might be to record locally and use a
>>> second RTC-Web
>>> > session to transmit from the local file to the >remote
>>> recorder. What I
>>> > failed to say was that in this case the local file would be
>>> a temporary
>>> > repository - just a buffer between the two sessions.
>>> > This makes sense. Also, if you look at the API proposals
>>> available, it
>>> > would be quite easy to forward (in real time) a stream
>>> being received
>>> > to another entity. There is no explicit recording, a stream being
>>> > received (via RTP) is just streamed to another entity (via
>>> a separate
>>> > RTC-Web session). I think this would solve this case.
>>> >
>>> > Stefan
>>> > _______________________________________________
>>> > 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  Mon Aug 22 01:31:02 2011
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To: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Mon, 22 Aug 2011 10:32:04 +0200
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> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Stefan H=E5kansson LK
> Sent: 19 August 2011 15:56
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>=20
> >However, I did suggest (in other text in my previous message) that one
> possible solution might be to record locally and use a second RTC-Web
> session to transmit from the local file to the >remote recorder. What I
> failed to say was that in this case the local file would be a temporary
> repository - just a buffer between the two sessions.
> This makes sense. Also, if you look at the API proposals available, it
> would be quite easy to forward (in real time) a stream being received
> to another entity. There is no explicit recording, a stream being
> received (via RTP) is just streamed to another entity (via a separate
> RTC-Web session). I think this would solve this case.

[AndyH] It might do but I assume this would limit the RTP model that could =
be used to the endpoint model. We are currently reviewing how the RTP model=
 for a session recording client should look like with the AVT experts (See =
http://tools.ietf.org/html/draft-eckel-siprec-rtp-rec-01). This might resul=
t in some requirements on how the client handles RTCP for example.


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

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To: Harald Alvestrand <harald@alvestrand.no>
Date: Mon, 22 Aug 2011 11:25:40 +0200
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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>=20
> RTCWEB doesn't seem to going in the direction of mandating SIP, but
> still, I would think that it is reasonable to say something along the
> lines of "the remote recording case is handled by connecting to a
> SIPREC-capable recorder" (with the usual degree of gatewaying help from
> our signaling proxies).
>=20
[AndyH] - I agree this would just seem to be another example of how the Ses=
sion Recording Client can be decomposed and is similar to Figure 4 in the S=
IPREC architecture draft (http://tools.ietf.org/html/draft-ietf-siprec-arch=
itecture-02#section-3.1.2) but in the case of RTCWEB the media server is th=
e browser and the application is the web server and of course RTCWEB will n=
ot use mediactrl. I have been resisting attempts to add more diagrams to th=
e SIPREC architecture draft to show the multitude of ways the SRC and SRS c=
ould be decomposed but maybe RTCWEB is a good reason to do this.


> That will, of course, require that the SIPREC recorder is capable of
> participiating in an RTCWEB session (that is, support ICE and the
> mandatory codecs), that the RTCWEB implementation be capable of copying
> incoming media streams to an outgoing interface, and that negotiation
> can down-negotiate to something that is supported by both call
> participants and the recording device. Does SIPREC envision
> establishing
> minimum requirements for codec's and profiles?

[AndyH] - No the SIPREC requirements (http://tools.ietf.org/html/rfc6341) d=
on't include minimum codec requirements.

>=20
> Once we can satisfy ourselves that we have all the pieces required to
> send media off to a remote API, we should see if we can do something
> very similar for sending media off to some kind of local recorder; it
> seems less likely that we'll get into trouble with locking ourselves
> into a wrong model if we do things in that order.
>=20
> My $0.02.
>=20
>                Harald
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

From Ranjit.Avasarala@Polycom.com  Mon Aug 22 03:16:23 2011
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: Ravindran Parthasarathi <pravindran@sonusnet.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Mon, 22 Aug 2011 18:17:22 +0800
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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Hi Partha

I agree with Paul that we could use the recording mechanism discussed in si=
prec mailing list could be added here as webbrowser would essentially be a =
SIP UA which could initiate and terminate SIP sessions.=20

Regards
Ranjit


-----Original Message-----
From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]=20
Sent: Monday, August 22, 2011 1:46 PM
To: Avasarala, Ranjit; Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.org=
; public-webrtc@w3.org
Subject: RE: [rtcweb] Use cases - recording and voicemail

Hi Ranjit,

I have mentioned as JavaScript API because there is no specific dependency =
on browser. Let us discuss about the exact mechanism later. Currently, let =
us have common understanding whether recording usecase has to be added in R=
TCWeb or not.

Thanks
Partha=20

>-----Original Message-----
>From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>Sent: Monday, August 22, 2011 12:22 PM
>To: Ravindran Parthasarathi; Elwell, John; Stefan H=E5kansson LK;
>rtcweb@ietf.org; public-webrtc@w3.org
>Subject: RE: [rtcweb] Use cases - recording and voicemail
>
>Hi
>
>We could use websockets protocol to pass metadata information.
>
>Regards
>Ranjit
>
>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>Of Ravindran Parthasarathi
>Sent: Monday, August 22, 2011 12:12 PM
>To: Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.org; public-
>webrtc@w3.org
>Subject: Re: [rtcweb] Use cases - recording and voicemail
>
>John,
>
>I agree with you. JavaScript API should have the provision to pass the
>metadata.
>
>Thanks
>Partha
>
>>-----Original Message-----
>>From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>Sent: Monday, August 22, 2011 11:59 AM
>>To: Ravindran Parthasarathi; Stefan H=E5kansson LK; rtcweb@ietf.org;
>>public-webrtc@w3.org
>>Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>>Partha,
>>
>>You are talking here about the metadata, I think. I assume the web page
>>/ JavaScript has to deal with that - not the browser.
>>
>>John
>>
>>
>>> -----Original Message-----
>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>> Sent: 19 August 2011 18:19
>>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Stefan,
>>>
>>> In case recording similar to SIPREC, it is little bit more
>>> than spanning two media (RTP stream) alone because recording
>>> has to include some context data about recording apart from
>>> the media stream.
>>>
>>> Thanks
>>> Partha
>>>
>>> > -----Original Message-----
>>> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> > Behalf Of Stefan H=E5kansson LK
>>> > Sent: Friday, August 19, 2011 8:26 PM
>>> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>> > Subject: Re: [rtcweb] Use cases - recording and voicemail
>>> >
>>> > >However, I did suggest (in other text in my previous
>>> message) that one
>>> > possible solution might be to record locally and use a
>>> second RTC-Web
>>> > session to transmit from the local file to the >remote
>>> recorder. What I
>>> > failed to say was that in this case the local file would be
>>> a temporary
>>> > repository - just a buffer between the two sessions.
>>> > This makes sense. Also, if you look at the API proposals
>>> available, it
>>> > would be quite easy to forward (in real time) a stream
>>> being received
>>> > to another entity. There is no explicit recording, a stream being
>>> > received (via RTP) is just streamed to another entity (via
>>> a separate
>>> > RTC-Web session). I think this would solve this case.
>>> >
>>> > Stefan
>>> > _______________________________________________
>>> > 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 harald@alvestrand.no  Mon Aug 22 04:09:17 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 08/22/11 12:17, Avasarala, Ranjit wrote:
> Hi Partha
>
> I agree with Paul that we could use the recording mechanism discussed in siprec mailing list could be added here as webbrowser would essentially be a SIP UA which could initiate and terminate SIP sessions.
mandatory standard clarification (?):

.. as somewhere in the combination of the Web browser, the Javascript 
and the Web server it connects to, there would be something that 
performs the functions of a SIP UA which could initiate and terminate 
SIP sessions ...
> Regards
> Ranjit
>
>
> -----Original Message-----
> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
> Sent: Monday, August 22, 2011 1:46 PM
> To: Avasarala, Ranjit; Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: [rtcweb] Use cases - recording and voicemail
>
> Hi Ranjit,
>
> I have mentioned as JavaScript API because there is no specific dependency on browser. Let us discuss about the exact mechanism later. Currently, let us have common understanding whether recording usecase has to be added in RTCWeb or not.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>> Sent: Monday, August 22, 2011 12:22 PM
>> To: Ravindran Parthasarathi; Elwell, John; Stefan Håkansson LK;
>> rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Hi
>>
>> We could use websockets protocol to pass metadata information.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Ravindran Parthasarathi
>> Sent: Monday, August 22, 2011 12:12 PM
>> To: Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-
>> webrtc@w3.org
>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>
>> John,
>>
>> I agree with you. JavaScript API should have the provision to pass the
>> metadata.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Monday, August 22, 2011 11:59 AM
>>> To: Ravindran Parthasarathi; Stefan Håkansson LK; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Partha,
>>>
>>> You are talking here about the metadata, I think. I assume the web page
>>> / JavaScript has to deal with that - not the browser.
>>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>> Sent: 19 August 2011 18:19
>>>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Stefan,
>>>>
>>>> In case recording similar to SIPREC, it is little bit more
>>>> than spanning two media (RTP stream) alone because recording
>>>> has to include some context data about recording apart from
>>>> the media stream.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Stefan Håkansson LK
>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>>> However, I did suggest (in other text in my previous
>>>> message) that one
>>>>> possible solution might be to record locally and use a
>>>> second RTC-Web
>>>>> session to transmit from the local file to the>remote
>>>> recorder. What I
>>>>> failed to say was that in this case the local file would be
>>>> a temporary
>>>>> repository - just a buffer between the two sessions.
>>>>> This makes sense. Also, if you look at the API proposals
>>>> available, it
>>>>> would be quite easy to forward (in real time) a stream
>>>> being received
>>>>> to another entity. There is no explicit recording, a stream being
>>>>> received (via RTP) is just streamed to another entity (via
>>>> a separate
>>>>> RTC-Web session). I think this would solve this case.
>>>>>
>>>>> Stefan
>>>>> _______________________________________________
>>>>> 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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To: Ravindran Parthasarathi <pravindran@sonusnet.com>
References: <BBF498F2D030E84AB1179E24D1AC41D616C389F16D@ESESSCMS0362.eemea.ericsson.se><4E3AB4D4.4070308@jesup.org>, <A444A0F8084434499206E78C106220CA09BDB6A238@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F242@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A34F@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F245@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A397@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F24A@ESESSCMS0362.eemea.ericsson.se><2E239D6FCD033C4BAF15F386A979BF510640D0@sonusinmail02.sonusnet.com><A444A0F8084434499206E78C106220CA0B00FDABAD@MCHP058A.global-ad.net> <2E239D6FCD033C4BAF15F386A979BF51064106@sonusinmail02.sonusnet.com> <1F2A2C70609D9E41844A2126145FC09801F1550D@HKGMBOXPRD22.polycom.com> <2E239D6FCD033C4BAF15F386A979BF51064117@sonusinmail02.sonusnet.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 2011-08-22 10:16, Ravindran Parthasarathi wrote:

> ... Currently, let us have common understanding whether recording usecase has to be added in RTCWeb or not.
Agree. And also _which_ recording use case(s) in that case. I think this 
is what John was looking for when starting the thread.

My $0.02 says that we need to take some care before adding more and more 
usages and reqs - after all the schedules for the WGs are quite aggressive.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>> Sent: Monday, August 22, 2011 12:22 PM
>> To: Ravindran Parthasarathi; Elwell, John; Stefan Håkansson LK;
>> rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Hi
>>
>> We could use websockets protocol to pass metadata information.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Ravindran Parthasarathi
>> Sent: Monday, August 22, 2011 12:12 PM
>> To: Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-
>> webrtc@w3.org
>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>
>> John,
>>
>> I agree with you. JavaScript API should have the provision to pass the
>> metadata.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Monday, August 22, 2011 11:59 AM
>>> To: Ravindran Parthasarathi; Stefan Håkansson LK; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Partha,
>>>
>>> You are talking here about the metadata, I think. I assume the web page
>>> / JavaScript has to deal with that - not the browser.
>>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>> Sent: 19 August 2011 18:19
>>>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Stefan,
>>>>
>>>> In case recording similar to SIPREC, it is little bit more
>>>> than spanning two media (RTP stream) alone because recording
>>>> has to include some context data about recording apart from
>>>> the media stream.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Stefan Håkansson LK
>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>>> However, I did suggest (in other text in my previous
>>>> message) that one
>>>>> possible solution might be to record locally and use a
>>>> second RTC-Web
>>>>> session to transmit from the local file to the>remote
>>>> recorder. What I
>>>>> failed to say was that in this case the local file would be
>>>> a temporary
>>>>> repository - just a buffer between the two sessions.
>>>>> This makes sense. Also, if you look at the API proposals
>>>> available, it
>>>>> would be quite easy to forward (in real time) a stream
>>>> being received
>>>>> to another entity. There is no explicit recording, a stream being
>>>>> received (via RTP) is just streamed to another entity (via
>>>> a separate
>>>>> RTC-Web session). I think this would solve this case.
>>>>>
>>>>> Stefan
>>>>> _______________________________________________
>>>>> 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 Ranjit.Avasarala@Polycom.com  Mon Aug 22 04:23:12 2011
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From: "Avasarala, Ranjit" <Ranjit.Avasarala@Polycom.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 22 Aug 2011 19:24:11 +0800
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References: <BBF498F2D030E84AB1179E24D1AC41D616C389F16D@ESESSCMS0362.eemea.ericsson.se><4E3AB4D4.4070308@jesup.org>, <A444A0F8084434499206E78C106220CA09BDB6A238@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F242@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A34F@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F245@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A397@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F24A@ESESSCMS0362.eemea.ericsson.se><2E239D6FCD033C4BAF15F386A979BF510640D0@sonusinmail02.sonusnet.com><A444A0F8084434499206E78C106220CA0B00FDABAD@MCHP058A.global-ad.net> <2E239D6FCD033C4BAF15F386A979BF51064106@sonusinmail02.sonusnet.com> <1F2A2C70609D9E41844A2126145FC09801F1550D@HKGMBOXPRD22.polycom.com> <2E239D6FCD033C4BAF15F386A979BF51064117@sonusinmail02.sonusnet.com> <1F2A2C70609D9E41844A2126145FC09801F1555B@HKGMBOXPRD22.polycom.com> <4E52391B.6000900@alvestrand.no>
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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True. One possible model could be integrating SIP stack with webbrowser and=
 send / receive SIP messages over websockets.=20

Regards
Ranjit


-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Harald Alvestrand
Sent: Monday, August 22, 2011 4:40 PM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Use cases - recording and voicemail

On 08/22/11 12:17, Avasarala, Ranjit wrote:
> Hi Partha
>
> I agree with Paul that we could use the recording mechanism discussed in =
siprec mailing list could be added here as webbrowser would essentially be =
a SIP UA which could initiate and terminate SIP sessions.
mandatory standard clarification (?):

.. as somewhere in the combination of the Web browser, the Javascript=20
and the Web server it connects to, there would be something that=20
performs the functions of a SIP UA which could initiate and terminate=20
SIP sessions ...
> Regards
> Ranjit
>
>
> -----Original Message-----
> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
> Sent: Monday, August 22, 2011 1:46 PM
> To: Avasarala, Ranjit; Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.o=
rg; public-webrtc@w3.org
> Subject: RE: [rtcweb] Use cases - recording and voicemail
>
> Hi Ranjit,
>
> I have mentioned as JavaScript API because there is no specific dependenc=
y on browser. Let us discuss about the exact mechanism later. Currently, le=
t us have common understanding whether recording usecase has to be added in=
 RTCWeb or not.
>
> Thanks
> Partha
>
>> -----Original Message-----
>> From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>> Sent: Monday, August 22, 2011 12:22 PM
>> To: Ravindran Parthasarathi; Elwell, John; Stefan H=E5kansson LK;
>> rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Hi
>>
>> We could use websockets protocol to pass metadata information.
>>
>> Regards
>> Ranjit
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>> Of Ravindran Parthasarathi
>> Sent: Monday, August 22, 2011 12:12 PM
>> To: Elwell, John; Stefan H=E5kansson LK; rtcweb@ietf.org; public-
>> webrtc@w3.org
>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>
>> John,
>>
>> I agree with you. JavaScript API should have the provision to pass the
>> metadata.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>> Sent: Monday, August 22, 2011 11:59 AM
>>> To: Ravindran Parthasarathi; Stefan H=E5kansson LK; rtcweb@ietf.org;
>>> public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Partha,
>>>
>>> You are talking here about the metadata, I think. I assume the web page
>>> / JavaScript has to deal with that - not the browser.
>>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>> Sent: 19 August 2011 18:19
>>>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Stefan,
>>>>
>>>> In case recording similar to SIPREC, it is little bit more
>>>> than spanning two media (RTP stream) alone because recording
>>>> has to include some context data about recording apart from
>>>> the media stream.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Stefan H=E5kansson LK
>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>>> However, I did suggest (in other text in my previous
>>>> message) that one
>>>>> possible solution might be to record locally and use a
>>>> second RTC-Web
>>>>> session to transmit from the local file to the>remote
>>>> recorder. What I
>>>>> failed to say was that in this case the local file would be
>>>> a temporary
>>>>> repository - just a buffer between the two sessions.
>>>>> This makes sense. Also, if you look at the API proposals
>>>> available, it
>>>>> would be quite easy to forward (in real time) a stream
>>>> being received
>>>>> to another entity. There is no explicit recording, a stream being
>>>>> received (via RTP) is just streamed to another entity (via
>>>> a separate
>>>>> RTC-Web session). I think this would solve this case.
>>>>>
>>>>> Stefan
>>>>> _______________________________________________
>>>>> 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 pkyzivat@alum.mit.edu  Mon Aug 22 06:12:07 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/22/11 7:10 AM, Harald Alvestrand wrote:
> On 08/22/11 12:17, Avasarala, Ranjit wrote:
>> Hi Partha
>>
>> I agree with Paul that we could use the recording mechanism discussed
>> in siprec mailing list could be added here as webbrowser would
>> essentially be a SIP UA which could initiate and terminate SIP sessions.
> mandatory standard clarification (?):
>
> .. as somewhere in the combination of the Web browser, the Javascript
> and the Web server it connects to, there would be something that
> performs the functions of a SIP UA which could initiate and terminate
> SIP sessions ...

Yes, *something* of this sort - the devil is in the details.

IIUC, its the intent of RTCWEB that the media flow directly between 
source and recipient, rather than following the path of the "signaling".

(That of course was also the intent of SIP with proxies, but the almost 
pervasive use of SBCs has usually forced media to follow the signaling 
path.)

In siprec, it is assumed that an entity called the SRC is in the 
signaling and media path of the call being recorded, as well as in the 
signaling and media path of a session with the recorder.

If we assume that the RTCWEB "client" is in the media path, but not in a 
sip signaling path, and that the RTCWEB "server" is sip-capable but not 
in the media path, then we have a structure that doesn't quite fit the 
siprec model.

However this could perhaps be made to work with siprec via some 
additional functionality in the RTCWEB client and server, and maybe some 
tweaks to what is being specified for siprec. The details will depend on 
which end is making the decision to do recording. If its the client, 
then that client will need a way to ask "the" server to establish a 
siprec session and to pass metadata. If its the server that makes the 
decision, then there needs to be a way for the server to ask the client 
to fork media and send it to multiple destinations.

There are of course privacy concerns in this. Whether they are different 
in RTCWEB than they are for siprec in general remains to be seen.

	Thanks,
	Paul

>> Regards
>> Ranjit
>>
>>
>> -----Original Message-----
>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>> Sent: Monday, August 22, 2011 1:46 PM
>> To: Avasarala, Ranjit; Elwell, John; Stefan Håkansson LK;
>> rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Hi Ranjit,
>>
>> I have mentioned as JavaScript API because there is no specific
>> dependency on browser. Let us discuss about the exact mechanism later.
>> Currently, let us have common understanding whether recording usecase
>> has to be added in RTCWeb or not.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: Avasarala, Ranjit [mailto:Ranjit.Avasarala@Polycom.com]
>>> Sent: Monday, August 22, 2011 12:22 PM
>>> To: Ravindran Parthasarathi; Elwell, John; Stefan Håkansson LK;
>>> rtcweb@ietf.org; public-webrtc@w3.org
>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>
>>> Hi
>>>
>>> We could use websockets protocol to pass metadata information.
>>>
>>> Regards
>>> Ranjit
>>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
>>> Of Ravindran Parthasarathi
>>> Sent: Monday, August 22, 2011 12:12 PM
>>> To: Elwell, John; Stefan Håkansson LK; rtcweb@ietf.org; public-
>>> webrtc@w3.org
>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>
>>> John,
>>>
>>> I agree with you. JavaScript API should have the provision to pass the
>>> metadata.
>>>
>>> Thanks
>>> Partha
>>>
>>>> -----Original Message-----
>>>> From: Elwell, John [mailto:john.elwell@siemens-enterprise.com]
>>>> Sent: Monday, August 22, 2011 11:59 AM
>>>> To: Ravindran Parthasarathi; Stefan Håkansson LK; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Partha,
>>>>
>>>> You are talking here about the metadata, I think. I assume the web page
>>>> / JavaScript has to deal with that - not the browser.
>>>>
>>>> John
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>>> Sent: 19 August 2011 18:19
>>>>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>>>>> public-webrtc@w3.org
>>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>> Stefan,
>>>>>
>>>>> In case recording similar to SIPREC, it is little bit more
>>>>> than spanning two media (RTP stream) alone because recording
>>>>> has to include some context data about recording apart from
>>>>> the media stream.
>>>>>
>>>>> Thanks
>>>>> Partha
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>>> Behalf Of Stefan Håkansson LK
>>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>>
>>>>>>> However, I did suggest (in other text in my previous
>>>>> message) that one
>>>>>> possible solution might be to record locally and use a
>>>>> second RTC-Web
>>>>>> session to transmit from the local file to the>remote
>>>>> recorder. What I
>>>>>> failed to say was that in this case the local file would be
>>>>> a temporary
>>>>>> repository - just a buffer between the two sessions.
>>>>>> This makes sense. Also, if you look at the API proposals
>>>>> available, it
>>>>>> would be quite easy to forward (in real time) a stream
>>>>> being received
>>>>>> to another entity. There is no explicit recording, a stream being
>>>>>> received (via RTP) is just streamed to another entity (via
>>>>> a separate
>>>>>> RTC-Web session). I think this would solve this case.
>>>>>>
>>>>>> Stefan
>>>>>> _______________________________________________
>>>>>> 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 pkyzivat@alum.mit.edu  Mon Aug 22 06:17:54 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/22/11 2:28 AM, Elwell, John wrote:
> Partha,
>
> You are talking here about the metadata, I think. I assume the web page / JavaScript has to deal with that - not the browser.

I don't understand your point John. The javascript is executed by the 
browser.

The actual awareness of information that is to become siprec metadata 
may well be distributed - some of it in the client/browser, and some of 
it in the server. That will somewhat complicate getting it all together.

	Thanks,
	Paul

> John
>
>
>> -----Original Message-----
>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>> Sent: 19 August 2011 18:19
>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>> public-webrtc@w3.org
>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>
>> Stefan,
>>
>> In case recording similar to SIPREC, it is little bit more
>> than spanning two media (RTP stream) alone because recording
>> has to include some context data about recording apart from
>> the media stream.
>>
>> Thanks
>> Partha
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Stefan Håkansson LK
>>> Sent: Friday, August 19, 2011 8:26 PM
>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>
>>>> However, I did suggest (in other text in my previous
>> message) that one
>>> possible solution might be to record locally and use a
>> second RTC-Web
>>> session to transmit from the local file to the>remote
>> recorder. What I
>>> failed to say was that in this case the local file would be
>> a temporary
>>> repository - just a buffer between the two sessions.
>>> This makes sense. Also, if you look at the API proposals
>> available, it
>>> would be quite easy to forward (in real time) a stream
>> being received
>>> to another entity. There is no explicit recording, a stream being
>>> received (via RTP) is just streamed to another entity (via
>> a separate
>>> RTC-Web session). I think this would solve this case.
>>>
>>> Stefan
>>> _______________________________________________
>>> 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 harald@alvestrand.no  Mon Aug 22 07:08:40 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 08/22/11 13:19, Stefan Håkansson LK wrote:
> On 2011-08-22 10:16, Ravindran Parthasarathi wrote:
>
>> ... Currently, let us have common understanding whether recording 
>> usecase has to be added in RTCWeb or not.
> Agree. And also _which_ recording use case(s) in that case. I think 
> this is what John was looking for when starting the thread.
>
> My $0.02 says that we need to take some care before adding more and 
> more usages and reqs - after all the schedules for the WGs are quite 
> aggressive.
Indeed. When thinking about the recording use cases, I could come up 
with a few interpretations:

- Record one media stream, as it arrives (the "voice mailbox" concept)
- Record a conversation: Outgoing and incoming audio and video in an 
1-on-1 or 1-on-N context (typically as permanent records of a meeting or 
conversation)
- Record the application screen as presented to the user, together with 
the audio tracks as presented to the user's audio device, but don't 
bother with incoming audio from the user at all (game recording, for 
instance - the "WoW movie" kind)

In each instance, recording technology might include recording:

   - As one track, somehow mixed, ready for playback
   - As multiple tracks recorded separately within a container that 
requires special playback devices
   - As multiple tracks, recorded separately to separate objects
   - As streams sent to some remote recording entity, that may choose 
one of the options above

I'd be happier about discussing which ones of these people find it 
critical to support before we dive into figuring out how to support them.

Send text!

                       Harald


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To: Harald Alvestrand <harald@alvestrand.no>, =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
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=20

> -----Original Message-----
> From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
> Sent: 22 August 2011 15:10
> To: Stefan H=E5kansson LK
> Cc: Ravindran Parthasarathi; Avasarala, Ranjit; Elwell, John;=20
> rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>=20
> On 08/22/11 13:19, Stefan H=E5kansson LK wrote:
> > On 2011-08-22 10:16, Ravindran Parthasarathi wrote:
> >
> >> ... Currently, let us have common understanding whether recording=20
> >> usecase has to be added in RTCWeb or not.
> > Agree. And also _which_ recording use case(s) in that case. I think=20
> > this is what John was looking for when starting the thread.
> >
> > My $0.02 says that we need to take some care before adding more and=20
> > more usages and reqs - after all the schedules for the WGs=20
> are quite=20
> > aggressive.
> Indeed. When thinking about the recording use cases, I could come up=20
> with a few interpretations:
>=20
> - Record one media stream, as it arrives (the "voice mailbox" concept)
[JRE] This was not what I originally proposed, but emerged during discussio=
n. From my point of view a mailbox is more likely to be centralized, not on=
 a user's device, so the web application can divert calls to a mailbox usin=
g signalling means, without any impact on the browser. Some people might wa=
nt to support a local mailbox, in which case browser support would be neede=
d, but that is second priority, from my perspective.

> - Record a conversation: Outgoing and incoming audio and video in an=20
> 1-on-1 or 1-on-N context (typically as permanent records of a=20
> meeting or=20
> conversation)
[JRE] This was what I was originally asking about. Seems to be important to=
 have this capability from day 1, or at least an architecture that can be e=
xtended to do this at day 2.


> - Record the application screen as presented to the user,=20
> together with=20
> the audio tracks as presented to the user's audio device, but don't=20
> bother with incoming audio from the user at all (game recording, for=20
> instance - the "WoW movie" kind)
[JRE] Not high priority from my perspective.

>=20
> In each instance, recording technology might include recording:
>=20
>    - As one track, somehow mixed, ready for playback
>    - As multiple tracks recorded separately within a container that=20
> requires special playback devices
[JRE] I think the container here is what is known as metadata, from a sessi=
on recording point of view. That container can be a matter for the web appl=
ication.

>    - As multiple tracks, recorded separately to separate objects
[JRE] Yes, although the possibility of mixing (e.g., the two directions of =
an audio stream) would be good to have.

>    - As streams sent to some remote recording entity, that may choose=20
> one of the options above
[JRE] Yes, this is my primary concern, coupled with the "record a conversat=
ion" above.

>=20
> I'd be happier about discussing which ones of these people find it=20
> critical to support before we dive into figuring out how to=20
> support them.
>=20
> Send text!
[JRE] I was just about to - look out for separate postings for local and re=
mote recording (I will leave others to proposed text for mailbox, if they r=
equire it).

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


>=20
>                        Harald
>=20
> =

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To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Mon, 22 Aug 2011 16:42:11 +0200
Thread-Topic: Proposed text for local recording use case
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4.2.xx Local Session Recording
In this use case, the web application user wishes to record a real-time com=
munication locally, such that transmitted and received audio, video or othe=
r real-time media are stored in one or more files. For a given medium, the =
two directions of transmission can be stored in the same file (mixed) or in=
 separate files. The web application also stores metadata that gives contex=
t to the stored media.

New requirements:
Fxx1: The browser MUST be able to store transmitted and received media in l=
ocal files.

Axx1: The web application MUST be able to ask the browser to store transmit=
ted and received media in local files, and in the case of audio at least, a=
sk for the two directions of transmission to be stored mixed or separately.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

From john.elwell@siemens-enterprise.com  Mon Aug 22 07:41:19 2011
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Mon, 22 Aug 2011 16:42:21 +0200
Thread-Topic: Proposed text for remote recording use case
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4.2.yy Remote Session Recording
In this use case, the web application user wishes to record a real-time com=
munication at a remote recording device, such that transmitted and received=
 audio, video or other real-time media are transmitted in real-time to the =
remote device. The remote device can perform archiving, retrieval, playback=
, etc., but can also perform real-time analytics on the media. A typical de=
ployment might be in a contact centre. For a given medium, the two directio=
ns of transmission can be transmitted together (mixed) or separately. The w=
eb application also sends metadata that gives context to the stored media.

New requirements:
Fyy1: The browser MUST be able to send in real-time to a remote recording d=
evice media that are being transmitted to and received from remote particip=
ants.

Ayy1: The web application MUST be able to ask the browser to transmit in re=
al-time to a remote recording device media that are being transmitted to an=
d received from remote participants and, in the case of audio at least, ask=
 for the two directions of transmission to be transmitted to the remote rec=
ording device mixed or separately.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.=

From pkyzivat@alum.mit.edu  Mon Aug 22 07:59:07 2011
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Subject: Re: [rtcweb] Proposed text for local recording use case
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This is a good start. But I'd like to dig a little deeper into the 
intent here.

The below says the *user* wishes to record, and the *browser* must be 
able to do it. But is that the only case of interest?

ISTM that in a number of cases it will be the web application that wants 
the recording, even if there is an obligation to inform the user that it 
is happening. And the behavior of the application may be changed 
substantially if the recording cannot be made.

(Consider a web app provided by a brokerage to its clients.)

OTOH, maybe some of these cases are out of scope because the 
user+browser can't be sufficiently trusted, so that its necessary to do 
the recording from some secure server.

	Thanks,
	Paul

On 8/22/11 10:42 AM, Elwell, John wrote:
> 4.2.xx Local Session Recording
> In this use case, the web application user wishes to record a real-time communication locally, such that transmitted and received audio, video or other real-time media are stored in one or more files. For a given medium, the two directions of transmission can be stored in the same file (mixed) or in separate files. The web application also stores metadata that gives context to the stored media.
>
> New requirements:
> Fxx1: The browser MUST be able to store transmitted and received media in local files.
>
> Axx1: The web application MUST be able to ask the browser to store transmitted and received media in local files, and in the case of audio at least, ask for the two directions of transmission to be stored mixed or separately.
>
> John
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Mon, 22 Aug 2011 17:15:19 +0200
Thread-Topic: [rtcweb] Use cases - recording and voicemail
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 22 August 2011 14:19
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Use cases - recording and voicemail
>=20
> On 8/22/11 2:28 AM, Elwell, John wrote:
> > Partha,
> >
> > You are talking here about the metadata, I think. I assume=20
> the web page / JavaScript has to deal with that - not the browser.
>=20
> I don't understand your point John. The javascript is executed by the=20
> browser.
[JRE] My point is that in the same way as SIP signalling (at least in some =
people's minds) is a matter for JS, the inclusion of a metadata body part i=
n that SIP signalling is also a matter for the JS. Of course, if SIP is inc=
luded in the browser, this is a different matter.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

>=20
> The actual awareness of information that is to become siprec metadata=20
> may well be distributed - some of it in the client/browser,=20
> and some of=20
> it in the server. That will somewhat complicate getting it=20
> all together.
>=20
> 	Thanks,
> 	Paul
>=20
> > John
> >
> >
> >> -----Original Message-----
> >> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
> >> Sent: 19 August 2011 18:19
> >> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
> >> public-webrtc@w3.org
> >> Subject: RE: [rtcweb] Use cases - recording and voicemail
> >>
> >> Stefan,
> >>
> >> In case recording similar to SIPREC, it is little bit more
> >> than spanning two media (RTP stream) alone because recording
> >> has to include some context data about recording apart from
> >> the media stream.
> >>
> >> Thanks
> >> Partha
> >>
> >>> -----Original Message-----
> >>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >>> Behalf Of Stefan H=E5kansson LK
> >>> Sent: Friday, August 19, 2011 8:26 PM
> >>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> >>> Subject: Re: [rtcweb] Use cases - recording and voicemail
> >>>
> >>>> However, I did suggest (in other text in my previous
> >> message) that one
> >>> possible solution might be to record locally and use a
> >> second RTC-Web
> >>> session to transmit from the local file to the>remote
> >> recorder. What I
> >>> failed to say was that in this case the local file would be
> >> a temporary
> >>> repository - just a buffer between the two sessions.
> >>> This makes sense. Also, if you look at the API proposals
> >> available, it
> >>> would be quite easy to forward (in real time) a stream
> >> being received
> >>> to another entity. There is no explicit recording, a stream being
> >>> received (via RTP) is just streamed to another entity (via
> >> a separate
> >>> RTC-Web session). I think this would solve this case.
> >>>
> >>> Stefan
> >>> _______________________________________________
> >>> rtcweb mailing list
> >>> rtcweb@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rtcweb
> >>
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From pkyzivat@alum.mit.edu  Mon Aug 22 08:44:46 2011
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Date: Mon, 22 Aug 2011 11:45:41 -0400
From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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To: "Elwell, John" <john.elwell@siemens-enterprise.com>
References: <BBF498F2D030E84AB1179E24D1AC41D616C389F16D@ESESSCMS0362.eemea.ericsson.se><4E3AB4D4.4070308@jesup.org>, <A444A0F8084434499206E78C106220CA09BDB6A238@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F242@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A34F@MCHP058A.global-ad.net><BBF498F2D030E84AB1179E24D1AC41D616C389F245@ESESSCMS0362.eemea.ericsson.se>, <A444A0F8084434499206E78C106220CA09BDB6A397@MCHP058A.global-ad.net> <BBF498F2D030E84AB1179E24D1AC41D616C389F24A@ESESSCMS0362.eemea.ericsson.se> <2E239D6FCD033C4BAF15F386A979BF510640D0@sonusinmail02.sonusnet.com> <A444A0F8084434499206E78C106220CA0B00FDABAD@MCHP058A.global-ad.net> <4E525734.80209@alum.mit.edu> <A444A0F8084434499206E78C106220CA0B00FDAEA5@MCHP058A.global-ad.net>
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/22/11 11:15 AM, Elwell, John wrote:
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: 22 August 2011 14:19
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>
>> On 8/22/11 2:28 AM, Elwell, John wrote:
>>> Partha,
>>>
>>> You are talking here about the metadata, I think. I assume
>> the web page / JavaScript has to deal with that - not the browser.
>>
>> I don't understand your point John. The javascript is executed by the
>> browser.
> [JRE] My point is that in the same way as SIP signalling (at least in some people's minds) is a matter for JS, the inclusion of a metadata body part in that SIP signalling is also a matter for the JS. Of course, if SIP is included in the browser, this is a different matter.
>
> John

OK. Now I get it.

Well, then again this takes on a different cast based on whether, or 
not, you think the browser should have a sip stack that is "driven" by JS.

If so, then probably the JS will also need to establish the SIP 
Recording Session, and send the metadata in the sip signaling. 
Presumably that would have to be part of the JS API for dealing with the 
sip stack.

OTOH, if you don't think the browser and JS should be doing sip 
signaling, then the web server is probably establishing the Recording 
Session, and sending the metadata. But the media streams may not flow 
through the web server. So in that case, the web server, via the JS that 
it provides, will require the capability to provide a forked media 
stream that the server can then reference in its SDP for the RS. And 
also, the JS will need *some* way to provide to the server the info that 
should go into the metadata.

There had better be *some* standards for how to do it or else it will be 
a high bar for supporting recording. Perhaps the metadata could be 
provided by the JS to the web server using the same XML schema being 
defined by siprec, but conveyed differently, using http. But I expect it 
will be at least a little more complicated than that.

	Thanks,
	Paul

> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
>
>>
>> The actual awareness of information that is to become siprec metadata
>> may well be distributed - some of it in the client/browser,
>> and some of
>> it in the server. That will somewhat complicate getting it
>> all together.
>>
>> 	Thanks,
>> 	Paul
>>
>>> John
>>>
>>>
>>>> -----Original Message-----
>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>> Sent: 19 August 2011 18:19
>>>> To: Stefan Håkansson LK; Elwell, John; rtcweb@ietf.org;
>>>> public-webrtc@w3.org
>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>
>>>> Stefan,
>>>>
>>>> In case recording similar to SIPREC, it is little bit more
>>>> than spanning two media (RTP stream) alone because recording
>>>> has to include some context data about recording apart from
>>>> the media stream.
>>>>
>>>> Thanks
>>>> Partha
>>>>
>>>>> -----Original Message-----
>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>> Behalf Of Stefan Håkansson LK
>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>>> However, I did suggest (in other text in my previous
>>>> message) that one
>>>>> possible solution might be to record locally and use a
>>>> second RTC-Web
>>>>> session to transmit from the local file to the>remote
>>>> recorder. What I
>>>>> failed to say was that in this case the local file would be
>>>> a temporary
>>>>> repository - just a buffer between the two sessions.
>>>>> This makes sense. Also, if you look at the API proposals
>>>> available, it
>>>>> would be quite easy to forward (in real time) a stream
>>>> being received
>>>>> to another entity. There is no explicit recording, a stream being
>>>>> received (via RTP) is just streamed to another entity (via
>>>> a separate
>>>>> RTC-Web session). I think this would solve this case.
>>>>>
>>>>> Stefan
>>>>> _______________________________________________
>>>>> rtcweb mailing list
>>>>> rtcweb@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>


From csp@csperkins.org  Mon Aug 22 09:48:27 2011
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References: <7F2072F1E0DE894DA4B517B93C6A05852203D41B70@ESESSCMS0356.eemea.ericsson.se> <4e4d793b-8748-41c8-a9ea-ded2237cda03@email.android.com> <7F2072F1E0DE894DA4B517B93C6A05852203CD99E6@ESESSCMS0356.eemea.ericsson.se> <4E48D9E2.5080903@alvestrand.no>
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--Apple-Mail-8-717121590
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On 15 Aug 2011, at 09:33, Harald Alvestrand wrote:
> On 08/15/11 09:57, Christer Holmberg wrote:
>>=20
>> There are discussions in other SDOs about using AVPF for video and =
AVP for audio,             within a session.
>> =20
>> However, those discussions are not within the context of multiplex.
> I added the following text:
>=20
> The reason for the requirement for systematic proto is that there are =
many combinations that don't make sense (for instance "RTP/AVPF" in one =
section and "RTP/SAVP" in another would make encryption and availability =
of TMMBR depend on the outcome of negotiation, which seems strange). The =
cases where combinations make sense (RTP/AVPF with UDP/FEC for instance) =
also usually require that separate RTP sessions be used. [[QUESTION IN =
DRAFT: Are there sensible combinations?]]
>=20
> Makes sense?
>>=20


The only combinations I can think of that make sense are RTP/AVP with =
RTP/AVPF, and RTP/SAVP with RTP/SAVPF. It's not clear why you'd bother =
though, since the AVPF variant behaves the same as the AVP variant if =
you don't use the feedback.

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




--Apple-Mail-8-717121590
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	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><div><div>On 15 Aug 2011, at 09:33, Harald Alvestrand =
wrote:</div><blockquote type=3D"cite">

 =20
    <meta content=3D"text/html; charset=3DISO-8859-1" =
http-equiv=3D"Content-Type">
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000">
    On 08/15/11 09:57, Christer Holmberg wrote:
    <blockquote =
cite=3D"mid:7F2072F1E0DE894DA4B517B93C6A05852203CD99E6@ESESSCMS0356.eemea.=
ericsson.se" type=3D"cite">
      <meta http-equiv=3D"Content-Type" content=3D"text/html;
        charset=3DISO-8859-1">
      <meta content=3D"MSHTML 6.00.6002.18457" name=3D"GENERATOR">
      <div dir=3D"ltr" align=3D"left"><span class=3D"Apple-style-span" =
style=3D"color: rgb(0, 0, 255); font-family: Arial; font-size: small; =
">There are discussions
            in other SDOs about using AVPF for video and AVP for audio,
            within a session.</span></div>
      <div dir=3D"ltr" align=3D"left"><span =
class=3D"898335607-15082011"></span>&nbsp;</div>
      <div dir=3D"ltr" align=3D"left"><span =
class=3D"898335607-15082011"><font color=3D"#0000ff" face=3D"Arial" =
size=3D"2">However, those
            discussions are not within the context of =
multiplex.</font></span></div>
    </blockquote>
    I added the following text:<br>
    <br>
    The reason for the requirement for systematic proto is that there
    are many combinations that don't make sense (for instance "RTP/AVPF"
    in one section and "RTP/SAVP" in another would make encryption and
    availability of TMMBR depend on the outcome of negotiation, which
    seems strange). The cases where combinations make sense (RTP/AVPF
    with UDP/FEC for instance) also usually require that separate RTP
    sessions be used. [[QUESTION IN DRAFT: Are there sensible
    combinations?]]<br>
    <br>
    Makes sense?<br>
    <blockquote =
cite=3D"mid:7F2072F1E0DE894DA4B517B93C6A05852203CD99E6@ESESSCMS0356.eemea.=
ericsson.se" type=3D"cite">
      <div dir=3D"ltr" align=3D"left"><span =
class=3D"898335607-15082011"></span></div></blockquote></div></blockquote>=
</div><div><br class=3D"webkit-block-placeholder"></div><div><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div>The only =
combinations I can think of that make sense are RTP/AVP with RTP/AVPF, =
and RTP/SAVP with RTP/SAVPF. It's not clear why you'd bother though, =
since the AVPF variant behaves the same as the AVP variant if you don't =
use the feedback.<br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></body></html>=

--Apple-Mail-8-717121590--

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To: Muthu Arul Mozhi Perumal (mperumal) <mperumal@cisco.com>
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Subject: Re: [rtcweb] Fwd: I-D Action: draft-alvestrand-one-rtp-00.txt
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On 17 Aug 2011, at 10:27, Muthu Arul Mozhi Perumal (mperumal) wrote:
>> I think it should be up to the offer generator whether he adds =
"a=3Drtcp-mux" to the first section or not.
>> =20
> Since the goal is to minimize the number of transport connections I =
think we should put something stronger to encourage the offerer and =
answerer to support RTCP multiplexing. At the minimum, a normative =
statement recommending the use of RTCP multiplexing with this grouping =
semantics would go a long way (symmetric RTP for e.g is a =
recommendation, but virtually all implementations support it).

Such recommendations are in draft-perkins-rtcweb-rtp-usage-02. Feedback =
on that draft is appreciated.

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




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From: Jim McEachern <jim.mceachern@genband.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Thread-Topic: Proposed text for remote recording use case
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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John,
It has previously been pointed out that in some jurisdictions there is a le=
gal requirement to insert a tone (e.g. an intermittent beep) that indicates=
 the conversation is being recorded.  It seems like this would be a good th=
ing to mention here. =20

If this makes sense, I'm thinking something like the following sentence add=
ed to the end of 4.2.yy:
"... If required, the web application will direct the browser to insert an =
appropriate indication (e.g. an intermediate beep) into the media stream to=
 show that the communication is being recorded."

This also suggests an additional requirement. =20
"Ayy2: The web application MUST be able to ask the browser to insert an app=
ropriate indication into the media stream to indicate that the communicatio=
n is being recorded."

I believe that F15 already covers the requirments for the browser in this c=
ase.
F15:         The browser MUST be able to process and mix
                   sound objects (media that is retrieved from another
                   source than the established media stream(s) with the
                   peer(s) with audio streams).

Thoughts?
Jim

-----Original Message-----
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Elwell, John
Sent: Monday, August 22, 2011 10:42 AM
To: rtcweb@ietf.org; public-webrtc@w3.org
Subject: [rtcweb] Proposed text for remote recording use case

4.2.yy Remote Session Recording
In this use case, the web application user wishes to record a real-time com=
munication at a remote recording device, such that transmitted and received=
 audio, video or other real-time media are transmitted in real-time to the =
remote device. The remote device can perform archiving, retrieval, playback=
, etc., but can also perform real-time analytics on the media. A typical de=
ployment might be in a contact centre. For a given medium, the two directio=
ns of transmission can be transmitted together (mixed) or separately. The w=
eb application also sends metadata that gives context to the stored media.

New requirements:
Fyy1: The browser MUST be able to send in real-time to a remote recording d=
evice media that are being transmitted to and received from remote particip=
ants.

Ayy1: The web application MUST be able to ask the browser to transmit in re=
al-time to a remote recording device media that are being transmitted to an=
d received from remote participants and, in the case of audio at least, ask=
 for the two directions of transmission to be transmitted to the remote rec=
ording device mixed or separately.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb

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To: "Jim McEachern" <jim.mceachern@genband.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, <rtcweb@ietf.org>,  <public-webrtc@w3.org>
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Hi Jim,

SIPREC includes the notion of recording awareness, and in the case of a
recording aware device you can signal in SDP the fact that something is
being recorded rather than requiring an indication to be inserted into
the media stream. I think RTCWEB should be able to leverage this
mechanism. In that case, the indication of the media being recorded may
be communicated by the web application or the browser, but it need not
necessarily be inserted in the actual media.

Cheers,
Charles=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Jim McEachern
> Sent: Monday, August 22, 2011 11:30 AM
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Proposed text for remote recording use case
>=20
> John,
> It has previously been pointed out that in some jurisdictions there is
a legal requirement to insert a
> tone (e.g. an intermittent beep) that indicates the conversation is
being recorded.  It seems like
> this would be a good thing to mention here.
>=20
> If this makes sense, I'm thinking something like the following
sentence added to the end of 4.2.yy:
> "... If required, the web application will direct the browser to
insert an appropriate indication
> (e.g. an intermediate beep) into the media stream to show that the
communication is being recorded."
>=20
> This also suggests an additional requirement.
> "Ayy2: The web application MUST be able to ask the browser to insert
an appropriate indication into
> the media stream to indicate that the communication is being
recorded."
>=20
> I believe that F15 already covers the requirments for the browser in
this case.
> F15:         The browser MUST be able to process and mix
>                    sound objects (media that is retrieved from another
>                    source than the established media stream(s) with
the
>                    peer(s) with audio streams).
>=20
> Thoughts?
> Jim
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Elwell, John
> Sent: Monday, August 22, 2011 10:42 AM
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: [rtcweb] Proposed text for remote recording use case
>=20
> 4.2.yy Remote Session Recording
> In this use case, the web application user wishes to record a
real-time communication at a remote
> recording device, such that transmitted and received audio, video or
other real-time media are
> transmitted in real-time to the remote device. The remote device can
perform archiving, retrieval,
> playback, etc., but can also perform real-time analytics on the media.
A typical deployment might be
> in a contact centre. For a given medium, the two directions of
transmission can be transmitted
> together (mixed) or separately. The web application also sends
metadata that gives context to the
> stored media.
>=20
> New requirements:
> Fyy1: The browser MUST be able to send in real-time to a remote
recording device media that are being
> transmitted to and received from remote participants.
>=20
> Ayy1: The web application MUST be able to ask the browser to transmit
in real-time to a remote
> recording device media that are being transmitted to and received from
remote participants and, in the
> case of audio at least, ask for the two directions of transmission to
be transmitted to the remote
> recording device mixed or separately.
>=20
> John
>=20
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of
Siemens AG.
> _______________________________________________
> 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  Mon Aug 22 12:01:19 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Paul Kyzivat" <pkyzivat@alum.mit.edu>, "Elwell, John" <john.elwell@siemens-enterprise.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Use cases - recording and voicemail
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Please read inline

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>Of Paul Kyzivat
>Sent: Monday, August 22, 2011 9:16 PM
>To: Elwell, John
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Use cases - recording and voicemail
>
>On 8/22/11 11:15 AM, Elwell, John wrote:
>>
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
>>> Sent: 22 August 2011 14:19
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>
>>> On 8/22/11 2:28 AM, Elwell, John wrote:
>>>> Partha,
>>>>
>>>> You are talking here about the metadata, I think. I assume
>>> the web page / JavaScript has to deal with that - not the browser.
>>>
>>> I don't understand your point John. The javascript is executed by =
the
>>> browser.
>> [JRE] My point is that in the same way as SIP signalling (at least in
>some people's minds) is a matter for JS, the inclusion of a metadata
>body part in that SIP signalling is also a matter for the JS. Of =
course,
>if SIP is included in the browser, this is a different matter.
>>
>> John
>
>OK. Now I get it.
>
>Well, then again this takes on a different cast based on whether, or
>not, you think the browser should have a sip stack that is "driven" by
>JS.
>
>If so, then probably the JS will also need to establish the SIP
>Recording Session, and send the metadata in the sip signaling.
>Presumably that would have to be part of the JS API for dealing with =
the
>sip stack.
>
[Partha] This is my understanding. Here, it is possible to complete =
reuse SIPREC work with minor extensions specific to RTCWeb like codec =
profile.

>OTOH, if you don't think the browser and JS should be doing sip
>signaling, then the web server is probably establishing the Recording
>Session, and sending the metadata. But the media streams may not flow
>through the web server. So in that case, the web server, via the JS =
that
>it provides, will require the capability to provide a forked media
>stream that the server can then reference in its SDP for the RS. And
>also, the JS will need *some* way to provide to the server the info =
that
>should go into the metadata.
>
>There had better be *some* standards for how to do it or else it will =
be
>a high bar for supporting recording. Perhaps the metadata could be
>provided by the JS to the web server using the same XML schema being
>defined by siprec, but conveyed differently, using http. But I expect =
it
>will be at least a little more complicated than that.
[Partha] I agree with you that two way for passing metadata & media is =
more complicated which is avoided in SIPREC protocol design.
>
>	Thanks,
>	Paul
>
>> John Elwell
>> Tel: +44 1908 817801 (office and mobile)
>> Email: john.elwell@siemens-enterprise.com
>> http://www.siemens-enterprise.com/uk/
>>
>> Siemens Enterprise Communications Limited.
>> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
>0DJ.
>> Registered No: 5903714, England.
>>
>> Siemens Enterprise Communications Limited is a Trademark Licensee of
>Siemens AG.
>>
>>>
>>> The actual awareness of information that is to become siprec =
metadata
>>> may well be distributed - some of it in the client/browser,
>>> and some of
>>> it in the server. That will somewhat complicate getting it
>>> all together.
>>>
>>> 	Thanks,
>>> 	Paul
>>>
>>>> John
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: Ravindran Parthasarathi [mailto:pravindran@sonusnet.com]
>>>>> Sent: 19 August 2011 18:19
>>>>> To: Stefan H=E5kansson LK; Elwell, John; rtcweb@ietf.org;
>>>>> public-webrtc@w3.org
>>>>> Subject: RE: [rtcweb] Use cases - recording and voicemail
>>>>>
>>>>> Stefan,
>>>>>
>>>>> In case recording similar to SIPREC, it is little bit more
>>>>> than spanning two media (RTP stream) alone because recording
>>>>> has to include some context data about recording apart from
>>>>> the media stream.
>>>>>
>>>>> Thanks
>>>>> Partha
>>>>>
>>>>>> -----Original Message-----
>>>>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>>>>> Behalf Of Stefan H=E5kansson LK
>>>>>> Sent: Friday, August 19, 2011 8:26 PM
>>>>>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>>>>>> Subject: Re: [rtcweb] Use cases - recording and voicemail
>>>>>>
>>>>>>> However, I did suggest (in other text in my previous
>>>>> message) that one
>>>>>> possible solution might be to record locally and use a
>>>>> second RTC-Web
>>>>>> session to transmit from the local file to the>remote
>>>>> recorder. What I
>>>>>> failed to say was that in this case the local file would be
>>>>> a temporary
>>>>>> repository - just a buffer between the two sessions.
>>>>>> This makes sense. Also, if you look at the API proposals
>>>>> available, it
>>>>>> would be quite easy to forward (in real time) a stream
>>>>> being received
>>>>>> to another entity. There is no explicit recording, a stream being
>>>>>> received (via RTP) is just streamed to another entity (via
>>>>> a separate
>>>>>> RTC-Web session). I think this would solve this case.
>>>>>>
>>>>>> Stefan
>>>>>> _______________________________________________
>>>>>> 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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From: Jim McEachern <jim.mceachern@genband.com>
To: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>,  "public-webrtc@w3.org" <public-webrtc@w3.org>
Thread-Topic: [rtcweb] Proposed text for remote recording use case
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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Charles,=20
Does this mean that there isn't a need to say anything, or that we should s=
tate there is a requirement to  indicate the communication is being recorde=
d, but not assume you will have to insert something into the media path?

Jim


-----Original Message-----
From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]=20
Sent: Monday, August 22, 2011 2:43 PM
To: Jim McEachern; Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
Subject: RE: [rtcweb] Proposed text for remote recording use case

Hi Jim,

SIPREC includes the notion of recording awareness, and in the case of a rec=
ording aware device you can signal in SDP the fact that something is being =
recorded rather than requiring an indication to be inserted into the media =
stream. I think RTCWEB should be able to leverage this mechanism. In that c=
ase, the indication of the media being recorded may be communicated by the =
web application or the browser, but it need not necessarily be inserted in =
the actual media.

Cheers,
Charles=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Jim McEachern
> Sent: Monday, August 22, 2011 11:30 AM
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Proposed text for remote recording use case
>=20
> John,
> It has previously been pointed out that in some jurisdictions there is
a legal requirement to insert a
> tone (e.g. an intermittent beep) that indicates the conversation is
being recorded.  It seems like
> this would be a good thing to mention here.
>=20
> If this makes sense, I'm thinking something like the following
sentence added to the end of 4.2.yy:
> "... If required, the web application will direct the browser to
insert an appropriate indication
> (e.g. an intermediate beep) into the media stream to show that the
communication is being recorded."
>=20
> This also suggests an additional requirement.
> "Ayy2: The web application MUST be able to ask the browser to insert
an appropriate indication into
> the media stream to indicate that the communication is being
recorded."
>=20
> I believe that F15 already covers the requirments for the browser in
this case.
> F15:         The browser MUST be able to process and mix
>                    sound objects (media that is retrieved from another
>                    source than the established media stream(s) with
the
>                    peer(s) with audio streams).
>=20
> Thoughts?
> Jim
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf Of Elwell, John
> Sent: Monday, August 22, 2011 10:42 AM
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: [rtcweb] Proposed text for remote recording use case
>=20
> 4.2.yy Remote Session Recording
> In this use case, the web application user wishes to record a
real-time communication at a remote
> recording device, such that transmitted and received audio, video or
other real-time media are
> transmitted in real-time to the remote device. The remote device can
perform archiving, retrieval,
> playback, etc., but can also perform real-time analytics on the media.
A typical deployment might be
> in a contact centre. For a given medium, the two directions of
transmission can be transmitted
> together (mixed) or separately. The web application also sends
metadata that gives context to the
> stored media.
>=20
> New requirements:
> Fyy1: The browser MUST be able to send in real-time to a remote
recording device media that are being
> transmitted to and received from remote participants.
>=20
> Ayy1: The web application MUST be able to ask the browser to transmit
in real-time to a remote
> recording device media that are being transmitted to and received from
remote participants and, in the
> case of audio at least, ask for the two directions of transmission to
be transmitted to the remote
> recording device mixed or separately.
>=20
> John
>=20
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of
Siemens AG.
> _______________________________________________
> 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 - recording and voicemail
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On 8/22/2011 9:12 AM, Paul Kyzivat wrote:

> On 8/22/11 7:10 AM, Harald Alvestrand wrote:
>> On 08/22/11 12:17, Avasarala, Ranjit wrote:
>>> Hi Partha
>>>
>>> I agree with Paul that we could use the recording mechanism discussed
>>> in siprec mailing list could be added here as webbrowser would
>>> essentially be a SIP UA which could initiate and terminate SIP sessions.
>> mandatory standard clarification (?):
>>
>> .. as somewhere in the combination of the Web browser, the Javascript
>> and the Web server it connects to, there would be something that
>> performs the functions of a SIP UA which could initiate and terminate
>> SIP sessions ...
>
> Yes, *something* of this sort - the devil is in the details.
>
> IIUC, its the intent of RTCWEB that the media flow directly between
> source and recipient, rather than following the path of the "signaling".
>
> (That of course was also the intent of SIP with proxies, but the almost
> pervasive use of SBCs has usually forced media to follow the signaling
> path.)
>
> In siprec, it is assumed that an entity called the SRC is in the
> signaling and media path of the call being recorded, as well as in the
> signaling and media path of a session with the recorder.
>
> If we assume that the RTCWEB "client" is in the media path, but not in a
> sip signaling path, and that the RTCWEB "server" is sip-capable but not
> in the media path, then we have a structure that doesn't quite fit the
> siprec model.
>
> However this could perhaps be made to work with siprec via some
> additional functionality in the RTCWEB client and server, and maybe some
> tweaks to what is being specified for siprec. The details will depend on
> which end is making the decision to do recording. If its the client,
> then that client will need a way to ask "the" server to establish a
> siprec session and to pass metadata. If its the server that makes the
> decision, then there needs to be a way for the server to ask the client
> to fork media and send it to multiple destinations.
>
> There are of course privacy concerns in this. Whether they are different
> in RTCWEB than they are for siprec in general remains to be seen.

There are also other practical issues.  In particular with bandwidth: you've
now roughly tripled the bandwidth requirement of the endpoint: it must send
to the other end, AND mirror both outgoing (which it controls) and incoming
(which it doesn't, or only controls very indirectly) to the recording server.


For cases where you have bandwidth to burn (audio-only calls on broadband)
you might get away with it.  For a video call under any sort of bandwidth
restraint, it would be a huge problem and would seriously degrade quality.


-- 
Randell Jesup
randell-ietf@jesup.org


From pkyzivat@alum.mit.edu  Mon Aug 22 13:45:31 2011
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Subject: Re: [rtcweb] Use cases - recording and voicemail
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On 8/22/11 4:11 PM, Randell Jesup wrote:
> On 8/22/2011 9:12 AM, Paul Kyzivat wrote:
>
>> On 8/22/11 7:10 AM, Harald Alvestrand wrote:
>>> On 08/22/11 12:17, Avasarala, Ranjit wrote:
>>>> Hi Partha
>>>>
>>>> I agree with Paul that we could use the recording mechanism discussed
>>>> in siprec mailing list could be added here as webbrowser would
>>>> essentially be a SIP UA which could initiate and terminate SIP
>>>> sessions.
>>> mandatory standard clarification (?):
>>>
>>> .. as somewhere in the combination of the Web browser, the Javascript
>>> and the Web server it connects to, there would be something that
>>> performs the functions of a SIP UA which could initiate and terminate
>>> SIP sessions ...
>>
>> Yes, *something* of this sort - the devil is in the details.
>>
>> IIUC, its the intent of RTCWEB that the media flow directly between
>> source and recipient, rather than following the path of the "signaling".
>>
>> (That of course was also the intent of SIP with proxies, but the almost
>> pervasive use of SBCs has usually forced media to follow the signaling
>> path.)
>>
>> In siprec, it is assumed that an entity called the SRC is in the
>> signaling and media path of the call being recorded, as well as in the
>> signaling and media path of a session with the recorder.
>>
>> If we assume that the RTCWEB "client" is in the media path, but not in a
>> sip signaling path, and that the RTCWEB "server" is sip-capable but not
>> in the media path, then we have a structure that doesn't quite fit the
>> siprec model.
>>
>> However this could perhaps be made to work with siprec via some
>> additional functionality in the RTCWEB client and server, and maybe some
>> tweaks to what is being specified for siprec. The details will depend on
>> which end is making the decision to do recording. If its the client,
>> then that client will need a way to ask "the" server to establish a
>> siprec session and to pass metadata. If its the server that makes the
>> decision, then there needs to be a way for the server to ask the client
>> to fork media and send it to multiple destinations.
>>
>> There are of course privacy concerns in this. Whether they are different
>> in RTCWEB than they are for siprec in general remains to be seen.
>
> There are also other practical issues. In particular with bandwidth: you've
> now roughly tripled the bandwidth requirement of the endpoint: it must send
> to the other end, AND mirror both outgoing (which it controls) and incoming
> (which it doesn't, or only controls very indirectly) to the recording
> server.
>
>
> For cases where you have bandwidth to burn (audio-only calls on broadband)
> you might get away with it. For a video call under any sort of bandwidth
> restraint, it would be a huge problem and would seriously degrade quality.

Re recognize this in siprec. The SRC can be inserted into the signaling 
and media path in a variety of ways. Typical sip deployments today have 
servers in the path that are often not bandwidth constrained.

RTCWEB seems to be revisiting the deployment topologies. Topologies that 
work for recording may also need to be revisited.

	Thanks,
	Paul

From pkyzivat@alum.mit.edu  Mon Aug 22 14:30:05 2011
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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On 8/22/11 3:01 PM, Jim McEachern wrote:
> Charles,
> Does this mean that there isn't a need to say anything, or that we should state there is a requirement to  indicate the communication is being recorded, but not assume you will have to insert something into the media path?

We've discussed this in siprec.
Its our understanding that in some cases the entity doing the recording 
has an obligation to ensure that the participants in the recorded 
session are aware of the recording. Playing out special tones, like 
beeps, is one way of achieving that, but not necessarily the only way. 
Rules vary, so we need to be flexible about this.

For siprec we are providing a way for participants in the session being 
recorded to signal that they are capable of recognizing an indication of 
recording in the signaling path. For UAs that don't so signal, the SRC 
will then have to provide indication in the media.

This seems to provide sufficient flexibility for the cases siprec is 
covering. It covers things like pstn gateways.

For RTCWEB the same concepts probably apply, but the assumptions and 
details may differ.

Also note that this isn't just about audio. If we are recording video, 
and not audio, then playing out beeps in the audio stream may not make 
much sense. (Consider a video + real time text call involving a deaf 
person.) Providing an indication in the signaling allows the UA to 
tailor an indication most suited to its user.

	Thanks,
	Paul

> Jim
>
>
> -----Original Message-----
> From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
> Sent: Monday, August 22, 2011 2:43 PM
> To: Jim McEachern; Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: [rtcweb] Proposed text for remote recording use case
>
> Hi Jim,
>
> SIPREC includes the notion of recording awareness, and in the case of a recording aware device you can signal in SDP the fact that something is being recorded rather than requiring an indication to be inserted into the media stream. I think RTCWEB should be able to leverage this mechanism. In that case, the indication of the media being recorded may be communicated by the web application or the browser, but it need not necessarily be inserted in the actual media.
>
> Cheers,
> Charles
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Jim McEachern
>> Sent: Monday, August 22, 2011 11:30 AM
>> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: Re: [rtcweb] Proposed text for remote recording use case
>>
>> John,
>> It has previously been pointed out that in some jurisdictions there is
> a legal requirement to insert a
>> tone (e.g. an intermittent beep) that indicates the conversation is
> being recorded.  It seems like
>> this would be a good thing to mention here.
>>
>> If this makes sense, I'm thinking something like the following
> sentence added to the end of 4.2.yy:
>> "... If required, the web application will direct the browser to
> insert an appropriate indication
>> (e.g. an intermediate beep) into the media stream to show that the
> communication is being recorded."
>>
>> This also suggests an additional requirement.
>> "Ayy2: The web application MUST be able to ask the browser to insert
> an appropriate indication into
>> the media stream to indicate that the communication is being
> recorded."
>>
>> I believe that F15 already covers the requirments for the browser in
> this case.
>> F15:         The browser MUST be able to process and mix
>>                     sound objects (media that is retrieved from another
>>                     source than the established media stream(s) with
> the
>>                     peer(s) with audio streams).
>>
>> Thoughts?
>> Jim
>>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Elwell, John
>> Sent: Monday, August 22, 2011 10:42 AM
>> To: rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: [rtcweb] Proposed text for remote recording use case
>>
>> 4.2.yy Remote Session Recording
>> In this use case, the web application user wishes to record a
> real-time communication at a remote
>> recording device, such that transmitted and received audio, video or
> other real-time media are
>> transmitted in real-time to the remote device. The remote device can
> perform archiving, retrieval,
>> playback, etc., but can also perform real-time analytics on the media.
> A typical deployment might be
>> in a contact centre. For a given medium, the two directions of
> transmission can be transmitted
>> together (mixed) or separately. The web application also sends
> metadata that gives context to the
>> stored media.
>>
>> New requirements:
>> Fyy1: The browser MUST be able to send in real-time to a remote
> recording device media that are being
>> transmitted to and received from remote participants.
>>
>> Ayy1: The web application MUST be able to ask the browser to transmit
> in real-time to a remote
>> recording device media that are being transmitted to and received from
> remote participants and, in the
>> case of audio at least, ask for the two directions of transmission to
> be transmitted to the remote
>> recording device mixed or separately.
>>
>> John
>>
>>
>> John Elwell
>> Tel: +44 1908 817801 (office and mobile)
>> Email: john.elwell@siemens-enterprise.com
>> http://www.siemens-enterprise.com/uk/
>>
>> Siemens Enterprise Communications Limited.
>> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
> 0DJ.
>> Registered No: 5903714, England.
>>
>> Siemens Enterprise Communications Limited is a Trademark Licensee of
> Siemens AG.
>> _______________________________________________
>> 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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From: "Charles Eckel (eckelcu)" <eckelcu@cisco.com>
To: "Jim McEachern" <jim.mceachern@genband.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>, <rtcweb@ietf.org>,  <public-webrtc@w3.org>
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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Hi Jim,

I met to recommend the latter.

Cheers,
Charles

> -----Original Message-----
> From: Jim McEachern [mailto:jim.mceachern@genband.com]
> Sent: Monday, August 22, 2011 12:02 PM
> To: Charles Eckel (eckelcu); Elwell, John; rtcweb@ietf.org;
public-webrtc@w3.org
> Subject: RE: [rtcweb] Proposed text for remote recording use case
>=20
> Charles,
> Does this mean that there isn't a need to say anything, or that we
should state there is a requirement
> to  indicate the communication is being recorded, but not assume you
will have to insert something
> into the media path?
>=20
> Jim
>=20
>=20
> -----Original Message-----
> From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
> Sent: Monday, August 22, 2011 2:43 PM
> To: Jim McEachern; Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: [rtcweb] Proposed text for remote recording use case
>=20
> Hi Jim,
>=20
> SIPREC includes the notion of recording awareness, and in the case of
a recording aware device you can
> signal in SDP the fact that something is being recorded rather than
requiring an indication to be
> inserted into the media stream. I think RTCWEB should be able to
leverage this mechanism. In that
> case, the indication of the media being recorded may be communicated
by the web application or the
> browser, but it need not necessarily be inserted in the actual media.
>=20
> Cheers,
> Charles
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Jim McEachern
> > Sent: Monday, August 22, 2011 11:30 AM
> > To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> > Subject: Re: [rtcweb] Proposed text for remote recording use case
> >
> > John,
> > It has previously been pointed out that in some jurisdictions there
is
> a legal requirement to insert a
> > tone (e.g. an intermittent beep) that indicates the conversation is
> being recorded.  It seems like
> > this would be a good thing to mention here.
> >
> > If this makes sense, I'm thinking something like the following
> sentence added to the end of 4.2.yy:
> > "... If required, the web application will direct the browser to
> insert an appropriate indication
> > (e.g. an intermediate beep) into the media stream to show that the
> communication is being recorded."
> >
> > This also suggests an additional requirement.
> > "Ayy2: The web application MUST be able to ask the browser to insert
> an appropriate indication into
> > the media stream to indicate that the communication is being
> recorded."
> >
> > I believe that F15 already covers the requirments for the browser in
> this case.
> > F15:         The browser MUST be able to process and mix
> >                    sound objects (media that is retrieved from
another
> >                    source than the established media stream(s) with
> the
> >                    peer(s) with audio streams).
> >
> > Thoughts?
> > Jim
> >
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Elwell, John
> > Sent: Monday, August 22, 2011 10:42 AM
> > To: rtcweb@ietf.org; public-webrtc@w3.org
> > Subject: [rtcweb] Proposed text for remote recording use case
> >
> > 4.2.yy Remote Session Recording
> > In this use case, the web application user wishes to record a
> real-time communication at a remote
> > recording device, such that transmitted and received audio, video or
> other real-time media are
> > transmitted in real-time to the remote device. The remote device can
> perform archiving, retrieval,
> > playback, etc., but can also perform real-time analytics on the
media.
> A typical deployment might be
> > in a contact centre. For a given medium, the two directions of
> transmission can be transmitted
> > together (mixed) or separately. The web application also sends
> metadata that gives context to the
> > stored media.
> >
> > New requirements:
> > Fyy1: The browser MUST be able to send in real-time to a remote
> recording device media that are being
> > transmitted to and received from remote participants.
> >
> > Ayy1: The web application MUST be able to ask the browser to
transmit
> in real-time to a remote
> > recording device media that are being transmitted to and received
from
> remote participants and, in the
> > case of audio at least, ask for the two directions of transmission
to
> be transmitted to the remote
> > recording device mixed or separately.
> >
> > John
> >
> >
> > John Elwell
> > Tel: +44 1908 817801 (office and mobile)
> > Email: john.elwell@siemens-enterprise.com
> > http://www.siemens-enterprise.com/uk/
> >
> > Siemens Enterprise Communications Limited.
> > Registered office: Brickhill Street, Willen Lake, Milton Keynes,
MK15
> 0DJ.
> > Registered No: 5903714, England.
> >
> > Siemens Enterprise Communications Limited is a Trademark Licensee of
> Siemens AG.
> > _______________________________________________
> > 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 john.elwell@siemens-enterprise.com  Tue Aug 23 00:49:19 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 23 Aug 2011 09:50:22 +0200
Thread-Topic: [rtcweb] Proposed text for local recording use case
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 22 August 2011 16:00
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Proposed text for local recording use case
>=20
> This is a good start. But I'd like to dig a little deeper into the=20
> intent here.
>=20
> The below says the *user* wishes to record, and the *browser* must be=20
> able to do it. But is that the only case of interest?
>=20
> ISTM that in a number of cases it will be the web application=20
> that wants=20
> the recording, even if there is an obligation to inform the=20
> user that it=20
> is happening. And the behavior of the application may be changed=20
> substantially if the recording cannot be made.
>=20
> (Consider a web app provided by a brokerage to its clients.)
>=20
> OTOH, maybe some of these cases are out of scope because the=20
> user+browser can't be sufficiently trusted, so that its=20
> necessary to do=20
> the recording from some secure server.
[JRE] I think there is a significant difference between local recording and=
 remote recording. If there is a policy, say in a call centre, to record ca=
lls, the recording device is most likely going to be central, not local to =
the user's device. So I think for local recording it is largely up to the u=
ser whether to record or not. But yes, it could be that the application at =
least suggests recording.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

From john.elwell@siemens-enterprise.com  Tue Aug 23 00:50:10 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Jim McEachern <jim.mceachern@genband.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Tue, 23 Aug 2011 09:51:08 +0200
Thread-Topic: Proposed text for remote recording use case
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Jim,

Good point. Unfortunately the nature of  tones/announcements will vary cons=
iderably between jurisdictions, so the JS would either need to provide an e=
xplicit description of what is to be inserted, or would need to provide a s=
ource. In many cases nothing at all would be needed. Often it is the case t=
hat all callers into a contact centre are played an initial announcement th=
at calls may be recorded, before being forwarded to the agent. So if an RTC=
-Web-based agent is acting as the recording client for remote recording, it=
 perhaps won't need to provide any tone/announcement.=20

John


> -----Original Message-----
> From: Jim McEachern [mailto:jim.mceachern@genband.com]=20
> Sent: 22 August 2011 19:30
> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> Subject: RE: Proposed text for remote recording use case
>=20
> John,
> It has previously been pointed out that in some jurisdictions=20
> there is a legal requirement to insert a tone (e.g. an=20
> intermittent beep) that indicates the conversation is being=20
> recorded.  It seems like this would be a good thing to mention here. =20
>=20
> If this makes sense, I'm thinking something like the=20
> following sentence added to the end of 4.2.yy:
> "... If required, the web application will direct the browser=20
> to insert an appropriate indication (e.g. an intermediate=20
> beep) into the media stream to show that the communication is=20
> being recorded."
>=20
> This also suggests an additional requirement. =20
> "Ayy2: The web application MUST be able to ask the browser to=20
> insert an appropriate indication into the media stream to=20
> indicate that the communication is being recorded."
>=20
> I believe that F15 already covers the requirments for the=20
> browser in this case.
> F15:         The browser MUST be able to process and mix
>                    sound objects (media that is retrieved from another
>                    source than the established media=20
> stream(s) with the
>                    peer(s) with audio streams).
>=20
> Thoughts?
> Jim
>=20
> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Elwell, John
> Sent: Monday, August 22, 2011 10:42 AM
> To: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: [rtcweb] Proposed text for remote recording use case
>=20
> 4.2.yy Remote Session Recording
> In this use case, the web application user wishes to record a=20
> real-time communication at a remote recording device, such=20
> that transmitted and received audio, video or other real-time=20
> media are transmitted in real-time to the remote device. The=20
> remote device can perform archiving, retrieval, playback,=20
> etc., but can also perform real-time analytics on the media.=20
> A typical deployment might be in a contact centre. For a=20
> given medium, the two directions of transmission can be=20
> transmitted together (mixed) or separately. The web=20
> application also sends metadata that gives context to the=20
> stored media.
>=20
> New requirements:
> Fyy1: The browser MUST be able to send in real-time to a=20
> remote recording device media that are being transmitted to=20
> and received from remote participants.
>=20
> Ayy1: The web application MUST be able to ask the browser to=20
> transmit in real-time to a remote recording device media that=20
> are being transmitted to and received from remote=20
> participants and, in the case of audio at least, ask for the=20
> two directions of transmission to be transmitted to the=20
> remote recording device mixed or separately.
>=20
> John
>=20
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From john.elwell@siemens-enterprise.com  Tue Aug 23 00:51:07 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 23 Aug 2011 09:52:11 +0200
Thread-Topic: [rtcweb] Proposed text for remote recording use case
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: 22 August 2011 22:31
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Proposed text for remote recording use case
>=20
> On 8/22/11 3:01 PM, Jim McEachern wrote:
> > Charles,
> > Does this mean that there isn't a need to say anything, or=20
> that we should state there is a requirement to  indicate the=20
> communication is being recorded, but not assume you will have=20
> to insert something into the media path?
>=20
> We've discussed this in siprec.
> Its our understanding that in some cases the entity doing the=20
> recording=20
> has an obligation to ensure that the participants in the recorded=20
> session are aware of the recording. Playing out special tones, like=20
> beeps, is one way of achieving that, but not necessarily the=20
> only way.=20
> Rules vary, so we need to be flexible about this.
>=20
> For siprec we are providing a way for participants in the=20
> session being=20
> recorded to signal that they are capable of recognizing an=20
> indication of=20
> recording in the signaling path. For UAs that don't so=20
> signal, the SRC=20
> will then have to provide indication in the media.
[JRE] A caveat here. The fact that a participant's UA can signal that it ca=
n recognize an indication of recording and pass on some sort of indication =
to the participant should not necessarily be seen as removing the need for =
the SRC to insert a tone/announcement (or equivalent in other media). The s=
imple indication to the UA that recording is in progress for a given medium=
 might not convey the full meaning that can be conveyed by a country-specif=
ic or enterprise-specific announcement. So in many deployments I anticipate=
 the SRC providing in-band indications anyway, but allowing a recording-awa=
re UA to augment these with additional indications specific to its user int=
erface, e.g., light a lamp, flash text, etc..

John

>=20
> This seems to provide sufficient flexibility for the cases siprec is=20
> covering. It covers things like pstn gateways.
>=20
> For RTCWEB the same concepts probably apply, but the assumptions and=20
> details may differ.
>=20
> Also note that this isn't just about audio. If we are=20
> recording video,=20
> and not audio, then playing out beeps in the audio stream may=20
> not make=20
> much sense. (Consider a video + real time text call involving a deaf=20
> person.) Providing an indication in the signaling allows the UA to=20
> tailor an indication most suited to its user.
>=20
> 	Thanks,
> 	Paul
>=20
> > Jim
> >
> >
> > -----Original Message-----
> > From: Charles Eckel (eckelcu) [mailto:eckelcu@cisco.com]
> > Sent: Monday, August 22, 2011 2:43 PM
> > To: Jim McEachern; Elwell, John; rtcweb@ietf.org;=20
> public-webrtc@w3.org
> > Subject: RE: [rtcweb] Proposed text for remote recording use case
> >
> > Hi Jim,
> >
> > SIPREC includes the notion of recording awareness, and in=20
> the case of a recording aware device you can signal in SDP=20
> the fact that something is being recorded rather than=20
> requiring an indication to be inserted into the media stream.=20
> I think RTCWEB should be able to leverage this mechanism. In=20
> that case, the indication of the media being recorded may be=20
> communicated by the web application or the browser, but it=20
> need not necessarily be inserted in the actual media.
> >
> > Cheers,
> > Charles
> >
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Jim McEachern
> >> Sent: Monday, August 22, 2011 11:30 AM
> >> To: Elwell, John; rtcweb@ietf.org; public-webrtc@w3.org
> >> Subject: Re: [rtcweb] Proposed text for remote recording use case
> >>
> >> John,
> >> It has previously been pointed out that in some=20
> jurisdictions there is
> > a legal requirement to insert a
> >> tone (e.g. an intermittent beep) that indicates the conversation is
> > being recorded.  It seems like
> >> this would be a good thing to mention here.
> >>
> >> If this makes sense, I'm thinking something like the following
> > sentence added to the end of 4.2.yy:
> >> "... If required, the web application will direct the browser to
> > insert an appropriate indication
> >> (e.g. an intermediate beep) into the media stream to show that the
> > communication is being recorded."
> >>
> >> This also suggests an additional requirement.
> >> "Ayy2: The web application MUST be able to ask the browser=20
> to insert
> > an appropriate indication into
> >> the media stream to indicate that the communication is being
> > recorded."
> >>
> >> I believe that F15 already covers the requirments for the=20
> browser in
> > this case.
> >> F15:         The browser MUST be able to process and mix
> >>                     sound objects (media that is retrieved=20
> from another
> >>                     source than the established media=20
> stream(s) with
> > the
> >>                     peer(s) with audio streams).
> >>
> >> Thoughts?
> >> Jim
> >>
> >> -----Original Message-----
> >> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> > Behalf Of Elwell, John
> >> Sent: Monday, August 22, 2011 10:42 AM
> >> To: rtcweb@ietf.org; public-webrtc@w3.org
> >> Subject: [rtcweb] Proposed text for remote recording use case
> >>
> >> 4.2.yy Remote Session Recording
> >> In this use case, the web application user wishes to record a
> > real-time communication at a remote
> >> recording device, such that transmitted and received=20
> audio, video or
> > other real-time media are
> >> transmitted in real-time to the remote device. The remote=20
> device can
> > perform archiving, retrieval,
> >> playback, etc., but can also perform real-time analytics=20
> on the media.
> > A typical deployment might be
> >> in a contact centre. For a given medium, the two directions of
> > transmission can be transmitted
> >> together (mixed) or separately. The web application also sends
> > metadata that gives context to the
> >> stored media.
> >>
> >> New requirements:
> >> Fyy1: The browser MUST be able to send in real-time to a remote
> > recording device media that are being
> >> transmitted to and received from remote participants.
> >>
> >> Ayy1: The web application MUST be able to ask the browser=20
> to transmit
> > in real-time to a remote
> >> recording device media that are being transmitted to and=20
> received from
> > remote participants and, in the
> >> case of audio at least, ask for the two directions of=20
> transmission to
> > be transmitted to the remote
> >> recording device mixed or separately.
> >>
> >> John
> >>
> >>
> >> John Elwell
> >> Tel: +44 1908 817801 (office and mobile)
> >> Email: john.elwell@siemens-enterprise.com
> >> http://www.siemens-enterprise.com/uk/
> >>
> >> Siemens Enterprise Communications Limited.
> >> Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15
> > 0DJ.
> >> Registered No: 5903714, England.
> >>
> >> Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of
> > Siemens AG.
> >> _______________________________________________
> >> 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
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From john.elwell@siemens-enterprise.com  Tue Aug 23 00:57:25 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Date: Tue, 23 Aug 2011 09:58:30 +0200
Thread-Topic: Remote recording - RTC-Web client acting as SIPREC session recording client
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Subject: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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There has been some discussion recently on remote recording, mixed to some =
extent with discussions on local recording and with mailbox, but I would li=
ke to focus on remote recording and try to summarize.

First, some background on the IETF SIPREC WG. This is specifying support fo=
r SIP-based session recording, whereby a Session Recording Client (SRC) on =
the path of a call (communication session) can forward media and metadata t=
o a session recording server (SRS) or recording device. In conventional SIP=
 terms, the SRC can exist at an endpoint of the communication session being=
 recorded (i.e., at a SIP UA), or at a B2BUA that has access to the media a=
s well as the signalling. Very often in a contact centre, there are mandato=
ry requirements for recording some or all communication sessions, and often=
 calls are routed through a B2BUA that also provides the SRC. So in this ca=
se there is no responsibility on SIP UAs to support SRC functionality, and =
no issues of additional bandwidth on the device's access. However, it is an=
ticipated in SIPREC that in some deployments UA-located SRCs will be used. =
How a UA is organized internally to provide SRC functionality is not standa=
rdized.

So the question for RTC-Web is whether a SIP UA implemented as an RTC-Web c=
lient can provide SRC functionality, i.e., support remote recording. An RTC=
-Web SIP UA is implemented by a combination of functionality running on the=
 web server, functionality running in client side script (JS) and functiona=
lity embedded in the browser. The amount of functionality needed in the bro=
wser and needing to be exposed at the browser API in support of SRC will de=
pend to some extent on how much core functionality goes into the browser, i=
n particular whether the browser implements SIP or not. Some of the functio=
ns noted to date include:
- ability to take a copy of streams sent to / received from the remote part=
y and send them, in real-time, to a remote recording device (SRS);
- possible need to mix the copied streams before sending (e.g., mix the sen=
t and received audio streams)
- possible need to use a different codec or other parameters when sending t=
o the SRS;
- possible need to use a different encryption/integrity context when sendin=
g to the SRS;
- possible need to insert tones / announcements into the original media pat=
h being recorded;
- possible need to support SDP enhancements for indicating media that are b=
eing recorded or preferences for which media are being recorded;
- possible need to support SIP enhancements for indicating SRC/SRS capabili=
ty and recording awareness (if SIP is implemented in browser);
- possible need to support the sending of metadata to the SRS (if SIP is im=
plemented in browser).

Clearly there would be a fairly big hurdle for browsers to support SRC func=
tionality. But without this, RTC-Web clients would not be suitable for use =
in environments where remote recording is required and calls are not forced=
 through some middlebox that provides SRC functionality.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
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s AG.

From stefan.lk.hakansson@ericsson.com  Tue Aug 23 01:43:47 2011
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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Hm.

I must admit that my instinctive reaction when reading this is:

1. Say that SRC and SRS functionality is out of scope for rtcweb/webrtc 
version 1.
2. If someone wants to support this, force media through a middlebox 
(which has all media and can do stuff like inserting beeps, mixing, 
recording, ...).

I'm sure there are other opinions....

Cheers,
Stefan

On 2011-08-23 09:58, Elwell, John wrote:
> There has been some discussion recently on remote recording, mixed to some extent with discussions on local recording and with mailbox, but I would like to focus on remote recording and try to summarize.
>
> First, some background on the IETF SIPREC WG. This is specifying support for SIP-based session recording, whereby a Session Recording Client (SRC) on the path of a call (communication session) can forward media and metadata to a session recording server (SRS) or recording device. In conventional SIP terms, the SRC can exist at an endpoint of the communication session being recorded (i.e., at a SIP UA), or at a B2BUA that has access to the media as well as the signalling. Very often in a contact centre, there are mandatory requirements for recording some or all communication sessions, and often calls are routed through a B2BUA that also provides the SRC. So in this case there is no responsibility on SIP UAs to support SRC functionality, and no issues of additional bandwidth on the device's access. However, it is anticipated in SIPREC that in some deployments UA-located SRCs will be used. How a UA is organized internally to provide SRC functionality is not standardized.
>
> So the question for RTC-Web is whether a SIP UA implemented as an RTC-Web client can provide SRC functionality, i.e., support remote recording. An RTC-Web SIP UA is implemented by a combination of functionality running on the web server, functionality running in client side script (JS) and functionality embedded in the browser. The amount of functionality needed in the browser and needing to be exposed at the browser API in support of SRC will depend to some extent on how much core functionality goes into the browser, in particular whether the browser implements SIP or not. Some of the functions noted to date include:
> - ability to take a copy of streams sent to / received from the remote party and send them, in real-time, to a remote recording device (SRS);
> - possible need to mix the copied streams before sending (e.g., mix the sent and received audio streams)
> - possible need to use a different codec or other parameters when sending to the SRS;
> - possible need to use a different encryption/integrity context when sending to the SRS;
> - possible need to insert tones / announcements into the original media path being recorded;
> - possible need to support SDP enhancements for indicating media that are being recorded or preferences for which media are being recorded;
> - possible need to support SIP enhancements for indicating SRC/SRS capability and recording awareness (if SIP is implemented in browser);
> - possible need to support the sending of metadata to the SRS (if SIP is implemented in browser).
>
> Clearly there would be a fairly big hurdle for browsers to support SRC functionality. But without this, RTC-Web clients would not be suitable for use in environments where remote recording is required and calls are not forced through some middlebox that provides SRC functionality.
>
> John
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From pravindran@sonusnet.com  Tue Aug 23 02:34:05 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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Stefan,

As you expect, I have disagreement for not including this usecase in =
RTCweb 1.0. My reasoning are as follows:

1) As John pointed out and other folks responded, remote recording has =
importance in few segments like call centre deployment.

2) RTCweb1.0 solution without considering this usecase may end-up in a =
non-scalable RTCWeb remote recording solution for ever or break =
RTCWeb1.0 to meet this architectural need of remote recording.=20

3) Based on SIPREC discussion, it is very apparent that short-sighted =
SIP Endpoint implementation which assumes single session for the =
end-point are forced to interop with SRS using middle box only . I wish =
that the same situation does not occur for rtcweb clients.

Thanks
Partha=20

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On =
Behalf
>Of Stefan H=E5kansson LK
>Sent: Tuesday, August 23, 2011 2:15 PM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as =
SIPREC
>session recording client
>
>Hm.
>
>I must admit that my instinctive reaction when reading this is:
>
>1. Say that SRC and SRS functionality is out of scope for rtcweb/webrtc
>version 1.
>2. If someone wants to support this, force media through a middlebox
>(which has all media and can do stuff like inserting beeps, mixing,
>recording, ...).
>
>I'm sure there are other opinions....
>
>Cheers,
>Stefan
>
>On 2011-08-23 09:58, Elwell, John wrote:
>> There has been some discussion recently on remote recording, mixed to
>some extent with discussions on local recording and with mailbox, but I
>would like to focus on remote recording and try to summarize.
>>
>> First, some background on the IETF SIPREC WG. This is specifying
>support for SIP-based session recording, whereby a Session Recording
>Client (SRC) on the path of a call (communication session) can forward
>media and metadata to a session recording server (SRS) or recording
>device. In conventional SIP terms, the SRC can exist at an endpoint of
>the communication session being recorded (i.e., at a SIP UA), or at a
>B2BUA that has access to the media as well as the signalling. Very =
often
>in a contact centre, there are mandatory requirements for recording =
some
>or all communication sessions, and often calls are routed through a
>B2BUA that also provides the SRC. So in this case there is no
>responsibility on SIP UAs to support SRC functionality, and no issues =
of
>additional bandwidth on the device's access. However, it is anticipated
>in SIPREC that in some deployments UA-located SRCs will be used. How a
>UA is organized internally to provide SRC functionality is not
>standardized.
>>
>> So the question for RTC-Web is whether a SIP UA implemented as an =
RTC-
>Web client can provide SRC functionality, i.e., support remote
>recording. An RTC-Web SIP UA is implemented by a combination of
>functionality running on the web server, functionality running in =
client
>side script (JS) and functionality embedded in the browser. The amount
>of functionality needed in the browser and needing to be exposed at the
>browser API in support of SRC will depend to some extent on how much
>core functionality goes into the browser, in particular whether the
>browser implements SIP or not. Some of the functions noted to date
>include:
>> - ability to take a copy of streams sent to / received from the =
remote
>party and send them, in real-time, to a remote recording device (SRS);
>> - possible need to mix the copied streams before sending (e.g., mix
>the sent and received audio streams)
>> - possible need to use a different codec or other parameters when
>sending to the SRS;
>> - possible need to use a different encryption/integrity context when
>sending to the SRS;
>> - possible need to insert tones / announcements into the original
>media path being recorded;
>> - possible need to support SDP enhancements for indicating media that
>are being recorded or preferences for which media are being recorded;
>> - possible need to support SIP enhancements for indicating SRC/SRS
>capability and recording awareness (if SIP is implemented in browser);
>> - possible need to support the sending of metadata to the SRS (if SIP
>is implemented in browser).
>>
>> Clearly there would be a fairly big hurdle for browsers to support =
SRC
>functionality. But without this, RTC-Web clients would not be suitable
>for use in environments where remote recording is required and calls =
are
>not forced through some middlebox that provides SRC functionality.
>>
>> John
>>
>> John Elwell
>> Tel: +44 1908 817801 (office and mobile)
>> Email: john.elwell@siemens-enterprise.com
>> http://www.siemens-enterprise.com/uk/
>>
>> Siemens Enterprise Communications Limited.
>> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
>0DJ.
>> Registered No: 5903714, England.
>>
>> Siemens Enterprise Communications Limited is a Trademark Licensee of
>Siemens AG.
>> _______________________________________________
>> 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 pkyzivat@alum.mit.edu  Tue Aug 23 07:02:55 2011
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Subject: Re: [rtcweb] Proposed text for local recording use case
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On 8/23/11 3:50 AM, Elwell, John wrote:
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
>> Sent: 22 August 2011 16:00
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Proposed text for local recording use case
>>
>> This is a good start. But I'd like to dig a little deeper into the
>> intent here.
>>
>> The below says the *user* wishes to record, and the *browser* must be
>> able to do it. But is that the only case of interest?
>>
>> ISTM that in a number of cases it will be the web application
>> that wants
>> the recording, even if there is an obligation to inform the
>> user that it
>> is happening. And the behavior of the application may be changed
>> substantially if the recording cannot be made.
>>
>> (Consider a web app provided by a brokerage to its clients.)
>>
>> OTOH, maybe some of these cases are out of scope because the
>> user+browser can't be sufficiently trusted, so that its
>> necessary to do
>> the recording from some secure server.
> [JRE] I think there is a significant difference between local recording and remote recording. If there is a policy, say in a call centre, to record calls, the recording device is most likely going to be central, not local to the user's device. So I think for local recording it is largely up to the user whether to record or not. But yes, it could be that the application at least suggests recording.

I agree there is a significant difference between local and remote 
recording.

But in the RTCWEB context there are a number of alternatives for local 
recording. Specifically, how is the decision made to record locally?

- the user could have configured the browser to unconditionally
   record every media stream, without regard to what the JavaScript
   has to say about it. This might be an important policy choice if
   you don't trust what the web sites do

- the JS downloaded from a web server may come with instructions
   to make a local recording.

- the JS downloaded from a web server may contain code that checks
   local policy in the browser and/or queries the user, to decide
   whether to make a local recording

In the latter two of the above the decision to record or not rests with 
the web server. In the first case it does not. The first case also 
requires a different sort of implementation in the browser - it must be 
hooked into the stream processing machinery at a lower level.

I think its necessary to decide whether the first case needs to be 
supported. Its tricky, because recording everything is probably 
excessive, yet I'm not sure I'm ready to trust that all the web servers 
I want to use will do the right thing.

Is there a middle ground?

	Thanks,
	Paul

> John
>
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
>


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Thanks John. You did a much more thorough job of explaining than I did.

	Paul

On 8/23/11 3:58 AM, Elwell, John wrote:
> There has been some discussion recently on remote recording, mixed to some extent with discussions on local recording and with mailbox, but I would like to focus on remote recording and try to summarize.
>
> First, some background on the IETF SIPREC WG. This is specifying support for SIP-based session recording, whereby a Session Recording Client (SRC) on the path of a call (communication session) can forward media and metadata to a session recording server (SRS) or recording device. In conventional SIP terms, the SRC can exist at an endpoint of the communication session being recorded (i.e., at a SIP UA), or at a B2BUA that has access to the media as well as the signalling. Very often in a contact centre, there are mandatory requirements for recording some or all communication sessions, and often calls are routed through a B2BUA that also provides the SRC. So in this case there is no responsibility on SIP UAs to support SRC functionality, and no issues of additional bandwidth on the device's access. However, it is anticipated in SIPREC that in some deployments UA-located SRCs will be used. How a UA is organized internally to provide SRC functionality is not standardized.
>
> So the question for RTC-Web is whether a SIP UA implemented as an RTC-Web client can provide SRC functionality, i.e., support remote recording. An RTC-Web SIP UA is implemented by a combination of functionality running on the web server, functionality running in client side script (JS) and functionality embedded in the browser. The amount of functionality needed in the browser and needing to be exposed at the browser API in support of SRC will depend to some extent on how much core functionality goes into the browser, in particular whether the browser implements SIP or not. Some of the functions noted to date include:
> - ability to take a copy of streams sent to / received from the remote party and send them, in real-time, to a remote recording device (SRS);
> - possible need to mix the copied streams before sending (e.g., mix the sent and received audio streams)
> - possible need to use a different codec or other parameters when sending to the SRS;
> - possible need to use a different encryption/integrity context when sending to the SRS;
> - possible need to insert tones / announcements into the original media path being recorded;
> - possible need to support SDP enhancements for indicating media that are being recorded or preferences for which media are being recorded;
> - possible need to support SIP enhancements for indicating SRC/SRS capability and recording awareness (if SIP is implemented in browser);
> - possible need to support the sending of metadata to the SRS (if SIP is implemented in browser).
>
> Clearly there would be a fairly big hurdle for browsers to support SRC functionality. But without this, RTC-Web clients would not be suitable for use in environments where remote recording is required and calls are not forced through some middlebox that provides SRC functionality.
>
> John
>
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
> Registered No: 5903714, England.
>
> Siemens Enterprise Communications Limited is a Trademark Licensee of Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Subject: Re: [rtcweb] Proposed text for local recording use case
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Comments inline...

On Aug 23, 2011, at 10:03 AM, Paul Kyzivat wrote:

> On 8/23/11 3:50 AM, Elwell, John wrote:
>>> =20
>>> OTOH, maybe some of these cases are out of scope because the
>>> user+browser can't be sufficiently trusted, so that its
>>> necessary to do
>>> the recording from some secure server.
>> [JRE] I think there is a significant difference between local =
recording and remote recording. If there is a policy, say in a call =
centre, to record calls, the recording device is most likely going to be =
central, not local to the user's device. So I think for local recording =
it is largely up to the user whether to record or not. But yes, it could =
be that the application at least suggests recording.
>=20
> I agree there is a significant difference between local and remote =
recording.
>=20
> But in the RTCWEB context there are a number of alternatives for local =
recording.

DY>  I would also argue that there are any number of alternatives for =
local recording completely *outside* of what we are talking about for =
RTCWEB.  =20

DY> For instance, one the of the use cases *I* have personally for the =
RTCWEB work is to do audio (and potentially video) interviews with =
people and record them for various podcasts to which I contribute.  =
Typically I use Skype today for that, but I'd love an alternative where =
I could simply give someone a URL where they could go and click a button =
to connect to me and start the interview.  I can do this today, of =
course, using things like http://phono.com/ , but it does require the =
person calling me to use Flash, Java, etc. to do the connection from =
their browser to their audio devices.   I want to be free of all of that =
and just have it work in the browser.

DY> As I've been reading this whole thread on local/remote recording, I =
was thinking how I would use this in my use case... and the reality is =
that I probably *wouldn't* use it.  I already have additional software =
on my system that records my local computer audio and indeed can record =
my entire screen in a video format (which allows me to record video =
sessions).  I can then do all the relevant post-production I need to =
create the audio or video files I need.

DY> In fact, I have multiple solutions for local recording on my =
computer... some are commercial products I have purchased, some are open =
source.  Now true, they are not synced to an incoming call, so I would =
need to trigger them to initiate the recording, but they are available.

DY> All of this made me wonder if perhaps we are diving in a bit too =
deep here...  while it might be ideal for recording to be triggered in =
an RTCWEB connection, is that truly the vast majority of cases?  Or are =
we debating something that is more of an edge case?   In a call center =
environment, for instance, might the agents not already have other =
software installed on their computer that could be doing the local =
recording if required?

DY> I'm concerned in reading these threads that we are adding more and =
more layers of complexity which could get in the way of spurring =
widespread adoption.  (And yes, I do fully understand the =
counter-argument that if we don't think about this in advance we may not =
be able to add the functionality later.)

My 2 cents,
Dan

--=20
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Comments inline...<div><br><div><div>On Aug 23, 2011, at 10:03 AM, =
Paul Kyzivat wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>On =
8/23/11 3:50 AM, Elwell, John wrote:<br><blockquote =
type=3D"cite"><blockquote type=3D"cite"><font class=3D"Apple-style-span" =
color=3D"#006312">&nbsp;</font></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">OTOH, maybe some of these cases =
are out of scope because the<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">user+browser can't be =
sufficiently trusted, so that =
its<br></blockquote></blockquote><blockquote type=3D"cite"><blockquote =
type=3D"cite">necessary to do<br></blockquote></blockquote><blockquote =
type=3D"cite"><blockquote type=3D"cite">the recording from some secure =
server.<br></blockquote></blockquote><blockquote type=3D"cite">[JRE] I =
think there is a significant difference between local recording and =
remote recording. If there is a policy, say in a call centre, to record =
calls, the recording device is most likely going to be central, not =
local to the user's device. So I think for local recording it is largely =
up to the user whether to record or not. But yes, it could be that the =
application at least suggests recording.<br></blockquote><br>I agree =
there is a significant difference between local and remote =
recording.<br><br>But in the RTCWEB context there are a number of =
alternatives for local recording. =
</div></blockquote><div><br></div><div>DY&gt; &nbsp;I would also argue =
that there are any number of alternatives for local recording completely =
*outside* of what we are talking about for RTCWEB. =
&nbsp;&nbsp;</div><div><br></div><div>DY&gt; For instance, one the of =
the use cases *I* have personally for the RTCWEB work is to do audio =
(and potentially video) interviews with people and record them for =
various podcasts to which I contribute. &nbsp;Typically I use Skype =
today for that, but I'd love an alternative where I could simply give =
someone a URL where they could go and click a button to connect to me =
and start the interview. &nbsp;I can do this today, of course, using =
things like <a href=3D"http://phono.com/">http://phono.com/</a> , but it =
does require the person calling me to use Flash, Java, etc. to do the =
connection from their browser to their audio devices. &nbsp; I want to =
be free of all of that and just have it work in the =
browser.</div><div><br></div><div>DY&gt; As I've been reading this whole =
thread on local/remote recording, I was thinking how I would use this in =
my use case... and the reality is that I probably *wouldn't* use it. =
&nbsp;I already have additional software on my system that records my =
local computer audio and indeed can record my entire screen in a video =
format (which allows me to record video sessions). &nbsp;I can then do =
all the relevant post-production I need to create the audio or video =
files I need.</div><div><br></div><div>DY&gt; In fact, I have multiple =
solutions for local recording on my computer... some are commercial =
products I have purchased, some are open source. &nbsp;Now true, they =
are not synced to an incoming call, so I would need to trigger them to =
initiate the recording, but they are =
available.</div><br></div><div>DY&gt; All of this made me wonder if =
perhaps we are diving in a bit too deep here... &nbsp;while it might be =
ideal for recording to be triggered in an RTCWEB connection, is that =
truly the vast majority of cases? &nbsp;Or are we debating something =
that is more of an edge case? &nbsp; In a call center environment, for =
instance, might the agents not already have other software installed on =
their computer that could be doing the local recording if =
required?</div><div><br></div><div>DY&gt; I'm concerned in reading these =
threads that we are adding more and more layers of complexity which =
could get in the way of spurring widespread adoption. &nbsp;(And yes, I =
do fully understand the counter-argument that if we don't think about =
this in advance we may not be able to add the functionality =
later.)</div><div><br></div>My 2 =
cents,</div><div>Dan</div><div><br><div>
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after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
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auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
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-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
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medium; "><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; =
">--&nbsp;</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Dan York, CISSP, Director of =
Conversations</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; ">Voxeo =
Corporation&nbsp;&nbsp;&nbsp;<a =
href=3D"http://www.voxeo.com">http://www.voxeo.com</a>&nbsp;&nbsp;<a =
href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Phone: +1-321-710-9193 &nbsp;skype: =
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From pkyzivat@alum.mit.edu  Tue Aug 23 08:21:58 2011
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Proposed text for local recording use case
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IMO the key here is that we are thinking it through.
It may turn out that there are no new requirements on RTCWEB.
But then that decision will have been made consciously, rather than by 
default.

I'm all for allowing independent things to be dealt with independently, 
rather than including everything in one big mash up. But finding where 
to draw the lines between things isn't always easy.

	Thanks,
	Paul

On 8/23/11 10:37 AM, Dan York wrote:
> Comments inline...
>
> On Aug 23, 2011, at 10:03 AM, Paul Kyzivat wrote:
>
>> On 8/23/11 3:50 AM, Elwell, John wrote:
>>>> OTOH, maybe some of these cases are out of scope because the
>>>> user+browser can't be sufficiently trusted, so that its
>>>> necessary to do
>>>> the recording from some secure server.
>>> [JRE] I think there is a significant difference between local
>>> recording and remote recording. If there is a policy, say in a call
>>> centre, to record calls, the recording device is most likely going to
>>> be central, not local to the user's device. So I think for local
>>> recording it is largely up to the user whether to record or not. But
>>> yes, it could be that the application at least suggests recording.
>>
>> I agree there is a significant difference between local and remote
>> recording.
>>
>> But in the RTCWEB context there are a number of alternatives for local
>> recording.
>
> DY> I would also argue that there are any number of alternatives for
> local recording completely *outside* of what we are talking about for
> RTCWEB.
>
> DY> For instance, one the of the use cases *I* have personally for the
> RTCWEB work is to do audio (and potentially video) interviews with
> people and record them for various podcasts to which I contribute.
> Typically I use Skype today for that, but I'd love an alternative where
> I could simply give someone a URL where they could go and click a button
> to connect to me and start the interview. I can do this today, of
> course, using things like http://phono.com/ , but it does require the
> person calling me to use Flash, Java, etc. to do the connection from
> their browser to their audio devices. I want to be free of all of that
> and just have it work in the browser.
>
> DY> As I've been reading this whole thread on local/remote recording, I
> was thinking how I would use this in my use case... and the reality is
> that I probably *wouldn't* use it. I already have additional software on
> my system that records my local computer audio and indeed can record my
> entire screen in a video format (which allows me to record video
> sessions). I can then do all the relevant post-production I need to
> create the audio or video files I need.
>
> DY> In fact, I have multiple solutions for local recording on my
> computer... some are commercial products I have purchased, some are open
> source. Now true, they are not synced to an incoming call, so I would
> need to trigger them to initiate the recording, but they are available.
>
> DY> All of this made me wonder if perhaps we are diving in a bit too
> deep here... while it might be ideal for recording to be triggered in an
> RTCWEB connection, is that truly the vast majority of cases? Or are we
> debating something that is more of an edge case? In a call center
> environment, for instance, might the agents not already have other
> software installed on their computer that could be doing the local
> recording if required?
>
> DY> I'm concerned in reading these threads that we are adding more and
> more layers of complexity which could get in the way of spurring
> widespread adoption. (And yes, I do fully understand the
> counter-argument that if we don't think about this in advance we may not
> be able to add the functionality later.)
>
> My 2 cents,
> Dan
>
> --
> Dan York, CISSP, Director of Conversations
> Voxeo Corporation http://www.voxeo.com dyork@voxeo.com
> <mailto:dyork@voxeo.com>
> Phone: +1-321-710-9193 skype: danyork sip:dyork@voxeo.com
>
> *Join the Voxeo conversation:*
> *Blogs: http://blogs.voxeo.com*
> *Twitter: http://twitter.com/voxeo http://twitter.com/danyork*
> *Facebook: http://www.facebook.com/voxeo*
>


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John,

Many thanks for the SIPREC background. I have not had the cycles to =
follow that group and so your note here is extremely helpful. Thank you!

As I read your summary though, I keep hearing "Danger! Danger!" in my =
brain. I completely agree with this statement of yours:

> Clearly there would be a fairly big hurdle for browsers to support SRC =
functionality.=20


I think it would be a very large hurdle and really takes us back into =
basically baking a SIP UA into a web browser - and do we REALLY want to =
go down that road?

If I go back to the charter of this group ( =
http://tools.ietf.org/wg/rtcweb/charters ) and why I started following =
it from the start and reading all the traffic, I think we need to focus =
on how we enable direct communication between web browsers... or between =
web browsers and servers... but doing so in the most lightweight and =
easy way possible. =20

My interest has been in how we can do real-time communication *without* =
extensions or plugins.  Reading all of this, it's sounding like either =
we do need to have a plugin/extension to support SRC capability - or we =
need to bake a great amount of functionality directly into browsers.  =
And that in my mind limits the number of browsers that might support all =
the capabilities of RTCWEB. =20

I think that given the aggressive timelines for the working group =
deliverables (and the market reality that the longer a "standard" =
solution takes to come out the more developers will explore proprietary =
solutions), I agree with Stefan that we should put the recording out of =
scope for RTCWEB 1.0 and focus on getting a solution out there that lets =
developers start building RTCWEB apps.=20

For those environments that need recording (and I *do* understand the =
call center need), middleboxes can provide a solution today - or some =
vendors can support the RTCWEB communication and *also* provide a =
recording capability. Sure, that's not ideal... and yes, we need to make =
sure that what we do for RTCWEB 1.0 doesn't preclude adding recording to =
a RTCWEB 2.0...  but we need to get a spec out there that will be useful =
to the majority of developers and very easy for them to adopt. =20

The more complicated we make it - or the more requirements we impose on =
browsers - the less adoption we'll see.

My 2 cents,
Dan


On Aug 23, 2011, at 3:58 AM, Elwell, John wrote:

> There has been some discussion recently on remote recording, mixed to =
some extent with discussions on local recording and with mailbox, but I =
would like to focus on remote recording and try to summarize.
>=20
> First, some background on the IETF SIPREC WG. This is specifying =
support for SIP-based session recording, whereby a Session Recording =
Client (SRC) on the path of a call (communication session) can forward =
media and metadata to a session recording server (SRS) or recording =
device. In conventional SIP terms, the SRC can exist at an endpoint of =
the communication session being recorded (i.e., at a SIP UA), or at a =
B2BUA that has access to the media as well as the signalling. Very often =
in a contact centre, there are mandatory requirements for recording some =
or all communication sessions, and often calls are routed through a =
B2BUA that also provides the SRC. So in this case there is no =
responsibility on SIP UAs to support SRC functionality, and no issues of =
additional bandwidth on the device's access. However, it is anticipated =
in SIPREC that in some deployments UA-located SRCs will be used. How a =
UA is organized internally to provide SRC functionality is not =
standardized.
>=20
> So the question for RTC-Web is whether a SIP UA implemented as an =
RTC-Web client can provide SRC functionality, i.e., support remote =
recording. An RTC-Web SIP UA is implemented by a combination of =
functionality running on the web server, functionality running in client =
side script (JS) and functionality embedded in the browser. The amount =
of functionality needed in the browser and needing to be exposed at the =
browser API in support of SRC will depend to some extent on how much =
core functionality goes into the browser, in particular whether the =
browser implements SIP or not. Some of the functions noted to date =
include:
> - ability to take a copy of streams sent to / received from the remote =
party and send them, in real-time, to a remote recording device (SRS);
> - possible need to mix the copied streams before sending (e.g., mix =
the sent and received audio streams)
> - possible need to use a different codec or other parameters when =
sending to the SRS;
> - possible need to use a different encryption/integrity context when =
sending to the SRS;
> - possible need to insert tones / announcements into the original =
media path being recorded;
> - possible need to support SDP enhancements for indicating media that =
are being recorded or preferences for which media are being recorded;
> - possible need to support SIP enhancements for indicating SRC/SRS =
capability and recording awareness (if SIP is implemented in browser);
> - possible need to support the sending of metadata to the SRS (if SIP =
is implemented in browser).
>=20
> Clearly there would be a fairly big hurdle for browsers to support SRC =
functionality. But without this, RTC-Web clients would not be suitable =
for use in environments where remote recording is required and calls are =
not forced through some middlebox that provides SRC functionality.
>=20
> John
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 =
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of =
Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--=20
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo


--Apple-Mail-12-798348897
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	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">John,<div><br></div><div>Many thanks for the SIPREC background. I have =
not had the cycles to follow that group and so your note here is =
extremely helpful. Thank you!</div><div><br></div><div>As I read your =
summary though, I keep hearing "Danger! Danger!" in my brain. I =
completely agree with this statement of =
yours:</div><div><br></div><div><blockquote type=3D"cite"><div>Clearly =
there would be a fairly big hurdle for browsers to support SRC =
functionality.&nbsp;</div></blockquote></div><div><br></div><div>I think =
it would be a very large hurdle and really takes us back into basically =
baking a SIP UA into a web browser - and do we REALLY want to go down =
that road?</div><div><br></div><div>If I go back to the charter of this =
group (&nbsp;<a =
href=3D"http://tools.ietf.org/wg/rtcweb/charters">http://tools.ietf.org/wg=
/rtcweb/charters</a>&nbsp;) and why I started following it from the =
start and reading all the traffic, I think we need to focus on how we =
enable direct communication between web browsers... or between web =
browsers and servers... but doing so in the most lightweight and easy =
way possible. &nbsp;</div><div><br></div><div>My interest has been in =
how we can do real-time communication *without* extensions or plugins. =
&nbsp;Reading all of this, it's sounding like either we do need to have =
a plugin/extension to support SRC capability - or we need to bake a =
great amount of functionality directly into browsers. &nbsp;And that in =
my mind limits the number of browsers that might support all the =
capabilities of RTCWEB. &nbsp;</div><div><br></div><div>I think that =
given the aggressive timelines for the working group deliverables (and =
the market reality that the longer a "standard" solution takes to come =
out the more developers will explore proprietary solutions), I agree =
with Stefan that we should put the recording out of scope for RTCWEB 1.0 =
and focus on getting a solution out there that lets developers start =
building RTCWEB apps.&nbsp;</div><div><br></div><div>For those =
environments that need recording (and I *do* understand the call center =
need), middleboxes can provide a solution today - or some vendors can =
support the RTCWEB communication and *also* provide a recording =
capability. Sure, that's not ideal... and yes, we need to make sure that =
what we do for RTCWEB 1.0 doesn't preclude adding recording to a RTCWEB =
2.0... &nbsp;but we need to get a spec out there that will be useful to =
the majority of developers and very easy for them to adopt. =
&nbsp;</div><div><br></div><div>The more complicated we make it - or the =
more requirements we impose on browsers - the less adoption we'll =
see.</div><div><br></div><div>My 2 =
cents,</div><div>Dan</div><div><br></div><div><br><div><div>On Aug 23, =
2011, at 3:58 AM, Elwell, John wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div>There =
has been some discussion recently on remote recording, mixed to some =
extent with discussions on local recording and with mailbox, but I would =
like to focus on remote recording and try to summarize.<br><br>First, =
some background on the IETF SIPREC WG. This is specifying support for =
SIP-based session recording, whereby a Session Recording Client (SRC) on =
the path of a call (communication session) can forward media and =
metadata to a session recording server (SRS) or recording device. In =
conventional SIP terms, the SRC can exist at an endpoint of the =
communication session being recorded (i.e., at a SIP UA), or at a B2BUA =
that has access to the media as well as the signalling. Very often in a =
contact centre, there are mandatory requirements for recording some or =
all communication sessions, and often calls are routed through a B2BUA =
that also provides the SRC. So in this case there is no responsibility =
on SIP UAs to support SRC functionality, and no issues of additional =
bandwidth on the device's access. However, it is anticipated in SIPREC =
that in some deployments UA-located SRCs will be used. How a UA is =
organized internally to provide SRC functionality is not =
standardized.<br><br>So the question for RTC-Web is whether a SIP UA =
implemented as an RTC-Web client can provide SRC functionality, i.e., =
support remote recording. An RTC-Web SIP UA is implemented by a =
combination of functionality running on the web server, functionality =
running in client side script (JS) and functionality embedded in the =
browser. The amount of functionality needed in the browser and needing =
to be exposed at the browser API in support of SRC will depend to some =
extent on how much core functionality goes into the browser, in =
particular whether the browser implements SIP or not. Some of the =
functions noted to date include:<br>- ability to take a copy of streams =
sent to / received from the remote party and send them, in real-time, to =
a remote recording device (SRS);<br>- possible need to mix the copied =
streams before sending (e.g., mix the sent and received audio =
streams)<br>- possible need to use a different codec or other parameters =
when sending to the SRS;<br>- possible need to use a different =
encryption/integrity context when sending to the SRS;<br>- possible need =
to insert tones / announcements into the original media path being =
recorded;<br>- possible need to support SDP enhancements for indicating =
media that are being recorded or preferences for which media are being =
recorded;<br>- possible need to support SIP enhancements for indicating =
SRC/SRS capability and recording awareness (if SIP is implemented in =
browser);<br>- possible need to support the sending of metadata to the =
SRS (if SIP is implemented in browser).<br><br>Clearly there would be a =
fairly big hurdle for browsers to support SRC functionality. But without =
this, RTC-Web clients would not be suitable for use in environments =
where remote recording is required and calls are not forced through some =
middlebox that provides SRC functionality.<br><br>John<br><br>John =
Elwell<br>Tel: +44 1908 817801 (office and mobile)<br>Email: <a =
href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@siemens-ent=
erprise.com</a><br><a =
href=3D"http://www.siemens-enterprise.com/uk/">http://www.siemens-enterpri=
se.com/uk/</a><br><br>Siemens Enterprise Communications =
Limited.<br>Registered office: Brickhill Street, Willen Lake, Milton =
Keynes, MK15 0DJ.<br>Registered No: 5903714, England.<br><br>Siemens =
Enterprise Communications Limited is a Trademark Licensee of Siemens =
AG.<br>_______________________________________________<br>rtcweb mailing =
list<br>rtcweb@ietf.org<br>https://www.ietf.org/mailman/listinfo/rtcweb<br=
></div></blockquote></div><br><div>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
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break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
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style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
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auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
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medium; "><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; =
">--&nbsp;</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Dan York, CISSP, Director of =
Conversations</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; ">Voxeo =
Corporation&nbsp;&nbsp;&nbsp;<a =
href=3D"http://www.voxeo.com">http://www.voxeo.com</a>&nbsp;&nbsp;<a =
href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Phone: +1-321-710-9193 &nbsp;skype: =
danyork &nbsp;<a =
href=3D"sip:dyork@voxeo.com">sip:dyork@voxeo.com</a></div></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; "><br></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font-size: =
12px; "><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><b><font class=3D"Apple-style-span" =
color=3D"#2400BB">Join the Voxeo conversation:</font></b></div><div =
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margin-left: 0px; "><b><font class=3D"Apple-style-span" =
color=3D"#2400BB">Blogs: <a =
href=3D"http://blogs.voxeo.com">http://blogs.voxeo.com</a></font></b></div=
><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12px/normal Helvetica; =
min-height: 14px; "><b><font class=3D"Apple-style-span" =
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href=3D"http://twitter.com/danyork">http://twitter.com/danyork</a></font><=
/b></div><div style=3D"margin-top: 0px; margin-right: 0px; =
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12px/normal Helvetica; min-height: 14px; "><b><font =
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href=3D"http://www.facebook.com/voxeo">http://www.facebook.com/voxeo</a></=
font></b></div></div></span></div></span></div></span></div></span></div><=
/span></div></span></div></span></div></span></div></span></div></span></d=
iv></span></div></span></div>
</div>
<br></div></body></html>=

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From john.elwell@siemens-enterprise.com  Tue Aug 23 08:36:13 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Dan York <dyork@voxeo.com>
Date: Tue, 23 Aug 2011 17:37:15 +0200
Thread-Topic: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, "public-webrtc@w3.org" <public-webrtc@w3.org>
Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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Dan,

Thanks for your input. I agree with much of what you say. Like Paul, I thin=
k it is good to have the discussion, so that we can at least understand whe=
ther or not we are going to specify browser support for remote recording an=
d how such a decision has been reached. However, I would like to comment on=
 one point below:=20

> -----Original Message-----
> From: Dan York [mailto:dyork@voxeo.com]=20
> Sent: 23 August 2011 16:23
> To: Elwell, John
> Cc: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Remote recording - RTC-Web client=20
> acting as SIPREC session recording client
>=20
> John,=20
>=20
> Many thanks for the SIPREC background. I have not had the=20
> cycles to follow that group and so your note here is=20
> extremely helpful. Thank you!
>=20
> As I read your summary though, I keep hearing "Danger!=20
> Danger!" in my brain. I completely agree with this statement of yours:
>=20
>=20
> 	Clearly there would be a fairly big hurdle for browsers=20
> to support SRC functionality.=20
>=20
>=20
> I think it would be a very large hurdle and really takes us=20
> back into basically baking a SIP UA into a web browser - and=20
> do we REALLY want to go down that road?
[JRE] Some people are still talking about SIP in the browser, I believe. Bu=
t one of the points I was making is that, assuming SIP is NOT in the browse=
r, there is somewhat less that needs to be done to provide browser support =
for an SRC (the rest would be up to the web application).

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.

>=20
> If I go back to the charter of this group (=20
> http://tools.ietf.org/wg/rtcweb/charters ) and why I started=20
> following it from the start and reading all the traffic, I=20
> think we need to focus on how we enable direct communication=20
> between web browsers... or between web browsers and=20
> servers... but doing so in the most lightweight and easy way=20
> possible. =20
>=20
> My interest has been in how we can do real-time communication=20
> *without* extensions or plugins.  Reading all of this, it's=20
> sounding like either we do need to have a plugin/extension to=20
> support SRC capability - or we need to bake a great amount of=20
> functionality directly into browsers.  And that in my mind=20
> limits the number of browsers that might support all the=20
> capabilities of RTCWEB. =20
>=20
> I think that given the aggressive timelines for the working=20
> group deliverables (and the market reality that the longer a=20
> "standard" solution takes to come out the more developers=20
> will explore proprietary solutions), I agree with Stefan that=20
> we should put the recording out of scope for RTCWEB 1.0 and=20
> focus on getting a solution out there that lets developers=20
> start building RTCWEB apps.=20
>=20
> For those environments that need recording (and I *do*=20
> understand the call center need), middleboxes can provide a=20
> solution today - or some vendors can support the RTCWEB=20
> communication and *also* provide a recording capability.=20
> Sure, that's not ideal... and yes, we need to make sure that=20
> what we do for RTCWEB 1.0 doesn't preclude adding recording=20
> to a RTCWEB 2.0...  but we need to get a spec out there that=20
> will be useful to the majority of developers and very easy=20
> for them to adopt. =20
>=20
> The more complicated we make it - or the more requirements we=20
> impose on browsers - the less adoption we'll see.
>=20
> My 2 cents,
> Dan
>=20
>=20
> On Aug 23, 2011, at 3:58 AM, Elwell, John wrote:
>=20
>=20
> 	There has been some discussion recently on remote=20
> recording, mixed to some extent with discussions on local=20
> recording and with mailbox, but I would like to focus on=20
> remote recording and try to summarize.
> =09
> 	First, some background on the IETF SIPREC WG. This is=20
> specifying support for SIP-based session recording, whereby a=20
> Session Recording Client (SRC) on the path of a call=20
> (communication session) can forward media and metadata to a=20
> session recording server (SRS) or recording device. In=20
> conventional SIP terms, the SRC can exist at an endpoint of=20
> the communication session being recorded (i.e., at a SIP UA),=20
> or at a B2BUA that has access to the media as well as the=20
> signalling. Very often in a contact centre, there are=20
> mandatory requirements for recording some or all=20
> communication sessions, and often calls are routed through a=20
> B2BUA that also provides the SRC. So in this case there is no=20
> responsibility on SIP UAs to support SRC functionality, and=20
> no issues of additional bandwidth on the device's access.=20
> However, it is anticipated in SIPREC that in some deployments=20
> UA-located SRCs will be used. How a UA is organized=20
> internally to provide SRC functionality is not standardized.
> =09
> 	So the question for RTC-Web is whether a SIP UA=20
> implemented as an RTC-Web client can provide SRC=20
> functionality, i.e., support remote recording. An RTC-Web SIP=20
> UA is implemented by a combination of functionality running=20
> on the web server, functionality running in client side=20
> script (JS) and functionality embedded in the browser. The=20
> amount of functionality needed in the browser and needing to=20
> be exposed at the browser API in support of SRC will depend=20
> to some extent on how much core functionality goes into the=20
> browser, in particular whether the browser implements SIP or=20
> not. Some of the functions noted to date include:
> 	- ability to take a copy of streams sent to / received=20
> from the remote party and send them, in real-time, to a=20
> remote recording device (SRS);
> 	- possible need to mix the copied streams before=20
> sending (e.g., mix the sent and received audio streams)
> 	- possible need to use a different codec or other=20
> parameters when sending to the SRS;
> 	- possible need to use a different encryption/integrity=20
> context when sending to the SRS;
> 	- possible need to insert tones / announcements into=20
> the original media path being recorded;
> 	- possible need to support SDP enhancements for=20
> indicating media that are being recorded or preferences for=20
> which media are being recorded;
> 	- possible need to support SIP enhancements for=20
> indicating SRC/SRS capability and recording awareness (if SIP=20
> is implemented in browser);
> 	- possible need to support the sending of metadata to=20
> the SRS (if SIP is implemented in browser).
> =09
> 	Clearly there would be a fairly big hurdle for browsers=20
> to support SRC functionality. But without this, RTC-Web=20
> clients would not be suitable for use in environments where=20
> remote recording is required and calls are not forced through=20
> some middlebox that provides SRC functionality.
> =09
> 	John
> =09
> 	John Elwell
> 	Tel: +44 1908 817801 (office and mobile)
> 	Email: john.elwell@siemens-enterprise.com
> 	http://www.siemens-enterprise.com/uk/
> =09
> 	Siemens Enterprise Communications Limited.
> 	Registered office: Brickhill Street, Willen Lake,=20
> Milton Keynes, MK15 0DJ.
> 	Registered No: 5903714, England.
> =09
> 	Siemens Enterprise Communications Limited is a=20
> Trademark Licensee of Siemens AG.
> 	_______________________________________________
> 	rtcweb mailing list
> 	rtcweb@ietf.org
> 	https://www.ietf.org/mailman/listinfo/rtcweb
> =09
>=20
>=20
> --=20
> Dan York, CISSP, Director of Conversations
> Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
> Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com
>=20
> Join the Voxeo conversation:
> Blogs: http://blogs.voxeo.com
> Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
> Facebook: http://www.facebook.com/voxeo
>=20
> =

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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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23 aug 2011 kl. 17:23 skrev Dan York:

> The more complicated we make it - or the more requirements we impose =
on browsers - the less adoption we'll see.
>=20
Which we've proven with the large amount of RFCs and drafts for SIP.

We need to be very careful here.=20

+1 and "like" on Dan's mail.

/O


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
"><br><div><div>23 aug 2011 kl. 17:23 skrev Dan York:</div><br =
class=3D"Apple-interchange-newline"><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>The more =
complicated we make it - or the more requirements we impose on browsers =
- the less adoption we'll =
see.</div><div><br></div></span></blockquote>Which we've proven with the =
large amount of RFCs and drafts for SIP.</div><div><br></div><div>We =
need to be very careful here.&nbsp;</div><div><br></div><div>+1 and =
"like" on Dan's =
mail.</div><div><br></div><div>/O</div><br></body></html>=

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From: Dan York <dyork@voxeo.com>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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John,

You're welcome... and I agree with you and Paul on the need to have the =
discussion and to consciously make a choice as to whether to include or =
not include some functionality.   I also appreciate that you took the =
time to draft up text for requirements to add to the use cases... it was =
seeing that new requirement text that made me start violently twitching =
here in my home office and lured me out of lurking on the list.  :-)

Dan

P.S. And I agree with your point below.

On Aug 23, 2011, at 11:37 AM, Elwell, John wrote:

> Dan,
>=20
> Thanks for your input. I agree with much of what you say. Like Paul, I =
think it is good to have the discussion, so that we can at least =
understand whether or not we are going to specify browser support for =
remote recording and how such a decision has been reached. However, I =
would like to comment on one point below:=20
>=20
>> -----Original Message-----
>> From: Dan York [mailto:dyork@voxeo.com]=20
>> Sent: 23 August 2011 16:23
>> To: Elwell, John
>> Cc: rtcweb@ietf.org; public-webrtc@w3.org
>> Subject: Re: [rtcweb] Remote recording - RTC-Web client=20
>> acting as SIPREC session recording client
>>=20
>> John,=20
>>=20
>> Many thanks for the SIPREC background. I have not had the=20
>> cycles to follow that group and so your note here is=20
>> extremely helpful. Thank you!
>>=20
>> As I read your summary though, I keep hearing "Danger!=20
>> Danger!" in my brain. I completely agree with this statement of =
yours:
>>=20
>>=20
>> 	Clearly there would be a fairly big hurdle for browsers=20
>> to support SRC functionality.=20
>>=20
>>=20
>> I think it would be a very large hurdle and really takes us=20
>> back into basically baking a SIP UA into a web browser - and=20
>> do we REALLY want to go down that road?
> [JRE] Some people are still talking about SIP in the browser, I =
believe. But one of the points I was making is that, assuming SIP is NOT =
in the browser, there is somewhat less that needs to be done to provide =
browser support for an SRC (the rest would be up to the web =
application).
>=20
> John
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 =
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of =
Siemens AG.
>=20
>>=20
>> If I go back to the charter of this group (=20
>> http://tools.ietf.org/wg/rtcweb/charters ) and why I started=20
>> following it from the start and reading all the traffic, I=20
>> think we need to focus on how we enable direct communication=20
>> between web browsers... or between web browsers and=20
>> servers... but doing so in the most lightweight and easy way=20
>> possible. =20
>>=20
>> My interest has been in how we can do real-time communication=20
>> *without* extensions or plugins.  Reading all of this, it's=20
>> sounding like either we do need to have a plugin/extension to=20
>> support SRC capability - or we need to bake a great amount of=20
>> functionality directly into browsers.  And that in my mind=20
>> limits the number of browsers that might support all the=20
>> capabilities of RTCWEB. =20
>>=20
>> I think that given the aggressive timelines for the working=20
>> group deliverables (and the market reality that the longer a=20
>> "standard" solution takes to come out the more developers=20
>> will explore proprietary solutions), I agree with Stefan that=20
>> we should put the recording out of scope for RTCWEB 1.0 and=20
>> focus on getting a solution out there that lets developers=20
>> start building RTCWEB apps.=20
>>=20
>> For those environments that need recording (and I *do*=20
>> understand the call center need), middleboxes can provide a=20
>> solution today - or some vendors can support the RTCWEB=20
>> communication and *also* provide a recording capability.=20
>> Sure, that's not ideal... and yes, we need to make sure that=20
>> what we do for RTCWEB 1.0 doesn't preclude adding recording=20
>> to a RTCWEB 2.0...  but we need to get a spec out there that=20
>> will be useful to the majority of developers and very easy=20
>> for them to adopt. =20
>>=20
>> The more complicated we make it - or the more requirements we=20
>> impose on browsers - the less adoption we'll see.
>>=20
>> My 2 cents,
>> Dan
>>=20
>>=20
>> On Aug 23, 2011, at 3:58 AM, Elwell, John wrote:
>>=20
>>=20
>> 	There has been some discussion recently on remote=20
>> recording, mixed to some extent with discussions on local=20
>> recording and with mailbox, but I would like to focus on=20
>> remote recording and try to summarize.
>> =09
>> 	First, some background on the IETF SIPREC WG. This is=20
>> specifying support for SIP-based session recording, whereby a=20
>> Session Recording Client (SRC) on the path of a call=20
>> (communication session) can forward media and metadata to a=20
>> session recording server (SRS) or recording device. In=20
>> conventional SIP terms, the SRC can exist at an endpoint of=20
>> the communication session being recorded (i.e., at a SIP UA),=20
>> or at a B2BUA that has access to the media as well as the=20
>> signalling. Very often in a contact centre, there are=20
>> mandatory requirements for recording some or all=20
>> communication sessions, and often calls are routed through a=20
>> B2BUA that also provides the SRC. So in this case there is no=20
>> responsibility on SIP UAs to support SRC functionality, and=20
>> no issues of additional bandwidth on the device's access.=20
>> However, it is anticipated in SIPREC that in some deployments=20
>> UA-located SRCs will be used. How a UA is organized=20
>> internally to provide SRC functionality is not standardized.
>> =09
>> 	So the question for RTC-Web is whether a SIP UA=20
>> implemented as an RTC-Web client can provide SRC=20
>> functionality, i.e., support remote recording. An RTC-Web SIP=20
>> UA is implemented by a combination of functionality running=20
>> on the web server, functionality running in client side=20
>> script (JS) and functionality embedded in the browser. The=20
>> amount of functionality needed in the browser and needing to=20
>> be exposed at the browser API in support of SRC will depend=20
>> to some extent on how much core functionality goes into the=20
>> browser, in particular whether the browser implements SIP or=20
>> not. Some of the functions noted to date include:
>> 	- ability to take a copy of streams sent to / received=20
>> from the remote party and send them, in real-time, to a=20
>> remote recording device (SRS);
>> 	- possible need to mix the copied streams before=20
>> sending (e.g., mix the sent and received audio streams)
>> 	- possible need to use a different codec or other=20
>> parameters when sending to the SRS;
>> 	- possible need to use a different encryption/integrity=20
>> context when sending to the SRS;
>> 	- possible need to insert tones / announcements into=20
>> the original media path being recorded;
>> 	- possible need to support SDP enhancements for=20
>> indicating media that are being recorded or preferences for=20
>> which media are being recorded;
>> 	- possible need to support SIP enhancements for=20
>> indicating SRC/SRS capability and recording awareness (if SIP=20
>> is implemented in browser);
>> 	- possible need to support the sending of metadata to=20
>> the SRS (if SIP is implemented in browser).
>> =09
>> 	Clearly there would be a fairly big hurdle for browsers=20
>> to support SRC functionality. But without this, RTC-Web=20
>> clients would not be suitable for use in environments where=20
>> remote recording is required and calls are not forced through=20
>> some middlebox that provides SRC functionality.
>> =09
>> 	John
>> =09
>> 	John Elwell
>> 	Tel: +44 1908 817801 (office and mobile)
>> 	Email: john.elwell@siemens-enterprise.com
>> 	http://www.siemens-enterprise.com/uk/
>> =09
>> 	Siemens Enterprise Communications Limited.
>> 	Registered office: Brickhill Street, Willen Lake,=20
>> Milton Keynes, MK15 0DJ.
>> 	Registered No: 5903714, England.
>> =09
>> 	Siemens Enterprise Communications Limited is a=20
>> Trademark Licensee of Siemens AG.
>> 	_______________________________________________
>> 	rtcweb mailing list
>> 	rtcweb@ietf.org
>> 	https://www.ietf.org/mailman/listinfo/rtcweb
>> =09
>>=20
>>=20
>> --=20
>> Dan York, CISSP, Director of Conversations
>> Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
>> Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com
>>=20
>> Join the Voxeo conversation:
>> Blogs: http://blogs.voxeo.com
>> Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
>> Facebook: http://www.facebook.com/voxeo
>>=20
>>=20

--=20
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo


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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">John,<div><br></div><div>You're welcome... and I agree with you and =
Paul on the need to have the discussion and to consciously make a choice =
as to whether to include or not include some functionality. &nbsp; I =
also appreciate that you took the time to draft up text for requirements =
to add to the use cases... it was seeing that new requirement text that =
made me start violently twitching here in my home office and lured me =
out of lurking on the list. =
&nbsp;:-)</div><div><br></div><div>Dan</div><div><br></div><div>P.S. And =
I agree with your point below.</div><div><br><div><div>On Aug 23, 2011, =
at 11:37 AM, Elwell, John wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div>Dan,<br><br>Thanks for your input. I agree with much =
of what you say. Like Paul, I think it is good to have the discussion, =
so that we can at least understand whether or not we are going to =
specify browser support for remote recording and how such a decision has =
been reached. However, I would like to comment on one point below: =
<br><br><blockquote type=3D"cite">-----Original =
Message-----<br></blockquote><blockquote type=3D"cite">From: Dan York =
[mailto:dyork@voxeo.com] <br></blockquote><blockquote type=3D"cite">Sent: =
23 August 2011 16:23<br></blockquote><blockquote type=3D"cite">To: =
Elwell, John<br></blockquote><blockquote type=3D"cite">Cc: <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>; <a =
href=3D"mailto:public-webrtc@w3.org">public-webrtc@w3.org</a><br></blockqu=
ote><blockquote type=3D"cite">Subject: Re: [rtcweb] Remote recording - =
RTC-Web client <br></blockquote><blockquote type=3D"cite">acting as =
SIPREC session recording client<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">John, =
<br></blockquote><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">Many thanks for the SIPREC background. I have not had the =
<br></blockquote><blockquote type=3D"cite">cycles to follow that group =
and so your note here is <br></blockquote><blockquote =
type=3D"cite">extremely helpful. Thank you!<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">As I read your =
summary though, I keep hearing "Danger! <br></blockquote><blockquote =
type=3D"cite">Danger!" in my brain. I completely agree with this =
statement of yours:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Clearly =
there would be a fairly big hurdle for browsers =
<br></blockquote><blockquote type=3D"cite">to support SRC functionality. =
<br></blockquote><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">I think it =
would be a very large hurdle and really takes us =
<br></blockquote><blockquote type=3D"cite">back into basically baking a =
SIP UA into a web browser - and <br></blockquote><blockquote =
type=3D"cite">do we REALLY want to go down that =
road?<br></blockquote>[JRE] Some people are still talking about SIP in =
the browser, I believe. But one of the points I was making is that, =
assuming SIP is NOT in the browser, there is somewhat less that needs to =
be done to provide browser support for an SRC (the rest would be up to =
the web application).<br><br>John<br><br>John Elwell<br>Tel: +44 1908 =
817801 (office and mobile)<br>Email: <a =
href=3D"mailto:john.elwell@siemens-enterprise.com">john.elwell@siemens-ent=
erprise.com</a><br><a =
href=3D"http://www.siemens-enterprise.com/uk/">http://www.siemens-enterpri=
se.com/uk/</a><br><br>Siemens Enterprise Communications =
Limited.<br>Registered office: Brickhill Street, Willen Lake, Milton =
Keynes, MK15 0DJ.<br>Registered No: 5903714, England.<br><br>Siemens =
Enterprise Communications Limited is a Trademark Licensee of Siemens =
AG.<br><br><blockquote type=3D"cite"><br></blockquote><blockquote =
type=3D"cite">If I go back to the charter of this group ( =
<br></blockquote><blockquote =
type=3D"cite">http://tools.ietf.org/wg/rtcweb/charters ) and why I =
started <br></blockquote><blockquote type=3D"cite">following it from the =
start and reading all the traffic, I <br></blockquote><blockquote =
type=3D"cite">think we need to focus on how we enable direct =
communication <br></blockquote><blockquote type=3D"cite">between web =
browsers... or between web browsers and <br></blockquote><blockquote =
type=3D"cite">servers... but doing so in the most lightweight and easy =
way <br></blockquote><blockquote type=3D"cite">possible. =
&nbsp;<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">My interest has =
been in how we can do real-time communication =
<br></blockquote><blockquote type=3D"cite">*without* extensions or =
plugins. &nbsp;Reading all of this, it's <br></blockquote><blockquote =
type=3D"cite">sounding like either we do need to have a plugin/extension =
to <br></blockquote><blockquote type=3D"cite">support SRC capability - =
or we need to bake a great amount of <br></blockquote><blockquote =
type=3D"cite">functionality directly into browsers. &nbsp;And that in my =
mind <br></blockquote><blockquote type=3D"cite">limits the number of =
browsers that might support all the <br></blockquote><blockquote =
type=3D"cite">capabilities of RTCWEB. &nbsp;<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">I think that =
given the aggressive timelines for the working =
<br></blockquote><blockquote type=3D"cite">group deliverables (and the =
market reality that the longer a <br></blockquote><blockquote =
type=3D"cite">"standard" solution takes to come out the more developers =
<br></blockquote><blockquote type=3D"cite">will explore proprietary =
solutions), I agree with Stefan that <br></blockquote><blockquote =
type=3D"cite">we should put the recording out of scope for RTCWEB 1.0 =
and <br></blockquote><blockquote type=3D"cite">focus on getting a =
solution out there that lets developers <br></blockquote><blockquote =
type=3D"cite">start building RTCWEB apps. <br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">For those =
environments that need recording (and I *do* =
<br></blockquote><blockquote type=3D"cite">understand the call center =
need), middleboxes can provide a <br></blockquote><blockquote =
type=3D"cite">solution today - or some vendors can support the RTCWEB =
<br></blockquote><blockquote type=3D"cite">communication and *also* =
provide a recording capability. <br></blockquote><blockquote =
type=3D"cite">Sure, that's not ideal... and yes, we need to make sure =
that <br></blockquote><blockquote type=3D"cite">what we do for RTCWEB =
1.0 doesn't preclude adding recording <br></blockquote><blockquote =
type=3D"cite">to a RTCWEB 2.0... &nbsp;but we need to get a spec out =
there that <br></blockquote><blockquote type=3D"cite">will be useful to =
the majority of developers and very easy <br></blockquote><blockquote =
type=3D"cite">for them to adopt. &nbsp;<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">The more =
complicated we make it - or the more requirements we =
<br></blockquote><blockquote type=3D"cite">impose on browsers - the less =
adoption we'll see.<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">My 2 =
cents,<br></blockquote><blockquote =
type=3D"cite">Dan<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">On Aug 23, =
2011, at 3:58 AM, Elwell, John wrote:<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>There has =
been some discussion recently on remote <br></blockquote><blockquote =
type=3D"cite">recording, mixed to some extent with discussions on local =
<br></blockquote><blockquote type=3D"cite">recording and with mailbox, =
but I would like to focus on <br></blockquote><blockquote =
type=3D"cite">remote recording and try to =
summarize.<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>First, =
some background on the IETF SIPREC WG. This is =
<br></blockquote><blockquote type=3D"cite">specifying support for =
SIP-based session recording, whereby a <br></blockquote><blockquote =
type=3D"cite">Session Recording Client (SRC) on the path of a call =
<br></blockquote><blockquote type=3D"cite">(communication session) can =
forward media and metadata to a <br></blockquote><blockquote =
type=3D"cite">session recording server (SRS) or recording device. In =
<br></blockquote><blockquote type=3D"cite">conventional SIP terms, the =
SRC can exist at an endpoint of <br></blockquote><blockquote =
type=3D"cite">the communication session being recorded (i.e., at a SIP =
UA), <br></blockquote><blockquote type=3D"cite">or at a B2BUA that has =
access to the media as well as the <br></blockquote><blockquote =
type=3D"cite">signalling. Very often in a contact centre, there are =
<br></blockquote><blockquote type=3D"cite">mandatory requirements for =
recording some or all <br></blockquote><blockquote =
type=3D"cite">communication sessions, and often calls are routed through =
a <br></blockquote><blockquote type=3D"cite">B2BUA that also provides =
the SRC. So in this case there is no <br></blockquote><blockquote =
type=3D"cite">responsibility on SIP UAs to support SRC functionality, =
and <br></blockquote><blockquote type=3D"cite">no issues of additional =
bandwidth on the device's access. <br></blockquote><blockquote =
type=3D"cite">However, it is anticipated in SIPREC that in some =
deployments <br></blockquote><blockquote type=3D"cite">UA-located SRCs =
will be used. How a UA is organized <br></blockquote><blockquote =
type=3D"cite">internally to provide SRC functionality is not =
standardized.<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>So the =
question for RTC-Web is whether a SIP UA <br></blockquote><blockquote =
type=3D"cite">implemented as an RTC-Web client can provide SRC =
<br></blockquote><blockquote type=3D"cite">functionality, i.e., support =
remote recording. An RTC-Web SIP <br></blockquote><blockquote =
type=3D"cite">UA is implemented by a combination of functionality =
running <br></blockquote><blockquote type=3D"cite">on the web server, =
functionality running in client side <br></blockquote><blockquote =
type=3D"cite">script (JS) and functionality embedded in the browser. The =
<br></blockquote><blockquote type=3D"cite">amount of functionality =
needed in the browser and needing to <br></blockquote><blockquote =
type=3D"cite">be exposed at the browser API in support of SRC will =
depend <br></blockquote><blockquote type=3D"cite">to some extent on how =
much core functionality goes into the <br></blockquote><blockquote =
type=3D"cite">browser, in particular whether the browser implements SIP =
or <br></blockquote><blockquote type=3D"cite">not. Some of the functions =
noted to date include:<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- ability =
to take a copy of streams sent to / received =
<br></blockquote><blockquote type=3D"cite">from the remote party and =
send them, in real-time, to a <br></blockquote><blockquote =
type=3D"cite">remote recording device (SRS);<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>- possible need to mix the copied streams before =
<br></blockquote><blockquote type=3D"cite">sending (e.g., mix the sent =
and received audio streams)<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>- possible need to use a different codec or other =
<br></blockquote><blockquote type=3D"cite">parameters when sending to =
the SRS;<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- =
possible need to use a different encryption/integrity =
<br></blockquote><blockquote type=3D"cite">context when sending to the =
SRS;<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- =
possible need to insert tones / announcements into =
<br></blockquote><blockquote type=3D"cite">the original media path being =
recorded;<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- =
possible need to support SDP enhancements for =
<br></blockquote><blockquote type=3D"cite">indicating media that are =
being recorded or preferences for <br></blockquote><blockquote =
type=3D"cite">which media are being =
recorded;<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>- =
possible need to support SIP enhancements for =
<br></blockquote><blockquote type=3D"cite">indicating SRC/SRS capability =
and recording awareness (if SIP <br></blockquote><blockquote =
type=3D"cite">is implemented in browser);<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>- possible need to support the sending of metadata to =
<br></blockquote><blockquote type=3D"cite">the SRS (if SIP is =
implemented in browser).<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Clearly =
there would be a fairly big hurdle for browsers =
<br></blockquote><blockquote type=3D"cite">to support SRC functionality. =
But without this, RTC-Web <br></blockquote><blockquote =
type=3D"cite">clients would not be suitable for use in environments =
where <br></blockquote><blockquote type=3D"cite">remote recording is =
required and calls are not forced through <br></blockquote><blockquote =
type=3D"cite">some middlebox that provides SRC =
functionality.<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>John<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>John =
Elwell<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Tel: +44 =
1908 817801 (office and mobile)<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>Email: =
john.elwell@siemens-enterprise.com<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>http://www.siemens-enterprise.com/uk/<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Siemens =
Enterprise Communications Limited.<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>Registered office: Brickhill Street, Willen Lake, =
<br></blockquote><blockquote type=3D"cite">Milton Keynes, MK15 =
0DJ.<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>Registered No: 5903714, England.<br></blockquote><blockquote =
type=3D"cite"><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span><br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	</span>Siemens =
Enterprise Communications Limited is a <br></blockquote><blockquote =
type=3D"cite">Trademark Licensee of Siemens =
AG.<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>_______________________________________________<br></blockquote><bl=
ockquote type=3D"cite"><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span>rtcweb mailing =
list<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>rtcweb@ietf.org<br></blockquote><blockquote type=3D"cite"><span =
class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span>https://www.ietf.org/mailman/listinfo/rtcweb<br></blockquote><block=
quote type=3D"cite"><span class=3D"Apple-tab-span" =
style=3D"white-space:pre">	</span><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">-- =
<br></blockquote><blockquote type=3D"cite">Dan York, CISSP, Director of =
Conversations<br></blockquote><blockquote type=3D"cite">Voxeo =
Corporation &nbsp;&nbsp;http://www.voxeo.com =
&nbsp;dyork@voxeo.com<br></blockquote><blockquote type=3D"cite">Phone: =
+1-321-710-9193 &nbsp;skype: danyork =
&nbsp;sip:dyork@voxeo.com<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote type=3D"cite">Join the Voxeo =
conversation:<br></blockquote><blockquote type=3D"cite">Blogs: =
http://blogs.voxeo.com<br></blockquote><blockquote type=3D"cite">Twitter: =
http://twitter.com/voxeo =
&nbsp;http://twitter.com/danyork<br></blockquote><blockquote =
type=3D"cite">Facebook: =
http://www.facebook.com/voxeo<br></blockquote><blockquote =
type=3D"cite"><br></blockquote><blockquote =
type=3D"cite"><br></blockquote></div></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: auto; text-indent: 0px; =
text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: =
normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><span class=3D"Apple-style-span" style=3D"font-size: =
medium; "><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; =
">--&nbsp;</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; "><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Dan York, CISSP, Director of =
Conversations</div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font-size: 12px; ">Voxeo =
Corporation&nbsp;&nbsp;&nbsp;<a =
href=3D"http://www.voxeo.com">http://www.voxeo.com</a>&nbsp;&nbsp;<a =
href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; ">Phone: +1-321-710-9193 &nbsp;skype: =
danyork &nbsp;<a =
href=3D"sip:dyork@voxeo.com">sip:dyork@voxeo.com</a></div></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font-size: 12px; "><br></div><div style=3D"margin-top: =
0px; margin-right: 0px; margin-bottom: 0px; margin-left: 0px; font-size: =
12px; "><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: =
0px; margin-left: 0px; "><b><font class=3D"Apple-style-span" =
color=3D"#2400BB">Join the Voxeo conversation:</font></b></div><div =
style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; "><b><font class=3D"Apple-style-span" =
color=3D"#2400BB">Blogs: <a =
href=3D"http://blogs.voxeo.com">http://blogs.voxeo.com</a></font></b></div=
><div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; =
margin-left: 0px; font: normal normal normal 12px/normal Helvetica; =
min-height: 14px; "><b><font class=3D"Apple-style-span" =
color=3D"#2400BB">Twitter: <a =
href=3D"http://twitter.com/voxeo">http://twitter.com/voxeo</a> &nbsp;<a =
href=3D"http://twitter.com/danyork">http://twitter.com/danyork</a></font><=
/b></div><div style=3D"margin-top: 0px; margin-right: 0px; =
margin-bottom: 0px; margin-left: 0px; font: normal normal normal =
12px/normal Helvetica; min-height: 14px; "><b><font =
class=3D"Apple-style-span" color=3D"#2400BB">Facebook: <a =
href=3D"http://www.facebook.com/voxeo">http://www.facebook.com/voxeo</a></=
font></b></div></div></span></div></span></div></span></div></span></div><=
/span></div></span></div></span></div></span></div></span></div></span></d=
iv></span></div></span></div></span></span>
</div>
<br></div></body></html>=

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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Dan York <dyork@voxeo.com>, "Elwell, John" <john.elwell@siemens-enterprise.com>
Date: Tue, 23 Aug 2011 19:50:53 +0200
Thread-Topic: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC	session recording client
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Hi,

Also agree that implementing a full SIP Session Recording Client (SRC) as d=
efined by the SIPREC WG in to the browser would be a tall order and would o=
pen the flood gates for a lot of other requests for SIP functionality in th=
e browser.

If however SIP is not in the browser then I think it is a useful and intere=
sting test case to look at whether we could build a decomposed SRC with the=
 browser handling the media and the web server handling the signalling. Wha=
t I heard a number of times at IETF81 is that what people want is to build =
innovative new applications with a real time media enabled browser and to i=
t seems clear to me we need a browser API which is flexible and enables app=
lications to do clever things with media streams. So exploring some use cas=
es which require the browser to duplicate and copy media streams is a good =
idea.

Therefore I think we should keep explore the remote recording use case furt=
her.

Regards
Andy

Andrew Hutton
Siemens Enterprise Communications Limited.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.




From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Dan York
Sent: 23 August 2011 16:49
To: Elwell, John
Cc: rtcweb@ietf.org; public-webrtc@w3.org
Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC se=
ssion recording client

John,

You're welcome... and I agree with you and Paul on the need to have the dis=
cussion and to consciously make a choice as to whether to include or not in=
clude some functionality.   I also appreciate that you took the time to dra=
ft up text for requirements to add to the use cases... it was seeing that n=
ew requirement text that made me start violently twitching here in my home =
office and lured me out of lurking on the list.  :-)

Dan

P.S. And I agree with your point below.

On Aug 23, 2011, at 11:37 AM, Elwell, John wrote:


Dan,

Thanks for your input. I agree with much of what you say. Like Paul, I thin=
k it is good to have the discussion, so that we can at least understand whe=
ther or not we are going to specify browser support for remote recording an=
d how such a decision has been reached. However, I would like to comment on=
 one point below:


-----Original Message-----
From: Dan York [mailto:dyork@voxeo.com]
Sent: 23 August 2011 16:23
To: Elwell, John
Cc: rtcweb@ietf.org<mailto:rtcweb@ietf.org>; public-webrtc@w3.org<mailto:pu=
blic-webrtc@w3.org>
Subject: Re: [rtcweb] Remote recording - RTC-Web client
acting as SIPREC session recording client

John,

Many thanks for the SIPREC background. I have not had the
cycles to follow that group and so your note here is
extremely helpful. Thank you!

As I read your summary though, I keep hearing "Danger!
Danger!" in my brain. I completely agree with this statement of yours:


          Clearly there would be a fairly big hurdle for browsers
to support SRC functionality.


I think it would be a very large hurdle and really takes us
back into basically baking a SIP UA into a web browser - and
do we REALLY want to go down that road?
[JRE] Some people are still talking about SIP in the browser, I believe. Bu=
t one of the points I was making is that, assuming SIP is NOT in the browse=
r, there is somewhat less that needs to be done to provide browser support =
for an SRC (the rest would be up to the web application).

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com<mailto:john.elwell@siemens-enterp=
rise.com>
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.



If I go back to the charter of this group (
http://tools.ietf.org/wg/rtcweb/charters ) and why I started
following it from the start and reading all the traffic, I
think we need to focus on how we enable direct communication
between web browsers... or between web browsers and
servers... but doing so in the most lightweight and easy way
possible.

My interest has been in how we can do real-time communication
*without* extensions or plugins.  Reading all of this, it's
sounding like either we do need to have a plugin/extension to
support SRC capability - or we need to bake a great amount of
functionality directly into browsers.  And that in my mind
limits the number of browsers that might support all the
capabilities of RTCWEB.

I think that given the aggressive timelines for the working
group deliverables (and the market reality that the longer a
"standard" solution takes to come out the more developers
will explore proprietary solutions), I agree with Stefan that
we should put the recording out of scope for RTCWEB 1.0 and
focus on getting a solution out there that lets developers
start building RTCWEB apps.

For those environments that need recording (and I *do*
understand the call center need), middleboxes can provide a
solution today - or some vendors can support the RTCWEB
communication and *also* provide a recording capability.
Sure, that's not ideal... and yes, we need to make sure that
what we do for RTCWEB 1.0 doesn't preclude adding recording
to a RTCWEB 2.0...  but we need to get a spec out there that
will be useful to the majority of developers and very easy
for them to adopt.

The more complicated we make it - or the more requirements we
impose on browsers - the less adoption we'll see.

My 2 cents,
Dan


On Aug 23, 2011, at 3:58 AM, Elwell, John wrote:


          There has been some discussion recently on remote
recording, mixed to some extent with discussions on local
recording and with mailbox, but I would like to focus on
remote recording and try to summarize.

          First, some background on the IETF SIPREC WG. This is
specifying support for SIP-based session recording, whereby a
Session Recording Client (SRC) on the path of a call
(communication session) can forward media and metadata to a
session recording server (SRS) or recording device. In
conventional SIP terms, the SRC can exist at an endpoint of
the communication session being recorded (i.e., at a SIP UA),
or at a B2BUA that has access to the media as well as the
signalling. Very often in a contact centre, there are
mandatory requirements for recording some or all
communication sessions, and often calls are routed through a
B2BUA that also provides the SRC. So in this case there is no
responsibility on SIP UAs to support SRC functionality, and
no issues of additional bandwidth on the device's access.
However, it is anticipated in SIPREC that in some deployments
UA-located SRCs will be used. How a UA is organized
internally to provide SRC functionality is not standardized.

          So the question for RTC-Web is whether a SIP UA
implemented as an RTC-Web client can provide SRC
functionality, i.e., support remote recording. An RTC-Web SIP
UA is implemented by a combination of functionality running
on the web server, functionality running in client side
script (JS) and functionality embedded in the browser. The
amount of functionality needed in the browser and needing to
be exposed at the browser API in support of SRC will depend
to some extent on how much core functionality goes into the
browser, in particular whether the browser implements SIP or
not. Some of the functions noted to date include:
          - ability to take a copy of streams sent to / received
from the remote party and send them, in real-time, to a
remote recording device (SRS);
          - possible need to mix the copied streams before
sending (e.g., mix the sent and received audio streams)
          - possible need to use a different codec or other
parameters when sending to the SRS;
          - possible need to use a different encryption/integrity
context when sending to the SRS;
          - possible need to insert tones / announcements into
the original media path being recorded;
          - possible need to support SDP enhancements for
indicating media that are being recorded or preferences for
which media are being recorded;
          - possible need to support SIP enhancements for
indicating SRC/SRS capability and recording awareness (if SIP
is implemented in browser);
          - possible need to support the sending of metadata to
the SRS (if SIP is implemented in browser).

          Clearly there would be a fairly big hurdle for browsers
to support SRC functionality. But without this, RTC-Web
clients would not be suitable for use in environments where
remote recording is required and calls are not forced through
some middlebox that provides SRC functionality.

          John

          John Elwell
          Tel: +44 1908 817801 (office and mobile)
          Email: john.elwell@siemens-enterprise.com
          http://www.siemens-enterprise.com/uk/

          Siemens Enterprise Communications Limited.
          Registered office: Brickhill Street, Willen Lake,
Milton Keynes, MK15 0DJ.
          Registered No: 5903714, England.

          Siemens Enterprise Communications Limited is a
Trademark Licensee of Siemens AG.
          _______________________________________________
          rtcweb mailing list
          rtcweb@ietf.org
          https://www.ietf.org/mailman/listinfo/rtcweb



--
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo



--
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com<mailto:dyork@voxe=
o.com>
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo


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nk=3Dpurple style=3D'word-wrap: break-word;-webkit-nbsp-mode: space;-webkit=
-line-break: after-white-space'><div class=3DWordSection1><p class=3DMsoNor=
mal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";colo=
r:#1F497D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span style=3D'fon=
t-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'>Also agree that implementing a=
 full SIP Session Recording Client (SRC) as defined by the SIPREC WG in to =
the browser would be a tall order and would open the flood gates for a lot =
of other requests for SIP functionality in the browser.<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=3DMsoNor=
mal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";colo=
r:#1F497D'>If however SIP is not in the browser then I think it is a useful=
 and interesting test case to look at whether we could build a decomposed S=
RC with the browser handling the media and the web server handling the sign=
alling. What I heard a number of times at IETF81 is that what people want i=
s to build innovative new applications with a real time media enabled brows=
er and to it seems clear to me we need a browser API which is flexible and =
enables applications to do clever things with media streams. So exploring s=
ome use cases which require the browser to duplicate and copy media streams=
 is a good idea.&nbsp; <o:p></o:p></span></p><p class=3DMsoNormal><span sty=
le=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'>Therefore I think we =
should keep explore the remote recording use case further.<o:p></o:p></span=
></p><p class=3DMsoNormal><span style=3D'font-size:11.0pt;font-family:"Cali=
bri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMso=
Normal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";c=
olor:#1F497D'>Regards<o:p></o:p></span></p><p class=3DMsoNormal><span style=
=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Andy=
<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:7.0pt;font-family:"Arial",=
"sans-serif";color:#1F497D'>Andrew Hutton</span><span style=3D'font-size:8.=
0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p=
><p class=3DMsoNormal><span style=3D'font-size:8.0pt;font-family:Consolas;c=
olor:#1F497D'>Siemens Enterprise Communications Limited.<o:p></o:p></span><=
/p><p class=3DMsoNormal><span style=3D'font-size:8.0pt;font-family:Consolas=
;color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span styl=
e=3D'font-size:8.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Siem=
ens Enterprise Communications Limited is a Trademark Licensee of Siemens AG=
.<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:"Calib=
ri","sans-serif";color:#1F497D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoN=
ormal><span style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";co=
lor:#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'><o:p=
>&nbsp;</o:p></span></p><div style=3D'border:none;border-left:solid blue 1.=
5pt;padding:0cm 0cm 0cm 4.0pt'><div><div style=3D'border:none;border-top:so=
lid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=3DMsoNormal><b><span =
lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>F=
rom:</span></b><span lang=3DEN-US style=3D'font-size:10.0pt;font-family:"Ta=
homa","sans-serif"'> rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.or=
g] <b>On Behalf Of </b>Dan York<br><b>Sent:</b> 23 August 2011 16:49<br><b>=
To:</b> Elwell, John<br><b>Cc:</b> rtcweb@ietf.org; public-webrtc@w3.org<br=
><b>Subject:</b> Re: [rtcweb] Remote recording - RTC-Web client acting as S=
IPREC session recording client<o:p></o:p></span></p></div></div><p class=3D=
MsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>John,<o:p></o:p></p><di=
v><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=3DMsoNormal=
>You're welcome... and I agree with you and Paul on the need to have the di=
scussion and to consciously make a choice as to whether to include or not i=
nclude some functionality. &nbsp; I also appreciate that you took the time =
to draft up text for requirements to add to the use cases... it was seeing =
that new requirement text that made me start violently twitching here in my=
 home office and lured me out of lurking on the list. &nbsp;:-)<o:p></o:p><=
/p></div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><div><p class=
=3DMsoNormal>Dan<o:p></o:p></p></div><div><p class=3DMsoNormal><o:p>&nbsp;<=
/o:p></p></div><div><p class=3DMsoNormal>P.S. And I agree with your point b=
elow.<o:p></o:p></p></div><div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><d=
iv><div><p class=3DMsoNormal>On Aug 23, 2011, at 11:37 AM, Elwell, John wro=
te:<o:p></o:p></p></div><p class=3DMsoNormal><br><br><o:p></o:p></p><div><p=
 class=3DMsoNormal>Dan,<br><br>Thanks for your input. I agree with much of =
what you say. Like Paul, I think it is good to have the discussion, so that=
 we can at least understand whether or not we are going to specify browser =
support for remote recording and how such a decision has been reached. Howe=
ver, I would like to comment on one point below: <br><br><br><o:p></o:p></p=
><p class=3DMsoNormal>-----Original Message-----<o:p></o:p></p><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>From: D=
an York [mailto:dyork@voxeo.com] <o:p></o:p></p></blockquote><blockquote st=
yle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>Sent: 23 =
August 2011 16:23<o:p></o:p></p></blockquote><blockquote style=3D'margin-to=
p:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>To: Elwell, John<o:p></o:=
p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0p=
t'><p class=3DMsoNormal>Cc: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.=
org</a>; <a href=3D"mailto:public-webrtc@w3.org">public-webrtc@w3.org</a><o=
:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bott=
om:5.0pt'><p class=3DMsoNormal>Subject: Re: [rtcweb] Remote recording - RTC=
-Web client <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0=
pt;margin-bottom:5.0pt'><p class=3DMsoNormal>acting as SIPREC session recor=
ding client<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0p=
t;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal>John, <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5=
.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></block=
quote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=
=3DMsoNormal>Many thanks for the SIPREC background. I have not had the <o:p=
></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom=
:5.0pt'><p class=3DMsoNormal>cycles to follow that group and so your note h=
ere is <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;ma=
rgin-bottom:5.0pt'><p class=3DMsoNormal>extremely helpful. Thank you!<o:p><=
/o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5=
.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote st=
yle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>As I read=
 your summary though, I keep hearing &quot;Danger! <o:p></o:p></p></blockqu=
ote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DM=
soNormal>Danger!&quot; in my brain. I completely agree with this statement =
of yours:<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;=
margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote=
><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoN=
ormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.0=
pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Clearly there =
would be a fairly big hurdle for browsers <o:p></o:p></p></blockquote><bloc=
kquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>=
to support SRC functionality. <o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;<=
/o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5=
.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote st=
yle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>I think i=
t would be a very large hurdle and really takes us <o:p></o:p></p></blockqu=
ote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DM=
soNormal>back into basically baking a SIP UA into a web browser - and <o:p>=
</o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:=
5.0pt'><p class=3DMsoNormal>do we REALLY want to go down that road?<o:p></o=
:p></p></blockquote><p class=3DMsoNormal>[JRE] Some people are still talkin=
g about SIP in the browser, I believe. But one of the points I was making i=
s that, assuming SIP is NOT in the browser, there is somewhat less that nee=
ds to be done to provide browser support for an SRC (the rest would be up t=
o the web application).<br><br>John<br><br>John Elwell<br>Tel: +44 1908 817=
801 (office and mobile)<br>Email: <a href=3D"mailto:john.elwell@siemens-ent=
erprise.com">john.elwell@siemens-enterprise.com</a><br><a href=3D"http://ww=
w.siemens-enterprise.com/uk/">http://www.siemens-enterprise.com/uk/</a><br>=
<br>Siemens Enterprise Communications Limited.<br>Registered office: Brickh=
ill Street, Willen Lake, Milton Keynes, MK15 0DJ.<br>Registered No: 5903714=
, England.<br><br>Siemens Enterprise Communications Limited is a Trademark =
Licensee of Siemens AG.<br><br><br><o:p></o:p></p><p class=3DMsoNormal><o:p=
>&nbsp;</o:p></p><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'=
><p class=3DMsoNormal>If I go back to the charter of this group ( <o:p></o:=
p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0p=
t'><p class=3DMsoNormal>http://tools.ietf.org/wg/rtcweb/charters ) and why =
I started <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt=
;margin-bottom:5.0pt'><p class=3DMsoNormal>following it from the start and =
reading all the traffic, I <o:p></o:p></p></blockquote><blockquote style=3D=
'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>think we need t=
o focus on how we enable direct communication <o:p></o:p></p></blockquote><=
blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNor=
mal>between web browsers... or between web browsers and <o:p></o:p></p></bl=
ockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p clas=
s=3DMsoNormal>servers... but doing so in the most lightweight and easy way =
<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bo=
ttom:5.0pt'><p class=3DMsoNormal>possible. &nbsp;<o:p></o:p></p></blockquot=
e><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMso=
Normal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.=
0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>My interest has been in how w=
e can do real-time communication <o:p></o:p></p></blockquote><blockquote st=
yle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>*without*=
 extensions or plugins. &nbsp;Reading all of this, it's <o:p></o:p></p></bl=
ockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p clas=
s=3DMsoNormal>sounding like either we do need to have a plugin/extension to=
 <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-b=
ottom:5.0pt'><p class=3DMsoNormal>support SRC capability - or we need to ba=
ke a great amount of <o:p></o:p></p></blockquote><blockquote style=3D'margi=
n-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>functionality directl=
y into browsers. &nbsp;And that in my mind <o:p></o:p></p></blockquote><blo=
ckquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal=
>limits the number of browsers that might support all the <o:p></o:p></p></=
blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p cl=
ass=3DMsoNormal>capabilities of RTCWEB. &nbsp;<o:p></o:p></p></blockquote><=
blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNor=
mal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt=
;margin-bottom:5.0pt'><p class=3DMsoNormal>I think that given the aggressiv=
e timelines for the working <o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>group delive=
rables (and the market reality that the longer a <o:p></o:p></p></blockquot=
e><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMso=
Normal>&quot;standard&quot; solution takes to come out the more developers =
<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bo=
ttom:5.0pt'><p class=3DMsoNormal>will explore proprietary solutions), I agr=
ee with Stefan that <o:p></o:p></p></blockquote><blockquote style=3D'margin=
-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>we should put the reco=
rding out of scope for RTCWEB 1.0 and <o:p></o:p></p></blockquote><blockquo=
te style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>focu=
s on getting a solution out there that lets developers <o:p></o:p></p></blo=
ckquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=
=3DMsoNormal>start building RTCWEB apps. <o:p></o:p></p></blockquote><block=
quote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><=
o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;marg=
in-bottom:5.0pt'><p class=3DMsoNormal>For those environments that need reco=
rding (and I *do* <o:p></o:p></p></blockquote><blockquote style=3D'margin-t=
op:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>understand the call cent=
er need), middleboxes can provide a <o:p></o:p></p></blockquote><blockquote=
 style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>soluti=
on today - or some vendors can support the RTCWEB <o:p></o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal>communication and *also* provide a recording capability. <o:p></o:p=
></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt=
'><p class=3DMsoNormal>Sure, that's not ideal... and yes, we need to make s=
ure that <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;=
margin-bottom:5.0pt'><p class=3DMsoNormal>what we do for RTCWEB 1.0 doesn't=
 preclude adding recording <o:p></o:p></p></blockquote><blockquote style=3D=
'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>to a RTCWEB 2.0=
... &nbsp;but we need to get a spec out there that <o:p></o:p></p></blockqu=
ote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DM=
soNormal>will be useful to the majority of developers and very easy <o:p></=
o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.=
0pt'><p class=3DMsoNormal>for them to adopt. &nbsp;<o:p></o:p></p></blockqu=
ote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DM=
soNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:=
5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>The more complicated we mak=
e it - or the more requirements we <o:p></o:p></p></blockquote><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>impose =
on browsers - the less adoption we'll see.<o:p></o:p></p></blockquote><bloc=
kquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>=
<o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;mar=
gin-bottom:5.0pt'><p class=3DMsoNormal>My 2 cents,<o:p></o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal>Dan<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0p=
t;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5=
.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>On Aug 23, 2011, at 3:58 AM,=
 Elwell, John wrote:<o:p></o:p></p></blockquote><blockquote style=3D'margin=
-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><=
/blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p c=
lass=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'mar=
gin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple=
-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>The=
re has been some discussion recently on remote <o:p></o:p></p></blockquote>=
<blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNo=
rmal>recording, mixed to some extent with discussions on local <o:p></o:p><=
/p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>=
<p class=3DMsoNormal>recording and with mailbox, but I would like to focus =
on <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin=
-bottom:5.0pt'><p class=3DMsoNormal>remote recording and try to summarize.<=
o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bot=
tom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><o:p></o:p></p></blockquote=
><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoN=
ormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; </span>First, some background on the IETF SIPREC WG. This is=
 <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-b=
ottom:5.0pt'><p class=3DMsoNormal>specifying support for SIP-based session =
recording, whereby a <o:p></o:p></p></blockquote><blockquote style=3D'margi=
n-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>Session Recording Cli=
ent (SRC) on the path of a call <o:p></o:p></p></blockquote><blockquote sty=
le=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>(communica=
tion session) can forward media and metadata to a <o:p></o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal>session recording server (SRS) or recording device. In <o:p></o:p><=
/p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>=
<p class=3DMsoNormal>conventional SIP terms, the SRC can exist at an endpoi=
nt of <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;mar=
gin-bottom:5.0pt'><p class=3DMsoNormal>the communication session being reco=
rded (i.e., at a SIP UA), <o:p></o:p></p></blockquote><blockquote style=3D'=
margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>or at a B2BUA th=
at has access to the media as well as the <o:p></o:p></p></blockquote><bloc=
kquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>=
signalling. Very often in a contact centre, there are <o:p></o:p></p></bloc=
kquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=
=3DMsoNormal>mandatory requirements for recording some or all <o:p></o:p></=
p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><=
p class=3DMsoNormal>communication sessions, and often calls are routed thro=
ugh a <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;mar=
gin-bottom:5.0pt'><p class=3DMsoNormal>B2BUA that also provides the SRC. So=
 in this case there is no <o:p></o:p></p></blockquote><blockquote style=3D'=
margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>responsibility o=
n SIP UAs to support SRC functionality, and <o:p></o:p></p></blockquote><bl=
ockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNorma=
l>no issues of additional bandwidth on the device's access. <o:p></o:p></p>=
</blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal>However, it is anticipated in SIPREC that in some deploym=
ents <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;marg=
in-bottom:5.0pt'><p class=3DMsoNormal>UA-located SRCs will be used. How a U=
A is organized <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:=
5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>internally to provide SRC f=
unctionality is not standardized.<o:p></o:p></p></blockquote><blockquote st=
yle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span cla=
ss=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
</span><o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;ma=
rgin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>So the question for=
 RTC-Web is whether a SIP UA <o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>implemented =
as an RTC-Web client can provide SRC <o:p></o:p></p></blockquote><blockquot=
e style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>funct=
ionality, i.e., support remote recording. An RTC-Web SIP <o:p></o:p></p></b=
lockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p cla=
ss=3DMsoNormal>UA is implemented by a combination of functionality running =
<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bo=
ttom:5.0pt'><p class=3DMsoNormal>on the web server, functionality running i=
n client side <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5=
.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>script (JS) and functionalit=
y embedded in the browser. The <o:p></o:p></p></blockquote><blockquote styl=
e=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>amount of f=
unctionality needed in the browser and needing to <o:p></o:p></p></blockquo=
te><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMs=
oNormal>be exposed at the browser API in support of SRC will depend <o:p></=
o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.=
0pt'><p class=3DMsoNormal>to some extent on how much core functionality goe=
s into the <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0p=
t;margin-bottom:5.0pt'><p class=3DMsoNormal>browser, in particular whether =
the browser implements SIP or <o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>not. Some of=
 the functions noted to date include:<o:p></o:p></p></blockquote><blockquot=
e style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span=
 class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; </span>- ability to take a copy of streams sent to / received <o:p></o:=
p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0p=
t'><p class=3DMsoNormal>from the remote party and send them, in real-time, =
to a <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;marg=
in-bottom:5.0pt'><p class=3DMsoNormal>remote recording device (SRS);<o:p></=
o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.=
0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- possible need to mix the copied=
 streams before <o:p></o:p></p></blockquote><blockquote style=3D'margin-top=
:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>sending (e.g., mix the sen=
t and received audio streams)<o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=
=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </=
span>- possible need to use a different codec or other <o:p></o:p></p></blo=
ckquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=
=3DMsoNormal>parameters when sending to the SRS;<o:p></o:p></p></blockquote=
><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoN=
ormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp; </span>- possible need to use a different encryption/integri=
ty <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin=
-bottom:5.0pt'><p class=3DMsoNormal>context when sending to the SRS;<o:p></=
o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.=
0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- possible need to insert tones /=
 announcements into <o:p></o:p></p></blockquote><blockquote style=3D'margin=
-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>the original media pat=
h being recorded;<o:p></o:p></p></blockquote><blockquote style=3D'margin-to=
p:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-s=
pan>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- possibl=
e need to support SDP enhancements for <o:p></o:p></p></blockquote><blockqu=
ote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>ind=
icating media that are being recorded or preferences for <o:p></o:p></p></b=
lockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p cla=
ss=3DMsoNormal>which media are being recorded;<o:p></o:p></p></blockquote><=
blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNor=
mal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp; </span>- possible need to support SIP enhancements for <o:p></=
o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.=
0pt'><p class=3DMsoNormal>indicating SRC/SRS capability and recording aware=
ness (if SIP <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.=
0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>is implemented in browser);<o=
:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bott=
om:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>- possible need to support t=
he sending of metadata to <o:p></o:p></p></blockquote><blockquote style=3D'=
margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>the SRS (if SIP =
is implemented in browser).<o:p></o:p></p></blockquote><blockquote style=3D=
'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Da=
pple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span=
><o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-b=
ottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Clearly there would be a =
fairly big hurdle for browsers <o:p></o:p></p></blockquote><blockquote styl=
e=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>to support =
SRC functionality. But without this, RTC-Web <o:p></o:p></p></blockquote><b=
lockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNorm=
al>clients would not be suitable for use in environments where <o:p></o:p><=
/p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'>=
<p class=3DMsoNormal>remote recording is required and calls are not forced =
through <o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;m=
argin-bottom:5.0pt'><p class=3DMsoNormal>some middlebox that provides SRC f=
unctionality.<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.=
0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span><o:p></o:p></=
p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><=
p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>John<o:p></o:p></p></blockquote><blockq=
uote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><s=
pan class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp; </span><o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5=
.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span=
>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>John Elwell<=
o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bot=
tom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Tel: +44 1908 817801 (offic=
e and mobile)<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.=
0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-span>=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Email: john.e=
lwell@siemens-enterprise.com<o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=
=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </=
span>http://www.siemens-enterprise.com/uk/<o:p></o:p></p></blockquote><bloc=
kquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>=
<span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp; </span><o:p></o:p></p></blockquote><blockquote style=3D'margin-top=
:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-sp=
an>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Siemens En=
terprise Communications Limited.<o:p></o:p></p></blockquote><blockquote sty=
le=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span clas=
s=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <=
/span>Registered office: Brickhill Street, Willen Lake, <o:p></o:p></p></bl=
ockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p clas=
s=3DMsoNormal>Milton Keynes, MK15 0DJ.<o:p></o:p></p></blockquote><blockquo=
te style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><spa=
n class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp; </span>Registered No: 5903714, England.<o:p></o:p></p></blockquote><bl=
ockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNorma=
l><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; </span><o:p></o:p></p></blockquote><blockquote style=3D'margin-t=
op:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=3Dapple-tab-=
span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </span>Siemens =
Enterprise Communications Limited is a <o:p></o:p></p></blockquote><blockqu=
ote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>Tra=
demark Licensee of Siemens AG.<o:p></o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span class=
=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; </=
span>_______________________________________________<o:p></o:p></p></blockq=
uote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3D=
MsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; </span>rtcweb mailing list<o:p></o:p></p></blockquote><b=
lockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNorm=
al><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; </span>rtcweb@ietf.org<o:p></o:p></p></blockquote><blockquote s=
tyle=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><span cl=
ass=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
 </span>https://www.ietf.org/mailman/listinfo/rtcweb<o:p></o:p></p></blockq=
uote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3D=
MsoNormal><span class=3Dapple-tab-span>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp; </span><o:p></o:p></p></blockquote><blockquote style=3D'=
margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p=
></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt=
'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=
=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>-- <o:p></o:=
p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0p=
t'><p class=3DMsoNormal>Dan York, CISSP, Director of Conversations<o:p></o:=
p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0p=
t'><p class=3DMsoNormal>Voxeo Corporation &nbsp;&nbsp;http://www.voxeo.com =
&nbsp;dyork@voxeo.com<o:p></o:p></p></blockquote><blockquote style=3D'margi=
n-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNormal>Phone: +1-321-710-919=
3 &nbsp;skype: danyork &nbsp;sip:dyork@voxeo.com<o:p></o:p></p></blockquote=
><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoN=
ormal><o:p>&nbsp;</o:p></p></blockquote><blockquote style=3D'margin-top:5.0=
pt;margin-bottom:5.0pt'><p class=3DMsoNormal>Join the Voxeo conversation:<o=
:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;margin-bott=
om:5.0pt'><p class=3DMsoNormal>Blogs: http://blogs.voxeo.com<o:p></o:p></p>=
</blockquote><blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal>Twitter: http://twitter.com/voxeo &nbsp;http://twitter.co=
m/danyork<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;=
margin-bottom:5.0pt'><p class=3DMsoNormal>Facebook: http://www.facebook.com=
/voxeo<o:p></o:p></p></blockquote><blockquote style=3D'margin-top:5.0pt;mar=
gin-bottom:5.0pt'><p class=3DMsoNormal><o:p>&nbsp;</o:p></p></blockquote><b=
lockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p class=3DMsoNorm=
al><o:p>&nbsp;</o:p></p></blockquote></div></div><p class=3DMsoNormal><o:p>=
&nbsp;</o:p></p><div><div><div><div><div><div><div><div><div><div><div><div=
><div><div><p class=3DMsoNormal><span style=3D'font-size:9.0pt;font-family:=
"Helvetica","sans-serif";color:black'>--&nbsp;<o:p></o:p></span></p></div><=
div><div><p class=3DMsoNormal><span style=3D'font-size:9.0pt;font-family:"H=
elvetica","sans-serif";color:black'>Dan York, CISSP, Director of Conversati=
ons<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span style=3D'fon=
t-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'>Voxeo Corpor=
ation&nbsp;&nbsp;&nbsp;<a href=3D"http://www.voxeo.com">http://www.voxeo.co=
m</a>&nbsp;&nbsp;<a href=3D"mailto:dyork@voxeo.com">dyork@voxeo.com</a><o:p=
></o:p></span></p></div><div><p class=3DMsoNormal><span style=3D'font-size:=
9.0pt;font-family:"Helvetica","sans-serif";color:black'>Phone: +1-321-710-9=
193 &nbsp;skype: danyork &nbsp;<a href=3D"sip:dyork@voxeo.com">sip:dyork@vo=
xeo.com</a><o:p></o:p></span></p></div></div><div><p class=3DMsoNormal><spa=
n style=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black=
'><o:p>&nbsp;</o:p></span></p></div><div><div><p class=3DMsoNormal><b><span=
 style=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif";color:#2400B=
B'>Join the Voxeo conversation:</span></b><span style=3D'font-size:9.0pt;fo=
nt-family:"Helvetica","sans-serif";color:black'><o:p></o:p></span></p></div=
><div><p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"H=
elvetica","sans-serif";color:#2400BB'>Blogs: <a href=3D"http://blogs.voxeo.=
com">http://blogs.voxeo.com</a></span></b><span style=3D'font-size:9.0pt;fo=
nt-family:"Helvetica","sans-serif";color:black'><o:p></o:p></span></p></div=
><div><p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-family:"H=
elvetica","sans-serif";color:#2400BB'>Twitter: <a href=3D"http://twitter.co=
m/voxeo">http://twitter.com/voxeo</a> &nbsp;<a href=3D"http://twitter.com/d=
anyork">http://twitter.com/danyork</a></span></b><span style=3D'font-size:9=
.0pt;font-family:"Helvetica","sans-serif";color:black'><o:p></o:p></span></=
p></div><div><p class=3DMsoNormal><b><span style=3D'font-size:9.0pt;font-fa=
mily:"Helvetica","sans-serif";color:#2400BB'>Facebook: <a href=3D"http://ww=
w.facebook.com/voxeo">http://www.facebook.com/voxeo</a></span></b><span sty=
le=3D'font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'><o:=
p></o:p></span></p></div></div></div></div></div></div></div></div></div></=
div></div></div></div></div></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p=
></div></div></div></body></html>=

--_000_101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2MCHP058Agloba_--

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From: Dan York <dyork@voxeo.com>
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Subject: [rtcweb] Relation of Mozilla WebAPI announced today to RTCWEB/WebRTC work?
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--Apple-Mail-17-807351045
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Can anyone from Mozilla please comment on how the "WebAPI" announced =
today with the goal to " provide a basic HTML5 phone experience within 3 =
to 6 months" relates to the work happening in RTCWEB/WebRTC?

http://hacks.mozilla.org/2011/08/introducing-webapi/

https://wiki.mozilla.org/WebAPI
=09
It's not clear from the blog entry or site whether this is a project =
that will in fact be implementing RTCWEB/WebRTC ideas (as part of the =
larger framework) or if this is something separate?

Thanks,
Dan

--=20
Dan York, CISSP, Director of Conversations
Voxeo Corporation   http://www.voxeo.com  dyork@voxeo.com
Phone: +1-321-710-9193  skype: danyork  sip:dyork@voxeo.com

Join the Voxeo conversation:
Blogs: http://blogs.voxeo.com
Twitter: http://twitter.com/voxeo  http://twitter.com/danyork
Facebook: http://www.facebook.com/voxeo


--Apple-Mail-17-807351045
Content-Transfer-Encoding: quoted-printable
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	charset=us-ascii

<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Can =
anyone from Mozilla please comment on how the "WebAPI" announced today =
with the goal to "&nbsp;provide a basic&nbsp;HTML5&nbsp;phone experience =
within 3 to 6 months" relates to the work happening in =
RTCWEB/WebRTC?<div><br></div><div><a =
href=3D"http://hacks.mozilla.org/2011/08/introducing-webapi/">http://hacks=
.mozilla.org/2011/08/introducing-webapi/</a></div><div><br></div><div><a =
href=3D"https://wiki.mozilla.org/WebAPI">https://wiki.mozilla.org/WebAPI</=
a></div><div><span class=3D"Apple-tab-span" style=3D"white-space:pre">	=
</span></div><div>It's not clear from the blog entry or site whether =
this is a project that will in fact be implementing RTCWEB/WebRTC ideas =
(as part of the larger framework) or if this is something =
separate?</div><div><br></div><div>Thanks,</div><div>Dan</div><div><br><di=
v>
<div style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space; "><div style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; color: =
rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: =
normal; font-weight: normal; letter-spacing: normal; line-height: =
normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: =
normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-indent: 0px; text-transform: none; white-space: normal; widows: 2; =
word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; font-size: medium; "><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-size: medium; font-style: normal; font-variant: normal; =
font-weight: normal; letter-spacing: normal; line-height: normal; =
orphans: 2; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><div style=3D"word-wrap: =
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From prvs=20974d5c9=tterriberry@mozilla.com  Tue Aug 23 10:58:57 2011
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Subject: Re: [rtcweb] Relation of Mozilla WebAPI announced today to	RTCWEB/WebRTC work?
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> Can anyone from Mozilla please comment on how the "WebAPI" announced
> today with the goal to " provide a basic HTML5 phone experience within 3
> to 6 months" relates to the work happening in RTCWEB/WebRTC?

This is more along the lines of getting access to the APIs needed to 
instruct a mobile phone to initiate a phone call, rather than to host 
that call inside the browser. Presumably also with some way to detect 
that your on a device that supports that capability at all.

From harald@alvestrand.no  Tue Aug 23 11:27:57 2011
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On 08/23/11 19:59, Timothy B. Terriberry wrote:
>> Can anyone from Mozilla please comment on how the "WebAPI" announced
>> today with the goal to " provide a basic HTML5 phone experience within 3
>> to 6 months" relates to the work happening in RTCWEB/WebRTC?
>
> This is more along the lines of getting access to the APIs needed to 
> instruct a mobile phone to initiate a phone call, rather than to host 
> that call inside the browser. Presumably also with some way to detect 
> that your on a device that supports that capability at all.
The blog post sounds like a virtual clone of the W3C Device API charter.

Is that your planned venue for standardizing these APIs?

I can't say I'm overly happy with the amount of imagination spent on the 
name, but the effort seems to make sense!

                      Harald


From blizzard@mozilla.com  Tue Aug 23 11:32:45 2011
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On 8/23/2011 11:28 AM, Harald Alvestrand wrote:
> On 08/23/11 19:59, Timothy B. Terriberry wrote:
>>> Can anyone from Mozilla please comment on how the "WebAPI" announced
>>> today with the goal to " provide a basic HTML5 phone experience 
>>> within 3
>>> to 6 months" relates to the work happening in RTCWEB/WebRTC?
>>
>> This is more along the lines of getting access to the APIs needed to 
>> instruct a mobile phone to initiate a phone call, rather than to host 
>> that call inside the browser. Presumably also with some way to detect 
>> that your on a device that supports that capability at all.
> The blog post sounds like a virtual clone of the W3C Device API charter.
>
> Is that your planned venue for standardizing these APIs?
>
> I can't say I'm overly happy with the amount of imagination spent on 
> the name, but the effort seems to make sense!

We're not part of the device API working group (only Opera participates 
in that group from the major browser vendors.)

--Chris

--Chris

From prvs=20974d5c9=tterriberry@mozilla.com  Tue Aug 23 11:40:59 2011
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Harald Alvestrand wrote:
> On 08/23/11 19:59, Timothy B. Terriberry wrote:
>>> Can anyone from Mozilla please comment on how the "WebAPI" announced
>>> today with the goal to " provide a basic HTML5 phone experience within 3
>>> to 6 months" relates to the work happening in RTCWEB/WebRTC?
>>
>> This is more along the lines of getting access to the APIs needed to
>> instruct a mobile phone to initiate a phone call, rather than to host
>> that call inside the browser. Presumably also with some way to detect
>> that your on a device that supports that capability at all.
> The blog post sounds like a virtual clone of the W3C Device API charter.
>
> Is that your planned venue for standardizing these APIs?

I'm not sure that's been worked out yet, but before I say too many 
incorrect things, let me CC Jonas so he can correct me.

> I can't say I'm overly happy with the amount of imagination spent on the
> name, but the effort seems to make sense!

Well, I didn't name the thing, but keep in mind that initiating phone 
calls is only one small part of the total effort.

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Tim,

On Aug 23, 2011, at 1:59 PM, Timothy B. Terriberry wrote:

>> Can anyone from Mozilla please comment on how the "WebAPI" announced
>> today with the goal to " provide a basic HTML5 phone experience =
within 3
>> to 6 months" relates to the work happening in RTCWEB/WebRTC?
>=20
> This is more along the lines of getting access to the APIs needed to =
instruct a mobile phone to initiate a phone call, rather than to host =
that call inside the browser. Presumably also with some way to detect =
that your on a device that supports that capability at all.


Thanks for the reply... and yes that makes sense.  I was thinking of "a =
basic HTML5 phone experience" as having a real-time "phone" conversation =
via HTML5, rather than putting pieces into HTML5 to have it work better =
on phone *devices*.

Thanks,
Dan

--=20
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Tim,<div><br><div><div>On Aug 23, 2011, at 1:59 PM, Timothy B. =
Terriberry wrote:</div><br class=3D"Apple-interchange-newline"><blockquote=
 type=3D"cite"><div><blockquote type=3D"cite">Can anyone from Mozilla =
please comment on how the "WebAPI" announced<br></blockquote><blockquote =
type=3D"cite">today with the goal to " provide a basic HTML5 phone =
experience within 3<br></blockquote><blockquote type=3D"cite">to 6 =
months" relates to the work happening in =
RTCWEB/WebRTC?<br></blockquote><br>This is more along the lines of =
getting access to the APIs needed to instruct a mobile phone to initiate =
a phone call, rather than to host that call inside the browser. =
Presumably also with some way to detect that your on a device that =
supports that capability at all.<br></div></blockquote></div><div><br =
class=3D"webkit-block-placeholder"></div><div>Thanks for the reply... =
and yes that makes sense. &nbsp;I was thinking of "a basic HTML5 phone =
experience" as having a real-time "phone" conversation via HTML5, rather =
than putting pieces into HTML5 to have it work better on phone =
*devices*.</div><div><br></div><div>Thanks,</div><div>Dan</div><div><br =
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On 8/23/2011 1:50 PM, Hutton, Andrew wrote:


> ...

> What I heard a number of times at IETF81 is that what people want is 
> to build innovative new applications with a real time media enabled 
> browser and to it seems clear to me we need a browser API which is 
> flexible and enables applications to do clever things with media 
> streams. So exploring some use cases which require the browser to 
> duplicate and copy media streams is a good idea.
Indeed the browser API must be flexible and it must enable 
applications.  Strong ++ on that.

Inasmuch as we, in the IETF, are not really in control of API, to ensure 
its flexibility we can only make the right protocol selection.   To this 
end, the worst thing would be trying to reinvent the wheel.  People have 
been working on SIPREC for more than a year now--would it be not wise on 
our part to take the output of that work?

Overall, even if the browser does not support the full-blown SIP, I 
think it must support some parts of it (REFER and SUBSCRIBE/NOTIFY, for 
one thing) in order to enable flexibility.

A few thousand messages ago, I asked how (asynchronous) notifications 
could be implemented without having SIP in the browser, and I don't 
think we had a sufficient discussion.

Igor
>
>

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    <br>
    <br>
    On 8/23/2011 1:50 PM, Hutton, Andrew wrote:<br>
    <br>
    <br>
    <blockquote
cite="mid:101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net"
      type="cite">
      <div class="WordSection1"><span style="font-size: 11pt;
          font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
          color: rgb(31, 73, 125);"> ...</span></div>
    </blockquote>
    <br>
    <blockquote
cite="mid:101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net"
      type="cite">
      <div class="WordSection1"><span style="font-size: 11pt;
          font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
          color: rgb(31, 73, 125);">What I heard a number of times at
          IETF81 is that what people want is to build innovative new
          applications with a real time media enabled browser and to it
          seems clear to me we need a browser API which is flexible and
          enables applications to do clever things with media streams.
          So exploring some use cases which require the browser to
          duplicate and copy media streams is a good idea. <br>
        </span></div>
    </blockquote>
    Indeed the browser API must be flexible and it must enable
    applications.&nbsp; Strong ++ on that.&nbsp; <br>
    <br>
    Inasmuch as we, in the IETF, are not really in control of API, to
    ensure its flexibility we can only make the right protocol
    selection.&nbsp;&nbsp; To this end, the worst thing would be trying to
    reinvent the wheel.&nbsp; People have been working on SIPREC for more
    than a year now--would it be not wise on our part to take the output
    of that work?<br>
    <br>
    Overall, even if the browser does not support the full-blown SIP, I
    think it must support some parts of it (REFER and SUBSCRIBE/NOTIFY,
    for one thing) in order to enable flexibility.<br>
    <br>
    A few thousand messages ago, I asked how (asynchronous)
    notifications could be implemented without having SIP in the
    browser, and I don't think we had a sufficient discussion.<br>
    <br>
    Igor<br>
    <blockquote
cite="mid:101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net"
      type="cite">
      <div class="WordSection1"><span style="font-size: 11pt;
          font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
          color: rgb(31, 73, 125);"> <o:p></o:p></span>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
      </div>
      <br>
    </blockquote>
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From fluffy@cisco.com  Tue Aug 23 14:47:44 2011
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From: Cullen Jennings <fluffy@cisco.com>
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To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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Just to try and help clarify requirements ... I suspect the current =
requirements you wrote could be meant mostly by just taking a copy of =
the data to and from the speaker/mic/camera/screen and just sending them =
via a HTTP post to some web service that recorded them. I suspect you =
want something different than that so we might need to poke at these =
requirements a bit to see what it is. Alternatively, if data can be =
copied from one stream to anther stream using the JS, does that provide =
everything you need or is it something more.=20

(BTW... I am probably not in favor of supporting these requirements in =
the client as it is typically not a great place to do recording but =
that's a separate issues. Just trying to clarify what we talking about)

On Aug 22, 2011, at 8:42 AM, Elwell, John wrote:

> 4.2.yy Remote Session Recording
> In this use case, the web application user wishes to record a =
real-time communication at a remote recording device, such that =
transmitted and received audio, video or other real-time media are =
transmitted in real-time to the remote device. The remote device can =
perform archiving, retrieval, playback, etc., but can also perform =
real-time analytics on the media. A typical deployment might be in a =
contact centre. For a given medium, the two directions of transmission =
can be transmitted together (mixed) or separately. The web application =
also sends metadata that gives context to the stored media.
>=20
> New requirements:
> Fyy1: The browser MUST be able to send in real-time to a remote =
recording device media that are being transmitted to and received from =
remote participants.
>=20
> Ayy1: The web application MUST be able to ask the browser to transmit =
in real-time to a remote recording device media that are being =
transmitted to and received from remote participants and, in the case of =
audio at least, ask for the two directions of transmission to be =
transmitted to the remote recording device mixed or separately.
>=20
> John
>=20
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 =
0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of =
Siemens AG.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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To: "Elwell, John" <john.elwell@siemens-enterprise.com>
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On Aug 22, 2011, at 8:42 AM, Elwell, John wrote:

> 4.2.xx Local Session Recording
> In this use case, the web application user wishes to record a =
real-time communication locally, such that transmitted and received =
audio, video or other real-time media are stored in one or more files. =
For a given medium, the two directions of transmission can be stored in =
the same file (mixed) or in separate files. The web application also =
stores metadata that gives context to the stored media.
>=20
> New requirements:
> Fxx1: The browser MUST be able to store transmitted and received media =
in local files.
>=20
> Axx1: The web application MUST be able to ask the browser to store =
transmitted and received media in local files, and in the case of audio =
at least, ask for the two directions of transmission to be stored mixed =
or separately.
>=20
> John

I think I want something just slightly different and that is to send =
media that is data in the JS and to be able to hand media as data to the =
JS. If the JS happens to read / write that to a file that is fine with =
me but this allows the JS to operate on the data or synthetically =
generate media and opens the doors to the possibility of writing a codec =
in JS or NACL.=20=

From pravindran@sonusnet.com  Tue Aug 23 17:56:54 2011
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Subject: [rtcweb] SIP MUST NOT be used in browser?[was RE: Remote recording - RTC-Web client acting as	SIPREC	session recording client]
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I agree with Igor.  It will be good in case the discussion in the line
why SIP MUST NOT be used for browser. I guess that the following are not
the reason for not selecting SIP:

=20

1)       Memory usage in browser is not the limiting factor for SIP. I
know of the SIP application which works well in embedded device with
16MB RAM.

2)      In case the idea is to use the simple alternative protocol for
the basic dialog between browser to browser. It is possible for the same
protocol to become as complex as SIP in 5-10 years while adding more
feature or services.  We end-up with two protocol for real-time
communication from IETF and interact through gateways.=20

3)      As per IETF recommendation,  standalone application in endpoint
has to use SIP for real-time communication whereas the same application
running in browser has to use RTCWeb recommended protocol. It is tough
for each application to support both protocols or its related
infrastructure to make it work. Or the intention is not to use SIP
anywhere in endpoint?

=20

I'm interested in hearing the reasons for why SIP MUST NOT be used in
browser.

=20

Thanks

Partha

=20

From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf
Of Igor Faynberg
Sent: Wednesday, August 24, 2011 2:09 AM
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC
session recording client

=20



On 8/23/2011 1:50 PM, Hutton, Andrew wrote:





...





What I heard a number of times at IETF81 is that what people want is to
build innovative new applications with a real time media enabled browser
and to it seems clear to me we need a browser API which is flexible and
enables applications to do clever things with media streams. So
exploring some use cases which require the browser to duplicate and copy
media streams is a good idea.=20

Indeed the browser API must be flexible and it must enable applications.
Strong ++ on that. =20

Inasmuch as we, in the IETF, are not really in control of API, to ensure
its flexibility we can only make the right protocol selection.   To this
end, the worst thing would be trying to reinvent the wheel.  People have
been working on SIPREC for more than a year now--would it be not wise on
our part to take the output of that work?

Overall, even if the browser does not support the full-blown SIP, I
think it must support some parts of it (REFER and SUBSCRIBE/NOTIFY, for
one thing) in order to enable flexibility.

A few thousand messages ago, I asked how (asynchronous) notifications
could be implemented without having SIP in the browser, and I don't
think we had a sufficient discussion.

Igor



=20

=20


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<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>I agree with Igor. &nbsp;It will be good in case the =
discussion
in the line why SIP MUST NOT be used for browser. I guess that the =
following
are not the reason for not selecting SIP:<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=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>1)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>&nbsp;Memory usage in browser is not the limiting factor =
for SIP.
I know of the SIP application which works well in embedded device with =
16MB
RAM.<o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>2)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>In case the idea is to use the simple alternative =
protocol for
the basic dialog between browser to browser. It is possible for the same
protocol to become as complex as SIP in 5-10 years while adding more =
feature or
services. &nbsp;We end-up with two protocol for real-time communication =
from
IETF and interact through gateways. <o:p></o:p></span></p>

<p class=3DMsoListParagraph style=3D'text-indent:-.25in;mso-list:l0 =
level1 lfo1'><![if !supportLists]><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><span
style=3D'mso-list:Ignore'>3)<span style=3D'font:7.0pt "Times New =
Roman"'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span></span><![endif]><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>As per IETF recommendation,&nbsp; standalone application =
in
endpoint has to use SIP for real-time communication whereas the same =
application
running in browser has to use RTCWeb recommended protocol. It is tough =
for each
application to support both protocols or its related infrastructure to =
make it
work. Or the intention is not to use SIP anywhere in =
endpoint?<o:p></o:p></span></p>

<p class=3DMsoNormal style=3D'margin-left:.25in'><span =
style=3D'font-size:11.0pt;
font-family:"Calibri","sans-serif";color:#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'>I&#8217;m interested in hearing the reasons for why SIP =
MUST NOT
be used in browser.<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'>Thanks<o:p></o:p></span></p>

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

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

<div 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=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span =
style=3D'font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> rtcweb-bounces@ietf.org
[mailto:rtcweb-bounces@ietf.org] <b>On Behalf Of </b>Igor Faynberg<br>
<b>Sent:</b> Wednesday, August 24, 2011 2:09 AM<br>
<b>To:</b> rtcweb@ietf.org<br>
<b>Subject:</b> Re: [rtcweb] Remote recording - RTC-Web client acting as =
SIPREC
session recording client<o:p></o:p></span></p>

</div>

</div>

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

<p class=3DMsoNormal><br>
<br>
On 8/23/2011 1:50 PM, Hutton, Andrew wrote:<br>
<br>
<br>
<br>
<o:p></o:p></p>

<div>

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

</div>

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

<div>

<p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>What I heard a number of times at IETF81 is that what =
people
want is to build innovative new applications with a real time media =
enabled
browser and to it seems clear to me we need a browser API which is =
flexible and
enables applications to do clever things with media streams. So =
exploring some
use cases which require the browser to duplicate and copy media streams =
is a
good idea. </span><o:p></o:p></p>

</div>

<p class=3DMsoNormal>Indeed the browser API must be flexible and it must =
enable
applications.&nbsp; Strong ++ on that.&nbsp; <br>
<br>
Inasmuch as we, in the IETF, are not really in control of API, to ensure =
its
flexibility we can only make the right protocol selection.&nbsp;&nbsp; =
To this
end, the worst thing would be trying to reinvent the wheel.&nbsp; People =
have
been working on SIPREC for more than a year now--would it be not wise on =
our
part to take the output of that work?<br>
<br>
Overall, even if the browser does not support the full-blown SIP, I =
think it
must support some parts of it (REFER and SUBSCRIBE/NOTIFY, for one =
thing) in
order to enable flexibility.<br>
<br>
A few thousand messages ago, I asked how (asynchronous) notifications =
could be
implemented without having SIP in the browser, and I don't think we had =
a
sufficient discussion.<br>
<br>
Igor<br>
<br>
<o:p></o:p></p>

<div>

<p class=3DMsoNormal =
style=3D'mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>&nbsp;</spa=
n><o:p></o:p></p>

</div>

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

</div>

</div>

</body>

</html>

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From dzonatas@gmail.com  Tue Aug 23 17:58:19 2011
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Native Client mode needs Native Server mode as different accounts local 
and remote; like, six hours worth of samples from physical grid 
simulation; scene data driven graphs store optional.

Geolocation schema for C.A.V.E.s was one of many common questions never 
made FAQ.

On 08/23/2011 02:56 PM, Cullen Jennings wrote:
> On Aug 22, 2011, at 8:42 AM, Elwell, John wrote:
>
>    
>> 4.2.xx Local Session Recording
>> In this use case, the web application user wishes to record a real-time communication locally, such that transmitted and received audio, video or other real-time media are stored in one or more files. For a given medium, the two directions of transmission can be stored in the same file (mixed) or in separate files. The web application also stores metadata that gives context to the stored media.
>>
>> New requirements:
>> Fxx1: The browser MUST be able to store transmitted and received media in local files.
>>
>> Axx1: The web application MUST be able to ask the browser to store transmitted and received media in local files, and in the case of audio at least, ask for the two directions of transmission to be stored mixed or separately.
>>
>> John
>>      
> I think I want something just slightly different and that is to send media that is data in the JS and to be able to hand media as data to the JS. If the JS happens to read / write that to a file that is fine with me but this allows the JS to operate on the data or synthetically generate media and opens the doors to the possibility of writing a codec in JS or NACL.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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Actually, I don't think that the "must not" decision has ever been made 
by the WG. In fact, two implementation reports from the last meeting 
cited SIP implementation...

I actually thought that the need for supporting SIP in the browser was 
well articulated in Quebec...

Igor



On 8/23/2011 8:57 PM, Ravindran Parthasarathi wrote:
>
> I agree with Igor.  It will be good in case the discussion in the line 
> why SIP MUST NOT be used for browser. I guess that the following are 
> not the reason for not selecting SIP:
>
> 1) Memory usage in browser is not the limiting factor for SIP. I know 
> of the SIP application which works well in embedded device with 16MB RAM.
>
> 2)In case the idea is to use the simple alternative protocol for the 
> basic dialog between browser to browser. It is possible for the same 
> protocol to become as complex as SIP in 5-10 years while adding more 
> feature or services.  We end-up with two protocol for real-time 
> communication from IETF and interact through gateways.
>
> 3)As per IETF recommendation,  standalone application in endpoint has 
> to use SIP for real-time communication whereas the same application 
> running in browser has to use RTCWeb recommended protocol. It is tough 
> for each application to support both protocols or its related 
> infrastructure to make it work. Or the intention is not to use SIP 
> anywhere in endpoint?
>
> I'm interested in hearing the reasons for why SIP MUST NOT be used in 
> browser.
>
> Thanks
>
> Partha
>
> *From:*rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] *On 
> Behalf Of *Igor Faynberg
> *Sent:* Wednesday, August 24, 2011 2:09 AM
> *To:* rtcweb@ietf.org
> *Subject:* Re: [rtcweb] Remote recording - RTC-Web client acting as 
> SIPREC session recording client
>
>
>
> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>
>
>
> ...
>
>
>
> What I heard a number of times at IETF81 is that what people want is 
> to build innovative new applications with a real time media enabled 
> browser and to it seems clear to me we need a browser API which is 
> flexible and enables applications to do clever things with media 
> streams. So exploring some use cases which require the browser to 
> duplicate and copy media streams is a good idea.
>
> Indeed the browser API must be flexible and it must enable 
> applications.  Strong ++ on that.
>
> Inasmuch as we, in the IETF, are not really in control of API, to 
> ensure its flexibility we can only make the right protocol 
> selection.   To this end, the worst thing would be trying to reinvent 
> the wheel.  People have been working on SIPREC for more than a year 
> now--would it be not wise on our part to take the output of that work?
>
> Overall, even if the browser does not support the full-blown SIP, I 
> think it must support some parts of it (REFER and SUBSCRIBE/NOTIFY, 
> for one thing) in order to enable flexibility.
>
> A few thousand messages ago, I asked how (asynchronous) notifications 
> could be implemented without having SIP in the browser, and I don't 
> think we had a sufficient discussion.
>
> Igor
>

--------------060201090000090906030102
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    Actually, I don't think that the "must not" decision has ever been
    made by the WG. In fact, two implementation reports from the last
    meeting cited SIP implementation...<br>
    <br>
    I actually thought that the need for supporting SIP in the browser
    was well articulated in Quebec...&nbsp; <br>
    <br>
    Igor<br>
    <br>
    <br>
    <br>
    On 8/23/2011 8:57 PM, Ravindran Parthasarathi wrote:
    <blockquote
cite="mid:2E239D6FCD033C4BAF15F386A979BF5106423F@sonusinmail02.sonusnet.com"
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        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">I agree with Igor. &nbsp;It will be good in case the
            discussion
            in the line why SIP MUST NOT be used for browser. I guess
            that the following
            are not the reason for not selecting SIP:<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><span style="">1)<span style="font: 7pt
                &quot;Times New Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span style="font-size:
            11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">&nbsp;Memory usage in browser is not the limiting
            factor for SIP.
            I know of the SIP application which works well in embedded
            device with 16MB
            RAM.<o:p></o:p></span></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><span style="">2)<span style="font: 7pt
                &quot;Times New Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span style="font-size:
            11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">In case the idea is to use the simple alternative
            protocol for
            the basic dialog between browser to browser. It is possible
            for the same
            protocol to become as complex as SIP in 5-10 years while
            adding more feature or
            services. &nbsp;We end-up with two protocol for real-time
            communication from
            IETF and interact through gateways. <o:p></o:p></span></p>
        <p class="MsoListParagraph" style="text-indent: -0.25in;"><!--[if !supportLists]--><span
            style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><span style="">3)<span style="font: 7pt
                &quot;Times New Roman&quot;;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
              </span></span></span><!--[endif]--><span style="font-size:
            11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">As per IETF recommendation,&nbsp; standalone
            application in
            endpoint has to use SIP for real-time communication whereas
            the same application
            running in browser has to use RTCWeb recommended protocol.
            It is tough for each
            application to support both protocols or its related
            infrastructure to make it
            work. Or the intention is not to use SIP anywhere in
            endpoint?<o:p></o:p></span></p>
        <p class="MsoNormal" style="margin-left: 0.25in;"><span
            style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
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        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
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            73, 125);">I&#8217;m interested in hearing the reasons for why SIP
            MUST NOT
            be used in browser.<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);"><o:p>&nbsp;</o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Thanks<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
            &quot;Calibri&quot;,&quot;sans-serif&quot;; color: rgb(31,
            73, 125);">Partha<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-size: 11pt; font-family:
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            73, 125);"><o:p>&nbsp;</o:p></span></p>
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              <p class="MsoNormal"><b><span style="font-size: 10pt;
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                    windowtext;">From:</span></b><span style="font-size:
                  10pt; font-family:
                  &quot;Tahoma&quot;,&quot;sans-serif&quot;; color:
                  windowtext;"> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>
                  [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] <b>On Behalf Of </b>Igor
                  Faynberg<br>
                  <b>Sent:</b> Wednesday, August 24, 2011 2:09 AM<br>
                  <b>To:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                  <b>Subject:</b> Re: [rtcweb] Remote recording -
                  RTC-Web client acting as SIPREC
                  session recording client<o:p></o:p></span></p>
            </div>
          </div>
          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
          <p class="MsoNormal"><br>
            <br>
            On 8/23/2011 1:50 PM, Hutton, Andrew wrote:<br>
            <br>
            <br>
            <br>
            <o:p></o:p></p>
          <div>
            <p class="MsoNormal"><span style="font-size: 11pt;
                font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
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            <br>
            <o:p></o:p></p>
          <div>
            <p class="MsoNormal"><span style="font-size: 11pt;
                font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
                color: rgb(31, 73, 125);">What I heard a number of times
                at IETF81 is that what people
                want is to build innovative new applications with a real
                time media enabled
                browser and to it seems clear to me we need a browser
                API which is flexible and
                enables applications to do clever things with media
                streams. So exploring some
                use cases which require the browser to duplicate and
                copy media streams is a
                good idea. </span><o:p></o:p></p>
          </div>
          <p class="MsoNormal">Indeed the browser API must be flexible
            and it must enable
            applications.&nbsp; Strong ++ on that.&nbsp; <br>
            <br>
            Inasmuch as we, in the IETF, are not really in control of
            API, to ensure its
            flexibility we can only make the right protocol selection.&nbsp;&nbsp;
            To this
            end, the worst thing would be trying to reinvent the wheel.&nbsp;
            People have
            been working on SIPREC for more than a year now--would it be
            not wise on our
            part to take the output of that work?<br>
            <br>
            Overall, even if the browser does not support the full-blown
            SIP, I think it
            must support some parts of it (REFER and SUBSCRIBE/NOTIFY,
            for one thing) in
            order to enable flexibility.<br>
            <br>
            A few thousand messages ago, I asked how (asynchronous)
            notifications could be
            implemented without having SIP in the browser, and I don't
            think we had a
            sufficient discussion.<br>
            <br>
            Igor<br>
            <br>
            <o:p></o:p></p>
          <div>
            <p class="MsoNormal" style=""><span style="font-size: 11pt;
                font-family:
                &quot;Calibri&quot;,&quot;sans-serif&quot;;">&nbsp;</span><o:p></o:p></p>
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          <p class="MsoNormal"><o:p>&nbsp;</o:p></p>
        </div>
      </div>
    </blockquote>
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On 8/23/2011 1:50 PM, Hutton, Andrew wrote:

> Hi,
>   
> Also agree that implementing a full SIP Session Recording Client (SRC) as defined ]
> by the SIPREC WG in to the browser would be a tall order and would open the flood
> gates for a lot of other requests for SIP functionality in the browser.

Agreed

> If however SIP is not in the browser then I think it is a useful and interesting test case
> to look at whether we could build a decomposed SRC with the browser handling the
> media and the web server handling the signalling. What I heard a number of times at
> IETF81 is that what people want is to build innovative new applications with a real time
> media enabled browser and to it seems clear to me we need a browser API which is
> flexible and enables applications to do clever things with media streams. So exploring
> some use cases which require the browser to duplicate and copy media streams is a good idea.
>   
> Therefore I think we should keep explore the remote recording use case further.

My personal (not Mozilla's) opinion is that SIPREC-style recording is 
out-of-scope,
and would be handled in this context by a webrtc-B2BUA (i.e. something 
that sits on the
signalling and media paths), or something that interacts directly with 
the app.

What I do want to consider for in-scope are local recording options.  
This can cover the "local
answering machine" case, but also "I want to record my call with Dad", 
"I want to record
my team's chatter in the game", "I want to interview someone for 
genealogy research", etc, etc.
Yes, these can be done (poorly) by using some external app to capture 
the screen and
speaker/mic audio - I don't consider that a replacement for local recording.

Local recording is far less problematical protocol-wise than SIPREC, and 
doesn't require
mandating SIP.

-- 
Randell Jesup
randell-ietf@jesup.org


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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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Randell,=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> Sent: 24 August 2011 08:43
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Remote recording - RTC-Web client=20
> acting as SIPREC session recording client
>=20
> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>=20
> > Hi,
> >  =20
> > Also agree that implementing a full SIP Session Recording=20
> Client (SRC) as defined ]
> > by the SIPREC WG in to the browser would be a tall order=20
> and would open the flood
> > gates for a lot of other requests for SIP functionality in=20
> the browser.
>=20
> Agreed
>=20
> > If however SIP is not in the browser then I think it is a=20
> useful and interesting test case
> > to look at whether we could build a decomposed SRC with the=20
> browser handling the
> > media and the web server handling the signalling. What I=20
> heard a number of times at
> > IETF81 is that what people want is to build innovative new=20
> applications with a real time
> > media enabled browser and to it seems clear to me we need a=20
> browser API which is
> > flexible and enables applications to do clever things with=20
> media streams. So exploring
> > some use cases which require the browser to duplicate and=20
> copy media streams is a good idea.
> >  =20
> > Therefore I think we should keep explore the remote=20
> recording use case further.
>=20
> My personal (not Mozilla's) opinion is that SIPREC-style recording is=20
> out-of-scope,
> and would be handled in this context by a webrtc-B2BUA (i.e.=20
> something=20
> that sits on the
> signalling and media paths), or something that interacts=20
> directly with=20
> the app.
>=20
> What I do want to consider for in-scope are local recording options. =20
> This can cover the "local
> answering machine" case, but also "I want to record my call=20
> with Dad",=20
> "I want to record
> my team's chatter in the game", "I want to interview someone for=20
> genealogy research", etc, etc.
> Yes, these can be done (poorly) by using some external app to capture=20
> the screen and
> speaker/mic audio - I don't consider that a replacement for=20
> local recording.
>=20
> Local recording is far less problematical protocol-wise than=20
> SIPREC, and=20
> doesn't require
> mandating SIP.
[JRE] I am not advocating supporting SIP in the browser just for the purpos=
e of supporting SIPREC SRC functionality. What I am saying is that if the b=
rowser is concerned only with media aspects, then only the media aspects of=
 SRC functionality need to be considered in the browser (e.g., copying the =
media streams). On the other hand, if, for other reasons, SIP is implemente=
d in the browser, it would require additional enhancements to support SIP a=
spects of SIPREC SRC functionality.

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


>=20
> --=20
> Randell Jesup
> randell-ietf@jesup.org
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From john.elwell@siemens-enterprise.com  Wed Aug 24 01:39:07 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Wed, 24 Aug 2011 10:40:14 +0200
Thread-Topic: [rtcweb] Proposed text for remote recording use case
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Subject: Re: [rtcweb] Proposed text for remote recording use case
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Cullen,

Since I posted the proposed text, I have attempted to clarify things in a d=
ifferent thread:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00707.html
This may help to clarify things. My proposed text might require refinement,=
 depending on the outcome of that other thread.

HTTP post is not what is really needed here - in common with the SIPREC wor=
k, the idea is to send real-time media in real-time (over RTP), for similar=
 reasons that RTC-Web uses peer-to-peer RTP. Some recording devices use rea=
l-time analytics, and need to receive the information without significant d=
elay.

John
=20

> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: 23 August 2011 22:49
> To: Elwell, John
> Cc: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Proposed text for remote recording use case
>=20
>=20
> Just to try and help clarify requirements ... I suspect the=20
> current requirements you wrote could be meant mostly by just=20
> taking a copy of the data to and from the=20
> speaker/mic/camera/screen and just sending them via a HTTP=20
> post to some web service that recorded them. I suspect you=20
> want something different than that so we might need to poke=20
> at these requirements a bit to see what it is. Alternatively,=20
> if data can be copied from one stream to anther stream using=20
> the JS, does that provide everything you need or is it=20
> something more.=20
>=20
> (BTW... I am probably not in favor of supporting these=20
> requirements in the client as it is typically not a great=20
> place to do recording but that's a separate issues. Just=20
> trying to clarify what we talking about)
>=20
> On Aug 22, 2011, at 8:42 AM, Elwell, John wrote:
>=20
> > 4.2.yy Remote Session Recording
> > In this use case, the web application user wishes to record=20
> a real-time communication at a remote recording device, such=20
> that transmitted and received audio, video or other real-time=20
> media are transmitted in real-time to the remote device. The=20
> remote device can perform archiving, retrieval, playback,=20
> etc., but can also perform real-time analytics on the media.=20
> A typical deployment might be in a contact centre. For a=20
> given medium, the two directions of transmission can be=20
> transmitted together (mixed) or separately. The web=20
> application also sends metadata that gives context to the=20
> stored media.
> >=20
> > New requirements:
> > Fyy1: The browser MUST be able to send in real-time to a=20
> remote recording device media that are being transmitted to=20
> and received from remote participants.
> >=20
> > Ayy1: The web application MUST be able to ask the browser=20
> to transmit in real-time to a remote recording device media=20
> that are being transmitted to and received from remote=20
> participants and, in the case of audio at least, ask for the=20
> two directions of transmission to be transmitted to the=20
> remote recording device mixed or separately.
> >=20
> > John
> >=20
> >=20
> > John Elwell
> > Tel: +44 1908 817801 (office and mobile)
> > Email: john.elwell@siemens-enterprise.com
> > http://www.siemens-enterprise.com/uk/
> >=20
> > Siemens Enterprise Communications Limited.
> > Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15 0DJ.
> > Registered No: 5903714, England.
> >=20
> > Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of Siemens AG.
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>=20
> =

From john.elwell@siemens-enterprise.com  Wed Aug 24 01:50:08 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Cullen Jennings <fluffy@cisco.com>
Date: Wed, 24 Aug 2011 10:51:15 +0200
Thread-Topic: [rtcweb] Proposed text for local recording use case
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Subject: Re: [rtcweb] Proposed text for local recording use case
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Cullen,

In principle, copying real-time media to JS would certainly be a very flexi=
ble solution. However, depending exactly what the JS wants to use it for, I=
 suspect there might possibly be performance issues.

Would there also be a requirement for passing media data from JS back down =
to the browser for transmission to the remote browser? I am thinking in par=
ticular of warning tone/announcements when recording is taking place, but o=
f course it could be used for whatever you want.

The concept of passing media data up to JS / down to the browser would also=
 be a solution for remote recording (if we decide we need it). The JS could=
 then pass the collected data back down to another PeerConnection object fo=
r sending to the remote recording device. This in particular would need to =
be considered from a performance point of view - would passing it up to the=
 JS and back down again be potentially slower than passing it between objec=
ts within the browser?

John

John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com]=20
> Sent: 23 August 2011 22:57
> To: Elwell, John
> Cc: rtcweb@ietf.org; public-webrtc@w3.org
> Subject: Re: [rtcweb] Proposed text for local recording use case
>=20
>=20
> On Aug 22, 2011, at 8:42 AM, Elwell, John wrote:
>=20
> > 4.2.xx Local Session Recording
> > In this use case, the web application user wishes to record=20
> a real-time communication locally, such that transmitted and=20
> received audio, video or other real-time media are stored in=20
> one or more files. For a given medium, the two directions of=20
> transmission can be stored in the same file (mixed) or in=20
> separate files. The web application also stores metadata that=20
> gives context to the stored media.
> >=20
> > New requirements:
> > Fxx1: The browser MUST be able to store transmitted and=20
> received media in local files.
> >=20
> > Axx1: The web application MUST be able to ask the browser=20
> to store transmitted and received media in local files, and=20
> in the case of audio at least, ask for the two directions of=20
> transmission to be stored mixed or separately.
> >=20
> > John
>=20
> I think I want something just slightly different and that is=20
> to send media that is data in the JS and to be able to hand=20
> media as data to the JS. If the JS happens to read / write=20
> that to a file that is fine with me but this allows the JS to=20
> operate on the data or synthetically generate media and opens=20
> the doors to the possibility of writing a codec in JS or NACL. =

From andrew.hutton@siemens-enterprise.com  Wed Aug 24 02:31:01 2011
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 24 Aug 2011 11:32:08 +0200
Thread-Topic: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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Randell,

I would like to make it clear that I am not saying that SIPREC is a reason =
for putting SIP in the browser but if SIP is not in the browser then we nee=
d to be able to build applications like a SIPREC SRC using a combination of=
 the browser and the web application.=20

Therefore I do think that the remote recording case is a useful use case to=
 explore and should not be out of scope as it helps us understand the type =
of media handling requirements that the browser needs to support innovative=
 applications. =20

Of course as John stated if a decision is made to put SIP in the browser th=
en there would be additional browser and API requirements to support SIPREC=
.

Regards
Andy



> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Elwell, John
> Sent: 24 August 2011 09:24
> To: Randell Jesup; rtcweb@ietf.org
> Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
> SIPREC session recording client
>=20
> Randell,
>=20
> > -----Original Message-----
> > From: rtcweb-bounces@ietf.org
> > [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> > Sent: 24 August 2011 08:43
> > To: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Remote recording - RTC-Web client
> > acting as SIPREC session recording client
> >
> > On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
> >
> > > Hi,
> > >
> > > Also agree that implementing a full SIP Session Recording
> > Client (SRC) as defined ]
> > > by the SIPREC WG in to the browser would be a tall order
> > and would open the flood
> > > gates for a lot of other requests for SIP functionality in
> > the browser.
> >
> > Agreed
> >
> > > If however SIP is not in the browser then I think it is a
> > useful and interesting test case
> > > to look at whether we could build a decomposed SRC with the
> > browser handling the
> > > media and the web server handling the signalling. What I
> > heard a number of times at
> > > IETF81 is that what people want is to build innovative new
> > applications with a real time
> > > media enabled browser and to it seems clear to me we need a
> > browser API which is
> > > flexible and enables applications to do clever things with
> > media streams. So exploring
> > > some use cases which require the browser to duplicate and
> > copy media streams is a good idea.
> > >
> > > Therefore I think we should keep explore the remote
> > recording use case further.
> >
> > My personal (not Mozilla's) opinion is that SIPREC-style recording is
> > out-of-scope,
> > and would be handled in this context by a webrtc-B2BUA (i.e.
> > something
> > that sits on the
> > signalling and media paths), or something that interacts
> > directly with
> > the app.
> >
> > What I do want to consider for in-scope are local recording options.
> > This can cover the "local
> > answering machine" case, but also "I want to record my call
> > with Dad",
> > "I want to record
> > my team's chatter in the game", "I want to interview someone for
> > genealogy research", etc, etc.
> > Yes, these can be done (poorly) by using some external app to capture
> > the screen and
> > speaker/mic audio - I don't consider that a replacement for
> > local recording.
> >
> > Local recording is far less problematical protocol-wise than
> > SIPREC, and
> > doesn't require
> > mandating SIP.
> [JRE] I am not advocating supporting SIP in the browser just for the
> purpose of supporting SIPREC SRC functionality. What I am saying is
> that if the browser is concerned only with media aspects, then only the
> media aspects of SRC functionality need to be considered in the browser
> (e.g., copying the media streams). On the other hand, if, for other
> reasons, SIP is implemented in the browser, it would require additional
> enhancements to support SIP aspects of SIPREC SRC functionality.
>=20
> John
>=20
> John Elwell
> Tel: +44 1908 817801 (office and mobile)
> Email: john.elwell@siemens-enterprise.com
> http://www.siemens-enterprise.com/uk/
>=20
> Siemens Enterprise Communications Limited.
> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
> 0DJ.
> Registered No: 5903714, England.
>=20
> Siemens Enterprise Communications Limited is a Trademark Licensee of
> Siemens AG.
>=20
>=20
> >
> > --
> > Randell Jesup
> > randell-ietf@jesup.org
> >
> > _______________________________________________
> > 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: <A444A0F8084434499206E78C106220CA0B00FDB08B@MCHP058A.global-ad.net>	<89177AB2-F721-47E4-8471-2180EDA10615@voxeo.com>	<A444A0F8084434499206E78C106220CA0B00FDB34D@MCHP058A.global-ad.net>	<496EE152-41F2-49AB-A136-05735FE5A9F9@voxeo.com>	<101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net> <4E540FE2.7020605@alcatel-lucent.com>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC	session recording client
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On 08/23/11 22:38, Igor Faynberg wrote:
>
>
> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>
>
>> ...
>
>> What I heard a number of times at IETF81 is that what people want is 
>> to build innovative new applications with a real time media enabled 
>> browser and to it seems clear to me we need a browser API which is 
>> flexible and enables applications to do clever things with media 
>> streams. So exploring some use cases which require the browser to 
>> duplicate and copy media streams is a good idea.
> Indeed the browser API must be flexible and it must enable 
> applications.  Strong ++ on that.
>
> Inasmuch as we, in the IETF, are not really in control of API, to 
> ensure its flexibility we can only make the right protocol 
> selection.   To this end, the worst thing would be trying to reinvent 
> the wheel.  People have been working on SIPREC for more than a year 
> now--would it be not wise on our part to take the output of that work?
>
> Overall, even if the browser does not support the full-blown SIP, I 
> think it must support some parts of it (REFER and SUBSCRIBE/NOTIFY, 
> for one thing) in order to enable flexibility.

I would like to focus on the functionality you think we need for this 
flexibility - if we really need it, I would like to capture it in an use 
case; if it's not vitally needed, I would like to leave it for further work.

I don't know what it would mean to support SUBSCRIBE/NOTIFY without 
supporting SIP. I'm not sure I want to have that discussion - it seems 
to me that some applications will require presence and subscription to 
presence, but that little of this has to be built into the browser; the 
Gmail chat function, which indeed exposes presence and subscription, has 
no browser component at all.

>
> A few thousand messages ago, I asked how (asynchronous) notifications 
> could be implemented without having SIP in the browser, and I don't 
> think we had a sufficient discussion.

have you absorbed this?

http://dev.w3.org/2006/webapi/WebNotifications/publish/

"Notification" is not a term that can be used without context - it means 
very many different things to different people. So one reason why you 
did not get sufficient discussion might be that we didn't have a shared 
understanding of what you were asking.



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      http-equiv="Content-Type">
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    On 08/23/11 22:38, Igor Faynberg wrote:
    <blockquote cite="mid:4E540FE2.7020605@alcatel-lucent.com"
      type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      <br>
      <br>
      On 8/23/2011 1:50 PM, Hutton, Andrew wrote:<br>
      <br>
      <br>
      <blockquote
cite="mid:101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net"
        type="cite">
        <div class="WordSection1"><span style="font-size: 11pt;
            font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
            color: rgb(31, 73, 125);"> ...</span></div>
      </blockquote>
      <br>
      <blockquote
cite="mid:101C6067BEC68246B0C3F6843BCCC1E31018BF6BE2@MCHP058A.global-ad.net"
        type="cite">
        <div class="WordSection1"><span style="font-size: 11pt;
            font-family: &quot;Calibri&quot;,&quot;sans-serif&quot;;
            color: rgb(31, 73, 125);">What I heard a number of times at
            IETF81 is that what people want is to build innovative new
            applications with a real time media enabled browser and to
            it seems clear to me we need a browser API which is flexible
            and enables applications to do clever things with media
            streams. So exploring some use cases which require the
            browser to duplicate and copy media streams is a good idea.
            <br>
          </span></div>
      </blockquote>
      Indeed the browser API must be flexible and it must enable
      applications.&nbsp; Strong ++ on that.&nbsp; <br>
      <br>
      Inasmuch as we, in the IETF, are not really in control of API, to
      ensure its flexibility we can only make the right protocol
      selection.&nbsp;&nbsp; To this end, the worst thing would be trying to
      reinvent the wheel.&nbsp; People have been working on SIPREC for more
      than a year now--would it be not wise on our part to take the
      output of that work?<br>
      <br>
      Overall, even if the browser does not support the full-blown SIP,
      I think it must support some parts of it (REFER and
      SUBSCRIBE/NOTIFY, for one thing) in order to enable flexibility.<br>
    </blockquote>
    <br>
    I would like to focus on the functionality you think we need for
    this flexibility - if we really need it, I would like to capture it
    in an use case; if it's not vitally needed, I would like to leave it
    for further work.<br>
    <br>
    I don't know what it would mean to support SUBSCRIBE/NOTIFY without
    supporting SIP. I'm not sure I want to have that discussion - it
    seems to me that some applications will require presence and
    subscription to presence, but that little of this has to be built
    into the browser; the Gmail chat function, which indeed exposes
    presence and subscription, has no browser component at all.<br>
    <br>
    <blockquote cite="mid:4E540FE2.7020605@alcatel-lucent.com"
      type="cite"> <br>
      A few thousand messages ago, I asked how (asynchronous)
      notifications could be implemented without having SIP in the
      browser, and I don't think we had a sufficient discussion.<br>
    </blockquote>
    <br>
    have you absorbed this?<br>
    <br>
    <a href="http://dev.w3.org/2006/webapi/WebNotifications/publish/">http://dev.w3.org/2006/webapi/WebNotifications/publish/</a><br>
    <br>
    "Notification" is not a term that can be used without context - it
    means very many different things to different people. So one reason
    why you did not get sufficient discussion might be that we didn't
    have a shared understanding of what you were asking.<br>
    <br>
    <br>
  </body>
</html>

--------------040301000207010508020202--

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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Overview: Real Time Protocols for Brower-based Applicati=
ons
	Author(s)       : Harald T. Alvestrand
	Filename        : draft-ietf-rtcweb-overview-01.txt
	Pages           : 17
	Date            : 2011-08-24

   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - &quot;real time communication on the Web&quot;.

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a candidate to become a work item of the RTCWEB
   working group.



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

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

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

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A few drawings, some more definitions, and a link to the W3C API spec.

If people are quick with comments, I can get another version done before 
our September 7 meeting; version numbers are cheap.

                Harald

-------- Original Message --------
Subject: 	New Version Notification for draft-ietf-rtcweb-overview-01.txt
Date: 	Wed, 24 Aug 2011 07:37:14 -0700
From: 	internet-drafts@ietf.org
To: 	harald@alvestrand.no
CC: 	harald@alvestrand.no



A new version of I-D, draft-ietf-rtcweb-overview-01.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 01
Title:		 Overview: Real Time Protocols for Brower-based Applications
Creation date:	 2011-08-24
WG ID:		 rtcweb
Number of pages: 17

Abstract:
    This document gives an overview and context of a protocol suite
    intended for use with real-time applications that can be deployed in
    browsers -&quot;real time communication on the Web&quot;.

    It intends to serve as a starting and coordination point to make sure
    all the parts that are needed to achieve this goal are findable, and
    that the parts that belong in the Internet protocol suite are fully
    specified and on the right publication track.

    This work is an attempt to synthesize the input of many people, but
    makes no claims to fully represent the views of any of them.  All
    parts of the document should be regarded as open for discussion,
    unless the RTCWEB chairs have declared consensus on an item.

    This document is a candidate to become a work item of the RTCWEB
    working group.





The IETF Secretariat



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    <meta http-equiv="content-type" content="text/html; charset=UTF-8">
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    A few drawings, some more definitions, and a link to the W3C API
    spec.<br>
    <br>
    If people are quick with comments, I can get another version done
    before our September 7 meeting; version numbers are cheap.<br>
    <br>
    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-ietf-rtcweb-overview-01.txt</td>
        </tr>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date: </th>
          <td>Wed, 24 Aug 2011 07:37:14 -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>
        <tr>
          <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC: </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-ietf-rtcweb-overview-01.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 01
Title:		 Overview: Real Time Protocols for Brower-based Applications
Creation date:	 2011-08-24
WG ID:		 rtcweb
Number of pages: 17

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - &amp;quot;real time communication on the Web&amp;quot;.

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a candidate to become a work item of the RTCWEB
   working group.


                                                                                  


The IETF Secretariat

</pre>
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</html>

--------------070005070304020808030604--

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Date: Wed, 24 Aug 2011 08:15:14 -0700
From: Dzonatas Sol <dzonatas@gmail.com>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 08/24/2011 12:43 AM, Randell Jesup wrote:
> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>
>> Hi,
>>   Also agree that implementing a full SIP Session Recording Client 
>> (SRC) as defined ]
>> by the SIPREC WG in to the browser would be a tall order and would 
>> open the flood
>> gates for a lot of other requests for SIP functionality in the browser.
>
> Agreed
>
>> If however SIP is not in the browser then I think it is a useful and 
>> interesting test case
>> to look at whether we could build a decomposed SRC with the browser 
>> handling the
>> media and the web server handling the signalling. What I heard a 
>> number of times at
>> IETF81 is that what people want is to build innovative new 
>> applications with a real time
>> media enabled browser and to it seems clear to me we need a browser 
>> API which is
>> flexible and enables applications to do clever things with media 
>> streams. So exploring
>> some use cases which require the browser to duplicate and copy media 
>> streams is a good idea.
>>   Therefore I think we should keep explore the remote recording use 
>> case further.
>
> My personal (not Mozilla's) opinion is that SIPREC-style recording is 
> out-of-scope,
> and would be handled in this context by a webrtc-B2BUA (i.e. something 
> that sits on the
> signalling and media paths), or something that interacts directly with 
> the app.
>
> What I do want to consider for in-scope are local recording options.  
> This can cover the "local
> answering machine" case, but also "I want to record my call with Dad", 
> "I want to record
> my team's chatter in the game", "I want to interview someone for 
> genealogy research", etc, etc.
> Yes, these can be done (poorly) by using some external app to capture 
> the screen and
> speaker/mic audio - I don't consider that a replacement for local 
> recording.
>
> Local recording is far less problematical protocol-wise than SIPREC, 
> and doesn't require
> mandating SIP.
>

Real-Time systems still imply static content. The source of that static 
content may be directly from such cases above, from client protocols, or 
from servers. How the capture is of static content done and distributed 
affects overall performance, yet that capture is useless to others in 
public cases if it packs credentials; we are redundancy aware.

Consider the blind user in 3D world (or earthquake survivor that NEEDS 
updated and accessible haptic maps), we shouldn't have to override 
SIP/vox-mixer or continue to call them ghosts in the machine to the 
non-blind. The word "incognito" (used instead) may seem like an 
evolutionary step forward for inclusion of such "accessibility" modes 
where before implementers thought the famous-dazzled words had no use. 
(Echo-location/sonar-simulation... anything but mere "ghosts" that pass 
through virtual walls because they can see them.)

Example: "This audio stream *beep* is in incognito mode while *beep* 
search and rescue is underway *beep*." If there is no emergency (or 
eavesdrop) then the beeps aren't mandatory in incognito usage.

-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From hoang.su.tk@gmail.com  Wed Aug 17 19:09:18 2011
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Subject: [rtcweb] Some misunderstandings about <Usecase & architecture: Browser application with separate webserver & voipserver>
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Hello everyone,

I read about Partha's question at RTCWeb mailing list at
http://www.ietf.org/mail-archive/web/rtcweb/current/msg00563.html

He said that there's interoprating issues due to mapping between
RTCWebserver and SIP client w.r.t mid-call services in the
introduction of new gateway. Could anyone please explain more about
those issues? I have a little bit difficult to understand his
meanings.

I couldn't send an email to Partha directly by partr@cisco.com, so I
decided to send to this email. Sorry for any inconvenience.

Thanks,
TuanND

From jonas@sicking.cc  Tue Aug 23 17:06:04 2011
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Subject: Re: [rtcweb] Relation of Mozilla WebAPI announced today to RTCWEB/WebRTC work?
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On Tue, Aug 23, 2011 at 11:41 AM, Timothy B. Terriberry
<tterriberry@mozilla.com> wrote:
> Harald Alvestrand wrote:
>>
>> On 08/23/11 19:59, Timothy B. Terriberry wrote:
>>>>
>>>> Can anyone from Mozilla please comment on how the "WebAPI" announced
>>>> today with the goal to " provide a basic HTML5 phone experience within 3
>>>> to 6 months" relates to the work happening in RTCWEB/WebRTC?
>>>
>>> This is more along the lines of getting access to the APIs needed to
>>> instruct a mobile phone to initiate a phone call, rather than to host
>>> that call inside the browser. Presumably also with some way to detect
>>> that your on a device that supports that capability at all.
>>
>> The blog post sounds like a virtual clone of the W3C Device API charter.
>>
>> Is that your planned venue for standardizing these APIs?
>
> I'm not sure that's been worked out yet, but before I say too many incorrect
> things, let me CC Jonas so he can correct me.

I'm working on a more detailed blog post which mentions our
standardization strategy. The short story is that we absolutely want
all of these APIs to be standardized. Our plan is to bring the APIs to
W3C once we've experimented enough with them to feel good about them.
Which group this ends up in depends on the API.

>> I can't say I'm overly happy with the amount of imagination spent on the
>> name, but the effort seems to make sense!
>
> Well, I didn't name the thing, but keep in mind that initiating phone calls
> is only one small part of the total effort.

Indeed. I'm not a big fan of the name either. But the goal is indeed
to be much broader than just applications for phone and SMS.

/ Jonas

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On 8/24/2011 8:04 AM, Harald Alvestrand wrote:
> ...
> I would like to focus on the functionality you think we need for this 
> flexibility - if we really need it, I would like to capture it in an 
> use case; if it's not vitally needed, I would like to leave it for 
> further work.

I agree. A written use case is a must and it is the first thing to do. 
The ball is in my yard.
>
> I don't know what it would mean to support SUBSCRIBE/NOTIFY without 
> supporting SIP. 

To me, that would mean supporting only REGISTER, SUBSCRIBE/NOTIFY, and 
REFER.

> I'm not sure I want to have that discussion - it seems to me that some 
> applications will require presence and subscription to presence, but 
> that little of this has to be built into the browser; the Gmail chat 
> function, which indeed exposes presence and subscription, has no 
> browser component at all.
>

There is a huge SIP-based legacy out there.  For the case where the 
browser is the de-facto OS on an Internet appliance, applications will 
need to be re-written to rely on WebRTC API and rely on RTCWeb 
protocol.  The question is whether SIP client can be efficiently 
implemented without the browser taking care of some critical pieces.  (I 
don't know the answer.)  This is what has been the gist of my intervention.




>> ...
>
> have you absorbed this?
>
> http://dev.w3.org/2006/webapi/WebNotifications/publish/
>
> "Notification" is not a term that can be used without context - it 
> means very many different things to different people. So one reason 
> why you did not get sufficient discussion might be that we didn't have 
> a shared understanding of what you were asking.

I will need to read this. Thanks for the pointer!

Igor
>
>

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
  <head>
    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
  </head>
  <body text="#000000" bgcolor="#ffffff">
    <br>
    <br>
    On 8/24/2011 8:04 AM, Harald Alvestrand wrote:
    <blockquote cite="mid:4E54E8B8.8040302@alvestrand.no" type="cite">
      <meta content="text/html; charset=ISO-8859-1"
        http-equiv="Content-Type">
      ...<br>
      I would like to focus on the functionality you think we need for
      this flexibility - if we really need it, I would like to capture
      it in an use case; if it's not vitally needed, I would like to
      leave it for further work.<br>
    </blockquote>
    <br>
    I agree. A written use case is a must and it is the first thing to
    do. The ball is in my yard.<br>
    <blockquote cite="mid:4E54E8B8.8040302@alvestrand.no" type="cite"> <br>
      I don't know what it would mean to support SUBSCRIBE/NOTIFY
      without supporting SIP. </blockquote>
    <br>
    To me, that would mean supporting only REGISTER, SUBSCRIBE/NOTIFY,
    and REFER.&nbsp; <br>
    <br>
    <blockquote cite="mid:4E54E8B8.8040302@alvestrand.no" type="cite">I'm
      not sure I want to have that discussion - it seems to me that some
      applications will require presence and subscription to presence,
      but that little of this has to be built into the browser; the
      Gmail chat function, which indeed exposes presence and
      subscription, has no browser component at all.<br>
      <br>
    </blockquote>
    <br>
    There is a huge SIP-based legacy out there.&nbsp; For the case where the
    browser is the de-facto OS on an Internet appliance, applications
    will need to be re-written to rely on WebRTC API and rely on RTCWeb
    protocol.&nbsp; The question is whether SIP client can be efficiently
    implemented without the browser taking care of some critical
    pieces.&nbsp; (I don't know the answer.)&nbsp; This is what has been the gist
    of my intervention.<br>
    <br>
    <br>
    <br>
    <br>
    <blockquote cite="mid:4E54E8B8.8040302@alvestrand.no" type="cite">
      <blockquote cite="mid:4E540FE2.7020605@alcatel-lucent.com"
        type="cite"> ...<br>
      </blockquote>
      <br>
      have you absorbed this?<br>
      <br>
      <a moz-do-not-send="true"
        href="http://dev.w3.org/2006/webapi/WebNotifications/publish/">http://dev.w3.org/2006/webapi/WebNotifications/publish/</a><br>
      <br>
      "Notification" is not a term that can be used without context - it
      means very many different things to different people. So one
      reason why you did not get sufficient discussion might be that we
      didn't have a shared understanding of what you were asking.<br>
    </blockquote>
    <br>
    I will need to read this. Thanks for the pointer!<br>
    <br>
    Igor<br>
    <blockquote cite="mid:4E54E8B8.8040302@alvestrand.no" type="cite"> <br>
      <br>
    </blockquote>
  </body>
</html>

--------------030804010000030200060805--

From pkyzivat@alum.mit.edu  Wed Aug 24 12:05:46 2011
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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 8/24/11 5:32 AM, Hutton, Andrew wrote:
> Randell,
>
> I would like to make it clear that I am not saying that SIPREC is a reason for putting SIP in the browser but if SIP is not in the browser then we need to be able to build applications like a SIPREC SRC using a combination of the browser and the web application.
>
> Therefore I do think that the remote recording case is a useful use case to explore and should not be out of scope as it helps us understand the type of media handling requirements that the browser needs to support innovative applications.
>
> Of course as John stated if a decision is made to put SIP in the browser then there would be additional browser and API requirements to support SIPREC.

I agree with John and Andrew.

Certainly it is possible to punt and say that if you want recording then 
you need to insert a B2BUA that is in the media path and that serves as 
the SRC. But before deciding that is "the story" for recording with 
RTCWEB, it would be good to understand the implications of that approach.

Since this is RTCWEB, I guess we can assume that there is a web server 
interacting with a browser. The browser will provide one end to some 
media streams. There will be another end to those media streams, which 
might be in another browser talking to the same web server, or the web 
server might have those media streams attached to a sip session. In some 
of the cases of interest, the web server won't itself ever touch the 
actual media.

If the web server wants to record the media, it needs to get an SRC into 
the signaling and media paths. If the session is already going to a sip 
server, then its probably not to hard to integrate SRC functionality 
there or juggle the signaling paths to involve a free standing SRC into 
the call path.

But if the web server is orchestrating media streams between browsers 
talking to it, without any sip, then adding recording will be more 
complex. It may need to pull in a sip server that can be an SRC, reroute 
the media through it, etc. Its not evident to me if this would be 
straightforward or not. I suspect not. It would be good to understand 
what would need to be done.

As John and/or Andrew mentioned, it could be advantageous if there were 
primitives by which the JavaScript in the browser had simple ways to 
fork the media and direct it different ways. That could eliminate the 
need to perturb the normal media path in order to do recording.

	Thanks,
	Paul

> Regards
> Andy
>
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> Behalf Of Elwell, John
>> Sent: 24 August 2011 09:24
>> To: Randell Jesup; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
>> SIPREC session recording client
>>
>> Randell,
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org
>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
>>> Sent: 24 August 2011 08:43
>>> To: rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Remote recording - RTC-Web client
>>> acting as SIPREC session recording client
>>>
>>> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>>>
>>>> Hi,
>>>>
>>>> Also agree that implementing a full SIP Session Recording
>>> Client (SRC) as defined ]
>>>> by the SIPREC WG in to the browser would be a tall order
>>> and would open the flood
>>>> gates for a lot of other requests for SIP functionality in
>>> the browser.
>>>
>>> Agreed
>>>
>>>> If however SIP is not in the browser then I think it is a
>>> useful and interesting test case
>>>> to look at whether we could build a decomposed SRC with the
>>> browser handling the
>>>> media and the web server handling the signalling. What I
>>> heard a number of times at
>>>> IETF81 is that what people want is to build innovative new
>>> applications with a real time
>>>> media enabled browser and to it seems clear to me we need a
>>> browser API which is
>>>> flexible and enables applications to do clever things with
>>> media streams. So exploring
>>>> some use cases which require the browser to duplicate and
>>> copy media streams is a good idea.
>>>>
>>>> Therefore I think we should keep explore the remote
>>> recording use case further.
>>>
>>> My personal (not Mozilla's) opinion is that SIPREC-style recording is
>>> out-of-scope,
>>> and would be handled in this context by a webrtc-B2BUA (i.e.
>>> something
>>> that sits on the
>>> signalling and media paths), or something that interacts
>>> directly with
>>> the app.
>>>
>>> What I do want to consider for in-scope are local recording options.
>>> This can cover the "local
>>> answering machine" case, but also "I want to record my call
>>> with Dad",
>>> "I want to record
>>> my team's chatter in the game", "I want to interview someone for
>>> genealogy research", etc, etc.
>>> Yes, these can be done (poorly) by using some external app to capture
>>> the screen and
>>> speaker/mic audio - I don't consider that a replacement for
>>> local recording.
>>>
>>> Local recording is far less problematical protocol-wise than
>>> SIPREC, and
>>> doesn't require
>>> mandating SIP.
>> [JRE] I am not advocating supporting SIP in the browser just for the
>> purpose of supporting SIPREC SRC functionality. What I am saying is
>> that if the browser is concerned only with media aspects, then only the
>> media aspects of SRC functionality need to be considered in the browser
>> (e.g., copying the media streams). On the other hand, if, for other
>> reasons, SIP is implemented in the browser, it would require additional
>> enhancements to support SIP aspects of SIPREC SRC functionality.
>>
>> John
>>
>> John Elwell
>> Tel: +44 1908 817801 (office and mobile)
>> Email: john.elwell@siemens-enterprise.com
>> http://www.siemens-enterprise.com/uk/
>>
>> Siemens Enterprise Communications Limited.
>> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
>> 0DJ.
>> Registered No: 5903714, England.
>>
>> Siemens Enterprise Communications Limited is a Trademark Licensee of
>> Siemens AG.
>>
>>
>>>
>>> --
>>> Randell Jesup
>>> randell-ietf@jesup.org
>>>
>>> _______________________________________________
>>> 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 harald@alvestrand.no  Thu Aug 25 05:00:25 2011
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 08/24/11 21:06, Paul Kyzivat wrote:
> On 8/24/11 5:32 AM, Hutton, Andrew wrote:
>> Randell,
>>
>> I would like to make it clear that I am not saying that SIPREC is a 
>> reason for putting SIP in the browser but if SIP is not in the 
>> browser then we need to be able to build applications like a SIPREC 
>> SRC using a combination of the browser and the web application.
>>
>> Therefore I do think that the remote recording case is a useful use 
>> case to explore and should not be out of scope as it helps us 
>> understand the type of media handling requirements that the browser 
>> needs to support innovative applications.
>>
>> Of course as John stated if a decision is made to put SIP in the 
>> browser then there would be additional browser and API requirements 
>> to support SIPREC.
>
> I agree with John and Andrew.
>
> Certainly it is possible to punt and say that if you want recording 
> then you need to insert a B2BUA that is in the media path and that 
> serves as the SRC. But before deciding that is "the story" for 
> recording with RTCWEB, it would be good to understand the implications 
> of that approach.
>
> Since this is RTCWEB, I guess we can assume that there is a web server 
> interacting with a browser. The browser will provide one end to some 
> media streams. There will be another end to those media streams, which 
> might be in another browser talking to the same web server, or the web 
> server might have those media streams attached to a sip session. In 
> some of the cases of interest, the web server won't itself ever touch 
> the actual media.
>
> If the web server wants to record the media, it needs to get an SRC 
> into the signaling and media paths. If the session is already going to 
> a sip server, then its probably not to hard to integrate SRC 
> functionality there or juggle the signaling paths to involve a free 
> standing SRC into the call path.
>
> But if the web server is orchestrating media streams between browsers 
> talking to it, without any sip, then adding recording will be more 
> complex. It may need to pull in a sip server that can be an SRC, 
> reroute the media through it, etc. Its not evident to me if this would 
> be straightforward or not. I suspect not. It would be good to 
> understand what would need to be done.
>
> As John and/or Andrew mentioned, it could be advantageous if there 
> were primitives by which the JavaScript in the browser had simple ways 
> to fork the media and direct it different ways. That could eliminate 
> the need to perturb the normal media path in order to do recording.
If we can get the signalling part done through some channel that doesn't 
touch the RTCWEB API, it may be enough to say something like:

The API MUST allow the application to specify that a copy of a media 
stream (incoming or outgoing) is sent out to some other correspondent.

In PeerConnection API terms, that means that some construct like this 
should work:

    mainConn = new PeerConnection(...)
<connect mainConn to remote participant, producing remoteStream and 
localStream>
    recorderConn = new PeerConnection(...)
<connect recorderConn to remote recorder>
    recorderConn.addStream(remoteStream)
    recorderConn.addStream(localStream)

The important part of this, implementation-wise, is that one stream 
needs to be able to have several destinations. What that means for codec 
negotiations, transcoding-or-not and so on may perhaps best be kept 
under the covers.
>
>     Thanks,
>     Paul
>
>> Regards
>> Andy
>>
>>
>>
>>> -----Original Message-----
>>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>>> Behalf Of Elwell, John
>>> Sent: 24 August 2011 09:24
>>> To: Randell Jesup; rtcweb@ietf.org
>>> Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
>>> SIPREC session recording client
>>>
>>> Randell,
>>>
>>>> -----Original Message-----
>>>> From: rtcweb-bounces@ietf.org
>>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
>>>> Sent: 24 August 2011 08:43
>>>> To: rtcweb@ietf.org
>>>> Subject: Re: [rtcweb] Remote recording - RTC-Web client
>>>> acting as SIPREC session recording client
>>>>
>>>> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Also agree that implementing a full SIP Session Recording
>>>> Client (SRC) as defined ]
>>>>> by the SIPREC WG in to the browser would be a tall order
>>>> and would open the flood
>>>>> gates for a lot of other requests for SIP functionality in
>>>> the browser.
>>>>
>>>> Agreed
>>>>
>>>>> If however SIP is not in the browser then I think it is a
>>>> useful and interesting test case
>>>>> to look at whether we could build a decomposed SRC with the
>>>> browser handling the
>>>>> media and the web server handling the signalling. What I
>>>> heard a number of times at
>>>>> IETF81 is that what people want is to build innovative new
>>>> applications with a real time
>>>>> media enabled browser and to it seems clear to me we need a
>>>> browser API which is
>>>>> flexible and enables applications to do clever things with
>>>> media streams. So exploring
>>>>> some use cases which require the browser to duplicate and
>>>> copy media streams is a good idea.
>>>>>
>>>>> Therefore I think we should keep explore the remote
>>>> recording use case further.
>>>>
>>>> My personal (not Mozilla's) opinion is that SIPREC-style recording is
>>>> out-of-scope,
>>>> and would be handled in this context by a webrtc-B2BUA (i.e.
>>>> something
>>>> that sits on the
>>>> signalling and media paths), or something that interacts
>>>> directly with
>>>> the app.
>>>>
>>>> What I do want to consider for in-scope are local recording options.
>>>> This can cover the "local
>>>> answering machine" case, but also "I want to record my call
>>>> with Dad",
>>>> "I want to record
>>>> my team's chatter in the game", "I want to interview someone for
>>>> genealogy research", etc, etc.
>>>> Yes, these can be done (poorly) by using some external app to capture
>>>> the screen and
>>>> speaker/mic audio - I don't consider that a replacement for
>>>> local recording.
>>>>
>>>> Local recording is far less problematical protocol-wise than
>>>> SIPREC, and
>>>> doesn't require
>>>> mandating SIP.
>>> [JRE] I am not advocating supporting SIP in the browser just for the
>>> purpose of supporting SIPREC SRC functionality. What I am saying is
>>> that if the browser is concerned only with media aspects, then only the
>>> media aspects of SRC functionality need to be considered in the browser
>>> (e.g., copying the media streams). On the other hand, if, for other
>>> reasons, SIP is implemented in the browser, it would require additional
>>> enhancements to support SIP aspects of SIPREC SRC functionality.
>>>
>>> John
>>>
>>> John Elwell
>>> Tel: +44 1908 817801 (office and mobile)
>>> Email: john.elwell@siemens-enterprise.com
>>> http://www.siemens-enterprise.com/uk/
>>>
>>> Siemens Enterprise Communications Limited.
>>> Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15
>>> 0DJ.
>>> Registered No: 5903714, England.
>>>
>>> Siemens Enterprise Communications Limited is a Trademark Licensee of
>>> Siemens AG.
>>>
>>>
>>>>
>>>> -- 
>>>> Randell Jesup
>>>> randell-ietf@jesup.org
>>>>
>>>> _______________________________________________
>>>> 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 john.elwell@siemens-enterprise.com  Thu Aug 25 05:50:13 2011
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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=20

> -----Original Message-----
> From: rtcweb-bounces@ietf.org=20
> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
> Sent: 25 August 2011 13:02
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Remote recording - RTC-Web client=20
> acting as SIPREC session recording client
>=20
> On 08/24/11 21:06, Paul Kyzivat wrote:
> > On 8/24/11 5:32 AM, Hutton, Andrew wrote:
> >> Randell,
> >>
> >> I would like to make it clear that I am not saying that=20
> SIPREC is a=20
> >> reason for putting SIP in the browser but if SIP is not in the=20
> >> browser then we need to be able to build applications like=20
> a SIPREC=20
> >> SRC using a combination of the browser and the web application.
> >>
> >> Therefore I do think that the remote recording case is a=20
> useful use=20
> >> case to explore and should not be out of scope as it helps us=20
> >> understand the type of media handling requirements that=20
> the browser=20
> >> needs to support innovative applications.
> >>
> >> Of course as John stated if a decision is made to put SIP in the=20
> >> browser then there would be additional browser and API=20
> requirements=20
> >> to support SIPREC.
> >
> > I agree with John and Andrew.
> >
> > Certainly it is possible to punt and say that if you want recording=20
> > then you need to insert a B2BUA that is in the media path and that=20
> > serves as the SRC. But before deciding that is "the story" for=20
> > recording with RTCWEB, it would be good to understand the=20
> implications=20
> > of that approach.
> >
> > Since this is RTCWEB, I guess we can assume that there is a=20
> web server=20
> > interacting with a browser. The browser will provide one=20
> end to some=20
> > media streams. There will be another end to those media=20
> streams, which=20
> > might be in another browser talking to the same web server,=20
> or the web=20
> > server might have those media streams attached to a sip session. In=20
> > some of the cases of interest, the web server won't itself=20
> ever touch=20
> > the actual media.
> >
> > If the web server wants to record the media, it needs to get an SRC=20
> > into the signaling and media paths. If the session is=20
> already going to=20
> > a sip server, then its probably not to hard to integrate SRC=20
> > functionality there or juggle the signaling paths to involve a free=20
> > standing SRC into the call path.
> >
> > But if the web server is orchestrating media streams=20
> between browsers=20
> > talking to it, without any sip, then adding recording will be more=20
> > complex. It may need to pull in a sip server that can be an SRC,=20
> > reroute the media through it, etc. Its not evident to me if=20
> this would=20
> > be straightforward or not. I suspect not. It would be good to=20
> > understand what would need to be done.
> >
> > As John and/or Andrew mentioned, it could be advantageous if there=20
> > were primitives by which the JavaScript in the browser had=20
> simple ways=20
> > to fork the media and direct it different ways. That could=20
> eliminate=20
> > the need to perturb the normal media path in order to do recording.
> If we can get the signalling part done through some channel=20
> that doesn't=20
> touch the RTCWEB API, it may be enough to say something like:
>=20
> The API MUST allow the application to specify that a copy of a media=20
> stream (incoming or outgoing) is sent out to some other correspondent.
[JRE] This is the essential part of what I was trying to get across when I =
originally proposed text for the use cases document. I reproduce it below, =
with a slight change to refer to a third party rather than a recording devi=
ce:
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
4.2.yy Remote Session Recording
In this use case, the web application user wishes to record a real-time com=
munication at a remote third party (e.g., a recording device), such that tr=
ansmitted and received audio, video or other real-time media are transmitte=
d in real-time to the third party. The third party can perform recording, a=
rchiving, retrieval, playback, etc., but can also perform real-time analyti=
cs on the media. A typical deployment might be in a contact centre. For a g=
iven medium, the two directions of transmission can be transmitted together=
 (mixed) or separately. The web application also sends metadata that gives =
context to the stored media.

New requirements:
Fyy1: The browser MUST be able to send in real-time to a third party media =
that are being transmitted to and received from remote participants.

Ayy1: The web application MUST be able to ask the browser to transmit in re=
al-time to a third party media that are being transmitted to and received f=
rom remote participants and, in the case of audio at least, ask for the two=
 directions of transmission to be transmitted to the remote recording devic=
e mixed or separately.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

I think it desirable that the application have control over whether to send=
 mixed or unmixed, but we can certainly discuss that. It also has implicati=
ons for the RTP model, because mixing implies an RTP mixer or endpoint mode=
l, rather than just relaying RTP packets. Other aspects such as whether a d=
ifferent security context is needed will also have impact.

>=20
> In PeerConnection API terms, that means that some construct like this=20
> should work:
>=20
>     mainConn =3D new PeerConnection(...)
> <connect mainConn to remote participant, producing remoteStream and=20
> localStream>
>     recorderConn =3D new PeerConnection(...)
> <connect recorderConn to remote recorder>
>     recorderConn.addStream(remoteStream)
>     recorderConn.addStream(localStream)
>=20
> The important part of this, implementation-wise, is that one stream=20
> needs to be able to have several destinations. What that=20
> means for codec=20
> negotiations, transcoding-or-not and so on may perhaps best be kept=20
> under the covers.
[JRE] By "under covers", I assume you mean not exposed at the API. That mig=
ht be sufficient, but the implications need thinking about.

John

> >
> >     Thanks,
> >     Paul
> >
> >> Regards
> >> Andy
> >>
> >>
> >>
> >>> -----Original Message-----
> >>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> >>> Behalf Of Elwell, John
> >>> Sent: 24 August 2011 09:24
> >>> To: Randell Jesup; rtcweb@ietf.org
> >>> Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
> >>> SIPREC session recording client
> >>>
> >>> Randell,
> >>>
> >>>> -----Original Message-----
> >>>> From: rtcweb-bounces@ietf.org
> >>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
> >>>> Sent: 24 August 2011 08:43
> >>>> To: rtcweb@ietf.org
> >>>> Subject: Re: [rtcweb] Remote recording - RTC-Web client
> >>>> acting as SIPREC session recording client
> >>>>
> >>>> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
> >>>>
> >>>>> Hi,
> >>>>>
> >>>>> Also agree that implementing a full SIP Session Recording
> >>>> Client (SRC) as defined ]
> >>>>> by the SIPREC WG in to the browser would be a tall order
> >>>> and would open the flood
> >>>>> gates for a lot of other requests for SIP functionality in
> >>>> the browser.
> >>>>
> >>>> Agreed
> >>>>
> >>>>> If however SIP is not in the browser then I think it is a
> >>>> useful and interesting test case
> >>>>> to look at whether we could build a decomposed SRC with the
> >>>> browser handling the
> >>>>> media and the web server handling the signalling. What I
> >>>> heard a number of times at
> >>>>> IETF81 is that what people want is to build innovative new
> >>>> applications with a real time
> >>>>> media enabled browser and to it seems clear to me we need a
> >>>> browser API which is
> >>>>> flexible and enables applications to do clever things with
> >>>> media streams. So exploring
> >>>>> some use cases which require the browser to duplicate and
> >>>> copy media streams is a good idea.
> >>>>>
> >>>>> Therefore I think we should keep explore the remote
> >>>> recording use case further.
> >>>>
> >>>> My personal (not Mozilla's) opinion is that SIPREC-style=20
> recording is
> >>>> out-of-scope,
> >>>> and would be handled in this context by a webrtc-B2BUA (i.e.
> >>>> something
> >>>> that sits on the
> >>>> signalling and media paths), or something that interacts
> >>>> directly with
> >>>> the app.
> >>>>
> >>>> What I do want to consider for in-scope are local=20
> recording options.
> >>>> This can cover the "local
> >>>> answering machine" case, but also "I want to record my call
> >>>> with Dad",
> >>>> "I want to record
> >>>> my team's chatter in the game", "I want to interview someone for
> >>>> genealogy research", etc, etc.
> >>>> Yes, these can be done (poorly) by using some external=20
> app to capture
> >>>> the screen and
> >>>> speaker/mic audio - I don't consider that a replacement for
> >>>> local recording.
> >>>>
> >>>> Local recording is far less problematical protocol-wise than
> >>>> SIPREC, and
> >>>> doesn't require
> >>>> mandating SIP.
> >>> [JRE] I am not advocating supporting SIP in the browser=20
> just for the
> >>> purpose of supporting SIPREC SRC functionality. What I am=20
> saying is
> >>> that if the browser is concerned only with media aspects,=20
> then only the
> >>> media aspects of SRC functionality need to be considered=20
> in the browser
> >>> (e.g., copying the media streams). On the other hand, if,=20
> for other
> >>> reasons, SIP is implemented in the browser, it would=20
> require additional
> >>> enhancements to support SIP aspects of SIPREC SRC functionality.
> >>>
> >>> John
> >>>
> >>> John Elwell
> >>> Tel: +44 1908 817801 (office and mobile)
> >>> Email: john.elwell@siemens-enterprise.com
> >>> http://www.siemens-enterprise.com/uk/
> >>>
> >>> Siemens Enterprise Communications Limited.
> >>> Registered office: Brickhill Street, Willen Lake, Milton=20
> Keynes, MK15
> >>> 0DJ.
> >>> Registered No: 5903714, England.
> >>>
> >>> Siemens Enterprise Communications Limited is a Trademark=20
> Licensee of
> >>> Siemens AG.
> >>>
> >>>
> >>>>
> >>>> --=20
> >>>> Randell Jesup
> >>>> randell-ietf@jesup.org
> >>>>
> >>>> _______________________________________________
> >>>> 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
> >
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> =

From randell-ietf@jesup.org  Thu Aug 25 07:41:06 2011
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Subject: Re: [rtcweb] Mapping of media streams to RTP (Re: Query: What does "context" mean in the context (sic) of requirement A15? [ACTION-6])
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On 8/25/2011 3:50 AM, Elwell, John wrote:
> Randell,
>
> Good point - if the first RTCP packet gets dropped, it will be a long time before the next one arrives, and in the meantime no CSRC to SSRC mapping will be available.

Well, we are looking at AVPF which allows (though does not mandate) faster RTCP,
and I think we're suggesting the "fast sync" rfc as well, so that might cover it
- but there still can be a delay before the RTCP gets through successfully.


>> -----Original Message-----
>> From: public-webrtc-request@w3.org
>> [mailto:public-webrtc-request@w3.org] On Behalf Of Randell Jesup
>> Sent: 24 August 2011 21:13
>> To: public-webrtc@w3.org
>> Subject: Re: Mapping of media streams to RTP (Re: Query: What
>> does "context" mean in the context (sic) of requirement A15?
>> [ACTION-6])
>>
>> On 8/24/2011 11:47 AM, Elwell, John wrote:
>>>> RTCP doesn't have a delay if a SR is the first thing sent on
>>>> a session,
>>>> as recommended in draft-perkins-avt-rapid-rtp-sync-03
>> section 2.1.1 -
>>>> this is permitted by RFC 3550 too, it notes.
>>> [JRE] OK, so CNAME can be used to map an RTP stream to a
>> given media description in SDP. CNAME alone does not indicate
>> the purpose of an RTP stream - it would still need something
>> else to map it to "game" or "agent", and that is where
>> a=content comes in.
>>
>> Don't count on RTCP getting received...  Start of a session
>> might even
>> be the worst place for it.
>>
-- 
Randell Jesup
randell-ietf@jesup.org


From henry.sinnreich@gmail.com  Thu Aug 25 08:49:43 2011
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> This message is in MIME format. Since your mail reader does not understand
this format, some or all of this message may not be legible.

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>If people are quick with comments, I can get another version done before o=
ur
September 7 meeting;

Some wording about the standards for the signaling between the two Web
servers in Fig.2 would help.
Even if only stating this is out of scope for the rtcweb WG.
(Or does this WG really cover only the single Web server case?)
Leaving many folks scratching their heads...

Thanks, Henry


On 8/24/11 10:08 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:

>    A few drawings, some more definitions, and a link to the W3C API spec.
> =20
>  If people are quick with comments, I can get another version done before=
 our
> September 7 meeting; version numbers are cheap.
> =20
>  =A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Harald
> =20
>  -------- Original Message --------
>  Subject:  New Version Notification for draft-ietf-rtcweb-overview-01.txt
>  Date:  Wed, 24 Aug 2011 07:37:14 -0700
>  From:  internet-drafts@ietf.org
>  To:  harald@alvestrand.no
>  CC:  harald@alvestrand.no
> =20
> =20
> A new version of I-D, draft-ietf-rtcweb-overview-01.txt has been successf=
ully
> submitted by Harald Alvestrand and posted to the IETF repository.
>=20
> Filename:  draft-ietf-rtcweb-overview
> Revision:  01
> Title:   Overview: Real Time Protocols for Brower-based Applications
> Creation date:  2011-08-24
> WG ID:   rtcweb
> Number of pages: 17
>=20
> Abstract:
>    This document gives an overview and context of a protocol suite
>    intended for use with real-time applications that can be deployed in
>    browsers - &quot;real time communication on the Web&quot;.
>=20
>    It intends to serve as a starting and coordination point to make sure
>    all the parts that are needed to achieve this goal are findable, and
>    that the parts that belong in the Internet protocol suite are fully
>    specified and on the right publication track.
>=20
>    This work is an attempt to synthesize the input of many people, but
>    makes no claims to fully represent the views of any of them.  All
>    parts of the document should be regarded as open for discussion,
>    unless the RTCWEB chairs have declared consensus on an item.
>=20
>    This document is a candidate to become a work item of the RTCWEB
>    working group.
>=20
>=20
>                 =20
>=20
>=20
> The IETF Secretariat
>=20
> =20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--B_3397114253_2066583
Content-type: text/html;
	charset="ISO-8859-1"
Content-transfer-encoding: quoted-printable

<HTML>
<HEAD>
<TITLE>Re: [rtcweb] Fwd: New Version Notification for draft-ietf-rtcweb-ove=
rview-01.txt</TITLE>
</HEAD>
<BODY>
<FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><SPAN STYLE=3D'font-size:13pt=
'>&gt;If people are quick with comments, I can get another version done befo=
re our September 7 meeting;<BR>
<BR>
Some wording about the standards for the signaling between the two Web serv=
ers in Fig.2 would help.<BR>
Even if only stating this is out of scope for the rtcweb WG.<BR>
(Or does this WG really cover only the single Web server case?)<BR>
Leaving many folks scratching their heads...<BR>
<BR>
Thanks, Henry<BR>
<BR>
<BR>
On 8/24/11 10:08 AM, &quot;Harald Alvestrand&quot; &lt;<a href=3D"harald@alve=
strand.no">harald@alvestrand.no</a>&gt; wrote:<BR>
<BR>
</SPAN></FONT><BLOCKQUOTE><FONT FACE=3D"Calibri, Verdana, Helvetica, Arial"><=
SPAN STYLE=3D'font-size:13pt'> &nbsp;&nbsp;A few drawings, some more definitio=
ns, and a link to the W3C API spec.<BR>
&nbsp;<BR>
&nbsp;If people are quick with comments, I can get another version done bef=
ore our September 7 meeting; version numbers are cheap.<BR>
&nbsp;<BR>
&nbsp;=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 Harald<BR>
&nbsp;<BR>
&nbsp;-------- Original Message -------- &nbsp;&nbsp;<BR>
&nbsp;Subject: &nbsp;New Version Notification for draft-ietf-rtcweb-overvie=
w-01.txt &nbsp;<BR>
&nbsp;Date: &nbsp;Wed, 24 Aug 2011 07:37:14 -0700 &nbsp;<BR>
&nbsp;From: &nbsp;<a href=3D"internet-drafts@ietf.org">internet-drafts@ietf.o=
rg</a> &nbsp;<BR>
&nbsp;To: &nbsp;<a href=3D"harald@alvestrand.no">harald@alvestrand.no</a> &nb=
sp;<BR>
&nbsp;CC: &nbsp;<a href=3D"harald@alvestrand.no">harald@alvestrand.no</a> &nb=
sp;&nbsp;&nbsp;<BR>
&nbsp;<BR>
&nbsp;<BR>
A new version of I-D, draft-ietf-rtcweb-overview-01.txt has been successful=
ly submitted by Harald Alvestrand and posted to the IETF repository.<BR>
<BR>
Filename:&nbsp; draft-ietf-rtcweb-overview<BR>
Revision:&nbsp; 01<BR>
Title:&nbsp;&nbsp; Overview: Real Time Protocols for Brower-based Applicati=
ons<BR>
Creation date:&nbsp; 2011-08-24<BR>
WG ID:&nbsp;&nbsp; rtcweb<BR>
Number of pages: 17<BR>
<BR>
Abstract:<BR>
&nbsp;&nbsp;&nbsp;This document gives an overview and context of a protocol=
 suite<BR>
&nbsp;&nbsp;&nbsp;intended for use with real-time applications that can be =
deployed in<BR>
&nbsp;&nbsp;&nbsp;browsers - &amp;quot;real time communication on the Web&a=
mp;quot;.<BR>
<BR>
&nbsp;&nbsp;&nbsp;It intends to serve as a starting and coordination point =
to make sure<BR>
&nbsp;&nbsp;&nbsp;all the parts that are needed to achieve this goal are fi=
ndable, and<BR>
&nbsp;&nbsp;&nbsp;that the parts that belong in the Internet protocol suite=
 are fully<BR>
&nbsp;&nbsp;&nbsp;specified and on the right publication track.<BR>
<BR>
&nbsp;&nbsp;&nbsp;This work is an attempt to synthesize the input of many p=
eople, but<BR>
&nbsp;&nbsp;&nbsp;makes no claims to fully represent the views of any of th=
em. &nbsp;All<BR>
&nbsp;&nbsp;&nbsp;parts of the document should be regarded as open for disc=
ussion,<BR>
&nbsp;&nbsp;&nbsp;unless the RTCWEB chairs have declared consensus on an it=
em.<BR>
<BR>
&nbsp;&nbsp;&nbsp;This document is a candidate to become a work item of the=
 RTCWEB<BR>
&nbsp;&nbsp;&nbsp;working group.<BR>
<BR>
<BR>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>
<BR>
<BR>
The IETF Secretariat<BR>
<BR>
&nbsp;<BR>
<HR ALIGN=3DCENTER SIZE=3D"3" WIDTH=3D"95%"></SPAN></FONT><SPAN STYLE=3D'font-size:=
13pt'><FONT FACE=3D"Consolas, Courier New, Courier">__________________________=
_____________________<BR>
rtcweb mailing list<BR>
<a href=3D"rtcweb@ietf.org">rtcweb@ietf.org</a><BR>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a><BR>
</FONT></SPAN></BLOCKQUOTE>
</BODY>
</HTML>


--B_3397114253_2066583--



From randell-ietf@jesup.org  Thu Aug 25 09:25:40 2011
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On 8/25/2011 11:55 AM, Matthew Kaufman wrote:

> On 8/24/2011 4:36 AM, Harald Alvestrand wrote:

Re: negotiate offer/answer separately in each direction


>>
>> I think that:
>> a) this doesn't make sense - it's a completely new SDP/RTP practice,
>> and we should not depart from established practice without a good
>> reason; it also flies against the "keep the number of RTP sessions as
>> low as we can" conclusion that came out of all the discussions about ICE.
>> b) it's not consistent with section 4.1.2 step 7.
>>
>> I think step 16 of section 4:
>> "If connection's ICE started flag is still false, start the
>> PeerConnection ICE Agent and send the initial offer. The initial offer
>> must include a media description for the PeerConnection data UDP media
>> stream, marked as "sendrecv", and for all the streams in localStreams
>> (marked as "sendonly")."
>>
>> is neither correct nor complete.
>
> I agree that "this doesn't make sense" and it is just yet another reason
> that I think SDP offer-answer is entirely inappropriate for WEBRTC.
>
> The web user agent should do what the web site's HTML and cooperating
> Javascript tell it to do. It should not be engaged in direct negotiation
> with the far end such that the outcome is either indeterminate or even
> unexpected, except where direct negotiation is explicitly required to
> meet a security requirement (the initial ICE handshake to determine that
> it is permitted to send data to that endpoint).
>
> Note that any perceived gains by doing this negotiation (like "what if
> my browser is on a slow connection and only wants to receive audio") are
> immediately negated the moment the site changes the SDP enroute to add
> "wants HD-resolution video" for you.

Ok, so that's a bad web-app - don't use it.


Are you really suggesting "send video to someone who doesn't want it anyways"?


"perceived gains" - sending me video at (say) 500K or more plus audio at<50K,
when I'm on a 128K link will kill my connection until I can somehow get the
other side to back off.  Horrid user experience.  Remember people without
broadband or with limited broadband will be using this.  What if I have 768
down, 128 up - but Johnny in the his bedroom is watching youtube videos or
downloading torrents, etc?


> In addition, because the spec is currently written with offer-answer, a
> wide array of use cases that would be possible if capabilities were
> instead exposed via Javascript become impossible. As an example, it
> should be possible for me as a web site developer to create a page that
> can determine, without prompting you to use your camera, whether or not
> a camera is available and if so what codecs it supports. That way I can
> put "I see you have a high-resolution camera and can encode H.264
> video... click here to call a live agent who can help you find the exact
> replacement part" for users who have that, and not if they don't have a
> codec that works with my call center or a camera with sufficient
> resolution to examine the parts I sell.

In what way is that use-case blocked by offer-answer?  Access to the video
needs user confirmation; access to capabilities info shouldn't.  And so the
page can generate an offer with audio and video, or just audio if they have
no camera.  A user declining to send video doesn't necessarily mean you
don't negotiate it - they advertise receive-only for video (if the web app
wants to).


You state above there are major possibilities blocked by offer-answer (I can
think of some minor issues that with O-A require a second O-A pass) - can you
detail some use-cases so we can see the benefit and not just the assertion?
Then we could balance that against the advantage of O-A being well-speced and
known and implemented.  (And the issues surrounding how well and how
interoperably web-app developers can implement their own capability
negotiations/etc)


-- 
Randell Jesup
randell-ietf@jesup.org


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, public-webrtc@w3.org
Subject: Re: [rtcweb] Additional requirement - audio-only communication
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On 8/25/2011 9:24 AM, Randell Jesup wrote:
> On 8/25/2011 11:55 AM, Matthew Kaufman wrote:
>
>> On 8/24/2011 4:36 AM, Harald Alvestrand wrote:
>
> Re: negotiate offer/answer separately in each direction
>
>
>>>
>>> I think that:
>>> a) this doesn't make sense - it's a completely new SDP/RTP practice,
>>> and we should not depart from established practice without a good
>>> reason; it also flies against the "keep the number of RTP sessions as
>>> low as we can" conclusion that came out of all the discussions about 
>>> ICE.
>>> b) it's not consistent with section 4.1.2 step 7.
>>>
>>> I think step 16 of section 4:
>>> "If connection's ICE started flag is still false, start the
>>> PeerConnection ICE Agent and send the initial offer. The initial offer
>>> must include a media description for the PeerConnection data UDP media
>>> stream, marked as "sendrecv", and for all the streams in localStreams
>>> (marked as "sendonly")."
>>>
>>> is neither correct nor complete.
>>
>> I agree that "this doesn't make sense" and it is just yet another reason
>> that I think SDP offer-answer is entirely inappropriate for WEBRTC.
>>
>> The web user agent should do what the web site's HTML and cooperating
>> Javascript tell it to do. It should not be engaged in direct negotiation
>> with the far end such that the outcome is either indeterminate or even
>> unexpected, except where direct negotiation is explicitly required to
>> meet a security requirement (the initial ICE handshake to determine that
>> it is permitted to send data to that endpoint).
>>
>> Note that any perceived gains by doing this negotiation (like "what if
>> my browser is on a slow connection and only wants to receive audio") are
>> immediately negated the moment the site changes the SDP enroute to add
>> "wants HD-resolution video" for you.
>
> Ok, so that's a bad web-app - don't use it.
>
>
> Are you really suggesting "send video to someone who doesn't want it 
> anyways"?

No, I don't think that's a good idea. But one of the arguments I've 
heard *for* doing O-A between the two ends is that it somehow ensures 
that the sender won't send things that the receiver isn't prepared to 
receive.

I was simply pointing out that this isn't true at all. Because the O-A 
can easily be modified in-transit, either end can be trivially convinced 
to do things it otherwise shouldn't be doing.

>
>
> "perceived gains" - sending me video at (say) 500K or more plus audio 
> at<50K,
> when I'm on a 128K link will kill my connection until I can somehow 
> get the
> other side to back off.  Horrid user experience.

Yes. And it will be possible for a "bad web site" to do this whether we 
use O-A or something else.

> Remember people without
> broadband or with limited broadband will be using this.  What if I 
> have 768
> down, 128 up - but Johnny in the his bedroom is watching youtube 
> videos or
> downloading torrents, etc?

In theory the sender will back off once they get reports of massive 
loss, assuming we do congestion control for the media. If not, they 
you're both out of luck. No different really than anything else 
though... I can easily build a web site that lets you connect over HTTP 
and then runs a modified TCP that doesn't slow down when there's loss.


>
>
>> In addition, because the spec is currently written with offer-answer, a
>> wide array of use cases that would be possible if capabilities were
>> instead exposed via Javascript become impossible. As an example, it
>> should be possible for me as a web site developer to create a page that
>> can determine, without prompting you to use your camera, whether or not
>> a camera is available and if so what codecs it supports. That way I can
>> put "I see you have a high-resolution camera and can encode H.264
>> video... click here to call a live agent who can help you find the exact
>> replacement part" for users who have that, and not if they don't have a
>> codec that works with my call center or a camera with sufficient
>> resolution to examine the parts I sell.
>
> In what way is that use-case blocked by offer-answer?  Access to the 
> video
> needs user confirmation; access to capabilities info shouldn't.

I'm reading http://dev.w3.org/2011/webrtc/editor/webrtc.html

I cannot see how to get it to generate an SDP offer before I try to open 
a media connection.

I also cannot see how one could possibly know *what* SDP to offer until 
you call "getUserMedia", which prompts the user. (As an example, I have 
two cameras attached to this computer over USB. One of them has an 
on-board H.264 encoder. The other does not. If my browser can do 
pass-through of the encoded H.264 but doesn't have its own H.264 
encoder, what offer do I generate before the camera is selected? SDP 
doesn't have a good way to encode "maybe".)

What we need are a set of Javascript APIs that let us enumerate the 
available audio input devices and encoders, video input devices and 
encoders, audio output devices and decoders, video output devices and 
decoders. I hate to say it but even H.245 is probably a better model for 
how to collect the capabilities than SDP O-A.

> And so the
> page can generate an offer with audio and video, or just audio if they 
> have
> no camera.  A user declining to send video doesn't necessarily mean you
> don't negotiate it - they advertise receive-only for video (if the web 
> app
> wants to).
>

I don't understand what you mean by "(if the web app wants to)"... the 
current proposed specification doesn't have a way for the web app to 
control whether the SDP offer is receive-only for video or not.

>
> You state above there are major possibilities blocked by offer-answer 
> (I can
> think of some minor issues that with O-A require a second O-A pass) - 
> can you
> detail some use-cases so we can see the benefit and not just the 
> assertion?

It is certainly possible that a repeated set of fake offer-answer 
exchanges can be used by either Javascript or the server to determine 
what the capabilities are, and that a set of rich Javascript-exposed 
capabilities may be used to generate SDP offers and answers, so in that 
sense they are mappable from one to another.

But I would argue that turning capabilities into SDP is easier than the 
reverse. This is really a question of whether we are trying to turn the 
browser into a *platform for building applications* (in which case we 
should be exposing, as an operating system does, APIs for determining 
what is possible and APIs for control) or turning the browser into *a 
phone*, in which case sure, SIP and SDP are both fine choices.

I think there's a whole lot of potential applications beyond what we're 
currently thinking of if we provide a platform and not just a phone.

I've already outlined one case (offering the user the ability to place a 
video call *if* there is an acceptable camera and encoder on the system).

Another obvious one is where there's a dozen people already in a video 
call and one more wishes to join... if the server knows what 
capabilities exist, the server can tell the new joiner "your browser 
can't support video in a format that the other users need" or can tell 
the existing user browsers to switch to a different codec that is now 
compatible with everyone or whatever, rather than having to re-run 
offer-answer with every party.


> Then we could balance that against the advantage of O-A being 
> well-speced and
> known and implemented.

But it *isn't* well-speced for use cases other than one-to-one calling, 
really. It works poorly for recording, it works poorly for large 
multi-party conferences, etc.

And on top of that, it is missing important attributes that we'll want 
to control (like whether the Opus codec is being forced to "music" mode 
or not).

> (And the issues surrounding how well and how
> interoperably web-app developers can implement their own capability
> negotiations/etc)
>
>

For interoperability, the browser (in Javascript) or the server (in any 
language you wish) can generate SDP, if that's how they wish to 
interoperate.

Turning capability and control APIs into SDP is exactly the same problem 
as turning operating system APIs into SDP... the browser should be the 
new operating system, not just a hardcoded phone.

Matthew Kaufman


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 8/25/2011 5:51 AM, Elwell, John wrote:
>
> New requirements:
> Fyy1: The browser MUST be able to send in real-time to a third party media that are being transmitted to and received from remote participants.

This is a bad idea.

> Ayy1: The web application MUST be able to ask the browser to transmit in real-time to a third party media that are being transmitted to

Yes. It should be possible for the browser to send two copies of the 
same media to two different places, possibly even with different encoders.

>   and received from remote participants

But this is a bad idea. Providing APIs that let a browser send audio 
that is being received from the other end to a third party open several 
different cans of worms simultaneously. One is that there's now another 
path by which a user's media may be sent, possibly without the same 
security constraints, without their knowledge. Sure, a B2BUA can do this 
as well, but it makes doing the wrong thing easier.

The bigger issue is that this essentially allows you to use a browser as 
a media relay. This will be objected to in corporate (and some 
education) environments where there are sound reasons for disallowing 
user applications to take advantage of a high-bandwidth connection to 
relay media for third parties. It also may result in unexpected resource 
consumption on mobile platforms, where the user is paying in bandwidth 
charges and battery life to relay media for a third party. And unless 
implemented carefully, it can lead to relaying that is very unexpected 
(a banner ad that doesn't ask for permission to use your camera and 
microphone but does relay media for a third party while running).

Not to mention all the protocol-level implications of being an RTP 
mixer, if we're trying to stay true to that particular choice of protocols.

Matthew Kaufman

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On 08/25/2011 09:24 AM, Randell Jesup wrote:
> On 8/25/2011 11:55 AM, Matthew Kaufman wrote:
>
>> On 8/24/2011 4:36 AM, Harald Alvestrand wrote:
>
> Re: negotiate offer/answer separately in each direction
>
>
>>>
>>> I think that:
>>> a) this doesn't make sense - it's a completely new SDP/RTP practice,
>>> and we should not depart from established practice without a good
>>> reason; it also flies against the "keep the number of RTP sessions as
>>> low as we can" conclusion that came out of all the discussions about 
>>> ICE.
>>> b) it's not consistent with section 4.1.2 step 7.
>>>
>>> I think step 16 of section 4:
>>> "If connection's ICE started flag is still false, start the
>>> PeerConnection ICE Agent and send the initial offer. The initial offer
>>> must include a media description for the PeerConnection data UDP media
>>> stream, marked as "sendrecv", and for all the streams in localStreams
>>> (marked as "sendonly")."
>>>
>>> is neither correct nor complete.
>>
>> I agree that "this doesn't make sense" and it is just yet another reason
>> that I think SDP offer-answer is entirely inappropriate for WEBRTC.
>>
>> The web user agent should do what the web site's HTML and cooperating
>> Javascript tell it to do. It should not be engaged in direct negotiation
>> with the far end such that the outcome is either indeterminate or even
>> unexpected, except where direct negotiation is explicitly required to
>> meet a security requirement (the initial ICE handshake to determine that
>> it is permitted to send data to that endpoint).
>>
>> Note that any perceived gains by doing this negotiation (like "what if
>> my browser is on a slow connection and only wants to receive audio") are
>> immediately negated the moment the site changes the SDP enroute to add
>> "wants HD-resolution video" for you.
>
> Ok, so that's a bad web-app - don't use it.
>
>
> Are you really suggesting "send video to someone who doesn't want it 
> anyways"?


"Streaming ads" already exist, so there is this more common usage.


>
>
> "perceived gains" - sending me video at (say) 500K or more plus audio 
> at<50K,
> when I'm on a 128K link will kill my connection until I can somehow 
> get the
> other side to back off.  Horrid user experience.  Remember people without
> broadband or with limited broadband will be using this.  What if I 
> have 768
> down, 128 up - but Johnny in the his bedroom is watching youtube 
> videos or
> downloading torrents, etc?



If we redirect the discussion back to web-browsers (and not composition 
over the network) then we can merely point out the activity of multiple 
tabs, or why should each tab have equal bandwidth, and why should tabs 
not actively-viewed have more bandwidth than those actively viewed.

Magic solution would be uPNP negotiates frame rate.


>
>
>> In addition, because the spec is currently written with offer-answer, a
>> wide array of use cases that would be possible if capabilities were
>> instead exposed via Javascript become impossible. As an example, it
>> should be possible for me as a web site developer to create a page that
>> can determine, without prompting you to use your camera, whether or not
>> a camera is available and if so what codecs it supports. That way I can
>> put "I see you have a high-resolution camera and can encode H.264
>> video... click here to call a live agent who can help you find the exact
>> replacement part" for users who have that, and not if they don't have a
>> codec that works with my call center or a camera with sufficient
>> resolution to examine the parts I sell.
>
> In what way is that use-case blocked by offer-answer?  Access to the 
> video
> needs user confirmation; access to capabilities info shouldn't.  And 
> so the
> page can generate an offer with audio and video, or just audio if they 
> have
> no camera.  A user declining to send video doesn't necessarily mean you
> don't negotiate it - they advertise receive-only for video (if the web 
> app
> wants to).
>
>
> You state above there are major possibilities blocked by offer-answer 
> (I can
> think of some minor issues that with O-A require a second O-A pass) - 
> can you
> detail some use-cases so we can see the benefit and not just the 
> assertion?
> Then we could balance that against the advantage of O-A being 
> well-speced and
> known and implemented.  (And the issues surrounding how well and how
> interoperably web-app developers can implement their own capability
> negotiations/etc)
>
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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On 08/25/2011 10:14 AM, Matthew Kaufman wrote:
> On 8/25/2011 5:51 AM, Elwell, John wrote:
>>
>> New requirements:
>> Fyy1: The browser MUST be able to send in real-time to a third party 
>> media that are being transmitted to and received from remote 
>> participants.
>
> This is a bad idea.
>
>> Ayy1: The web application MUST be able to ask the browser to transmit 
>> in real-time to a third party media that are being transmitted to
>
> Yes. It should be possible for the browser to send two copies of the 
> same media to two different places, possibly even with different 
> encoders.
>
>>   and received from remote participants
>
> But this is a bad idea. Providing APIs that let a browser send audio 
> that is being received from the other end to a third party open 
> several different cans of worms simultaneously. One is that there's 
> now another path by which a user's media may be sent, possibly without 
> the same security constraints, without their knowledge. Sure, a B2BUA 
> can do this as well, but it makes doing the wrong thing easier.
>
> The bigger issue is that this essentially allows you to use a browser 
> as a media relay.


Or, ... picture in picture capability.



> This will be objected to in corporate (and some education) 
> environments where there are sound reasons for disallowing user 
> applications to take advantage of a high-bandwidth connection to relay 
> media for third parties. It also may result in unexpected resource 
> consumption on mobile platforms, where the user is paying in bandwidth 
> charges and battery life to relay media for a third party. And unless 
> implemented carefully, it can lead to relaying that is very unexpected 
> (a banner ad that doesn't ask for permission to use your camera and 
> microphone but does relay media for a third party while running).

The internet does not end at hyper-commercials over your regularly 
broadcasted network series.

Maybe browsers should be aware of the ads in prime schedules; pinpoint 
that junction.



>
> Not to mention all the protocol-level implications of being an RTP 
> mixer, if we're trying to stay true to that particular choice of 
> protocols.


My tone-deafness still desires "crystal-clear auido" (above 192Kbps per 
speaker), yet the browser narrowed that down to 192Kbps per stream and 
then that divides furthers into only some fraction per speaker. This has 
been hell with dysfunctional lectures due to sound quality, especially 
when ad-streams appear after links of interest activate.

Each speaker needs it's own IPv6 address, so we should imagine the 
browser provided by the speaker device, and paravirtualize for multiple 
speakers.

Oh the days of elementary schools that use any speaker as two way 
communication seems almost over.



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


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From pravindran@sonusnet.com  Thu Aug 25 11:06:29 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Nguyen Duong Tuan" <hoang.su.tk@gmail.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Some misunderstandings about <Usecase & architecture: Browser application with separate webserver & voipserver>
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Hi TuanND,

<snip>
b) Browser application interacts with VoIP server (use HTTP)
        browser<------>RTCweb server & SIP client (new GW)<------>SIP
server

New Gateway is outside the scope of RTCWeb and IETF which provides the
opportunity for innovative way of building gateways but may leads to
interop issue due to mapping between RTCWebserver & SIP client w.r.t
mid-call services.
</snip>

The issue in case webserver acts as new RTCWEB & SIP GW for VoIP
communication, there is a need of protocol mapping between the protocol
used between browser & RTCWebserver (say RTCweb protocol) to SIP. In my
experience with signaling protocol interworking, the interop between two
vendors fails lot of times in mid-call services like Hold, resume,
transfer and IETF BLISS WG is good example for such failures. My
question was why RTCWeb was looking into that approach.=20

>From http://www.ietf.org/mail-archive/web/rtcweb/current/msg00730.html
mail thread, I understand that more discussion will occur in RTCWeb to
clarify this doubt. Please let me know in case you need more
explanation.

Thanks
Partha

PS: I have changed my company recently

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
>Of Nguyen Duong Tuan
>Sent: Thursday, August 18, 2011 7:40 AM
>To: rtcweb@ietf.org
>Subject: [rtcweb] Some misunderstandings about <Usecase & architecture:
>Browser application with separate webserver & voipserver>
>
>Hello everyone,
>
>I read about Partha's question at RTCWeb mailing list at
>http://www.ietf.org/mail-archive/web/rtcweb/current/msg00563.html
>
>He said that there's interoprating issues due to mapping between
>RTCWebserver and SIP client w.r.t mid-call services in the
>introduction of new gateway. Could anyone please explain more about
>those issues? I have a little bit difficult to understand his
>meanings.
>
>I couldn't send an email to Partha directly by partr@cisco.com, so I
>decided to send to this email. Sorry for any inconvenience.
>
>Thanks,
>TuanND
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Dzonatas Sol" <dzonatas@gmail.com>, <rtcweb@ietf.org>, <matthew.kaufman@skype.net>
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Matthew,

Please read inline.

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
>Of Dzonatas Sol
>Sent: Thursday, August 25, 2011 11:39 PM
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
SIPREC
>session recording client
>
>On 08/25/2011 10:14 AM, Matthew Kaufman wrote:
>> On 8/25/2011 5:51 AM, Elwell, John wrote:
>>>
>>> New requirements:
>>> Fyy1: The browser MUST be able to send in real-time to a third party
>>> media that are being transmitted to and received from remote
>>> participants.
>>
>> This is a bad idea.
>>
[Partha] Please provide good idea if you have any.

>>> Ayy1: The web application MUST be able to ask the browser to
transmit
>>> in real-time to a third party media that are being transmitted to
>>
>> Yes. It should be possible for the browser to send two copies of the
>> same media to two different places, possibly even with different
>> encoders.
>>
>>>   and received from remote participants
>>
>> But this is a bad idea. Providing APIs that let a browser send audio
>> that is being received from the other end to a third party open
>> several different cans of worms simultaneously. One is that there's
>> now another path by which a user's media may be sent, possibly
without
>> the same security constraints, without their knowledge. Sure, a B2BUA
>> can do this as well, but it makes doing the wrong thing easier.
[Partha] In case your concern is "security consideration", let us work
for it to solve the security issues. Please specific the list of
security concerns related to this.

>>
>> The bigger issue is that this essentially allows you to use a browser
>> as a media relay.
>
[Partha] why it is issue?
>
>Or, ... picture in picture capability.
>
>
>
>> This will be objected to in corporate (and some education)
>> environments where there are sound reasons for disallowing user
>> applications to take advantage of a high-bandwidth connection to
relay
>> media for third parties.=20
[Partha] It is just a policy mechanism within the particular corporate
or education environment. Of course, any real-time application will
undergo the same policy constraints. For example, I know of couple of
corporate which does not permit Skype audio application to execute
within the corporate because it consumes lot of bandwidth. IMO, the
bandwidth consumption should not be stopping factor for adding the
requirement.

It also may result in unexpected resource
>> consumption on mobile platforms, where the user is paying in
bandwidth
>> charges and battery life to relay media for a third party. And unless
>> implemented carefully, it can lead to relaying that is very
unexpected
>> (a banner ad that doesn't ask for permission to use your camera and
>> microphone but does relay media for a third party while running).
>
[Partha] Same banner ad may send stream from your camera and microphone
when you are not ask for it as it is malicious script. I agree that you
concern is related to security in browser but your concern is not
related to recording requirement. Please clarify in case I'm missing
something.

>The internet does not end at hyper-commercials over your regularly
>broadcasted network series.
>
>Maybe browsers should be aware of the ads in prime schedules; pinpoint
>that junction.
>
>
>
>>
>> Not to mention all the protocol-level implications of being an RTP
>> mixer, if we're trying to stay true to that particular choice of
>> protocols.
>
>
>My tone-deafness still desires "crystal-clear auido" (above 192Kbps per
>speaker), yet the browser narrowed that down to 192Kbps per stream and
>then that divides furthers into only some fraction per speaker. This
has
>been hell with dysfunctional lectures due to sound quality, especially
>when ad-streams appear after links of interest activate.
>
>Each speaker needs it's own IPv6 address, so we should imagine the
>browser provided by the speaker device, and paravirtualize for multiple
>speakers.
>
>Oh the days of elementary schools that use any speaker as two way
>communication seems almost over.
>
>
>
>>
>> Matthew Kaufman
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
>
>--
>--- http://twitter.com/Dzonatas_Sol ---
>Web Development, Software Engineering
>Ag-Biotech, Virtual Reality, Consultant
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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Matthew Kaufman wrote:
> But this is a bad idea. Providing APIs that let a browser send audio
> that is being received from the other end to a third party open several
> different cans of worms simultaneously. One is that there's now another

The currently proposed MediaStream Processing API 
(http://hg.mozilla.org/users/rocallahan_mozilla.com/specs/raw-file/tip/StreamProcessing/StreamProcessing.html) 
essentially allows exactly this (including mixing). It doesn't make any 
distinction about whether the source of a MediaStream is a local camera, 
a <video> tag, or a remote stream from a PeerConnection object. So if 
you want to prevent this kind of thing, you need to have an active 
method of doing so, because by default the API will allow it.

> Not to mention all the protocol-level implications of being an RTP
> mixer, if we're trying to stay true to that particular choice of protocols.

That's a good point, and it's important to think about what the 
implications of those are, especially for a ProcessedMediaStream. This 
is not something I, personally, have thought all the way through yet.

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Subject: Re: [rtcweb] Some misunderstandings about <Usecase & architecture: Browser application with separate webserver & voipserver>
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> The issue in case webserver acts as new RTCWEB & SIP GW for VoIP
> communication=2C there is a need of protocol mapping between the protocol
> used between browser & RTCWebserver (say RTCweb protocol) to SIP.  [BA] N=
ot necessarily.  For example=2C in the case of XMPP=2C BOSH can be usedto e=
ncapsulate XMPP over HTTP.  The BOSH Connection Manager/Web serverdoes not =
do "mapping"=2C it just encapsulates/decapsulates XMPP stanzas=2C enabling =
communication between XMPP clients supporting BOSH and XMPP servers.   This=
 be handled entirely in Javascript with no nativesupport for XMPP.  For det=
ailed examples of how this works (with sample code)=2C please see:http://pr=
ofessionalxmpp.com/ 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
&gt=3B The issue in case webserver acts as new RTCWEB &amp=3B SIP GW for Vo=
IP<br>&gt=3B communication=2C there is a need of protocol mapping between t=
he protocol<br>&gt=3B used between browser &amp=3B RTCWebserver (say RTCweb=
 protocol) to SIP. <BR>&nbsp=3B<BR>[BA] Not necessarily.&nbsp=3B For exampl=
e=2C in the case of XMPP=2C BOSH can be used<BR>to encapsulate XMPP over HT=
TP.&nbsp=3B The BOSH Connection Manager/Web server<BR>does not do "mapping"=
=2C it just&nbsp=3Bencapsulates/decapsulates XMPP stanzas=2C <BR>enabling c=
ommunication&nbsp=3Bbetween XMPP clients supporting BOSH and <BR>XMPP serve=
rs.&nbsp=3B&nbsp=3B This be handled entirely in Javascript with no native<B=
R>support for XMPP. <BR>&nbsp=3B<BR>For detailed examples of how this works=
 (with sample code)=2C please see:<BR><a href=3D"http://professionalxmpp.co=
m/">http://professionalxmpp.com/</a><BR> 		 	   		  </div></body>
</html>=

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From pravindran@sonusnet.com  Thu Aug 25 12:05:54 2011
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Thread-Topic: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Elwell, John" <john.elwell@siemens-enterprise.com>, "Harald Alvestrand" <harald@alvestrand.no>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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John,

I like remote recording requirement. IMO, few changes may be required.

In SIPREC architecture, decomposed SRC with signaling handling in one
physical device (webserver in RTCWeb) and media handling in another
physical device(browser in RTCWeb) is possible but the communication
protocol between signaling server (webserver) and media server (browser)
is outside the scope of SIPREC. I agree that IETF Megaco (H.248) or IETF
Mediactrl of or equivalent proprietary mechanism between webserver &
browser will serve the purpose. IMO, it is better to spelt out
explicitly protocol mechanism between webserver & browser if this
architecture is chosen.

Actually, I thought of using browser as SIP UA & SRC which is not
consume more bandwidth compare to browser acts as media server because
number of signaling message compare to media messages are very low. I'm
discussing SIP UA based architecture because SIP as a protocol choice is
an open item in RTCWeb
(http://www.ietf.org/mail-archive/web/rtcweb/current/msg00730.html)

Thanks
Partha


>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
>Of Elwell, John
>Sent: Thursday, August 25, 2011 6:21 PM
>To: Harald Alvestrand; rtcweb@ietf.org
>Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
SIPREC
>session recording client
>
>
>
>
>> -----Original Message-----
>> From: rtcweb-bounces@ietf.org
>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestrand
>> Sent: 25 August 2011 13:02
>> To: rtcweb@ietf.org
>> Subject: Re: [rtcweb] Remote recording - RTC-Web client
>> acting as SIPREC session recording client
>>
>> On 08/24/11 21:06, Paul Kyzivat wrote:
>> > On 8/24/11 5:32 AM, Hutton, Andrew wrote:
>> >> Randell,
>> >>
>> >> I would like to make it clear that I am not saying that
>> SIPREC is a
>> >> reason for putting SIP in the browser but if SIP is not in the
>> >> browser then we need to be able to build applications like
>> a SIPREC
>> >> SRC using a combination of the browser and the web application.
>> >>
>> >> Therefore I do think that the remote recording case is a
>> useful use
>> >> case to explore and should not be out of scope as it helps us
>> >> understand the type of media handling requirements that
>> the browser
>> >> needs to support innovative applications.
>> >>
>> >> Of course as John stated if a decision is made to put SIP in the
>> >> browser then there would be additional browser and API
>> requirements
>> >> to support SIPREC.
>> >
>> > I agree with John and Andrew.
>> >
>> > Certainly it is possible to punt and say that if you want recording
>> > then you need to insert a B2BUA that is in the media path and that
>> > serves as the SRC. But before deciding that is "the story" for
>> > recording with RTCWEB, it would be good to understand the
>> implications
>> > of that approach.
>> >
>> > Since this is RTCWEB, I guess we can assume that there is a
>> web server
>> > interacting with a browser. The browser will provide one
>> end to some
>> > media streams. There will be another end to those media
>> streams, which
>> > might be in another browser talking to the same web server,
>> or the web
>> > server might have those media streams attached to a sip session. In
>> > some of the cases of interest, the web server won't itself
>> ever touch
>> > the actual media.
>> >
>> > If the web server wants to record the media, it needs to get an SRC
>> > into the signaling and media paths. If the session is
>> already going to
>> > a sip server, then its probably not to hard to integrate SRC
>> > functionality there or juggle the signaling paths to involve a free
>> > standing SRC into the call path.
>> >
>> > But if the web server is orchestrating media streams
>> between browsers
>> > talking to it, without any sip, then adding recording will be more
>> > complex. It may need to pull in a sip server that can be an SRC,
>> > reroute the media through it, etc. Its not evident to me if
>> this would
>> > be straightforward or not. I suspect not. It would be good to
>> > understand what would need to be done.
>> >
>> > As John and/or Andrew mentioned, it could be advantageous if there
>> > were primitives by which the JavaScript in the browser had
>> simple ways
>> > to fork the media and direct it different ways. That could
>> eliminate
>> > the need to perturb the normal media path in order to do recording.
>> If we can get the signalling part done through some channel
>> that doesn't
>> touch the RTCWEB API, it may be enough to say something like:
>>
>> The API MUST allow the application to specify that a copy of a media
>> stream (incoming or outgoing) is sent out to some other
correspondent.
>[JRE] This is the essential part of what I was trying to get across
when
>I originally proposed text for the use cases document. I reproduce it
>below, with a slight change to refer to a third party rather than a
>recording device:
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>4.2.yy Remote Session Recording
>In this use case, the web application user wishes to record a real-time
>communication at a remote third party (e.g., a recording device), such
>that transmitted and received audio, video or other real-time media are
>transmitted in real-time to the third party. The third party can
perform
>recording, archiving, retrieval, playback, etc., but can also perform
>real-time analytics on the media. A typical deployment might be in a
>contact centre. For a given medium, the two directions of transmission
>can be transmitted together (mixed) or separately. The web application
>also sends metadata that gives context to the stored media.
>
>New requirements:
>Fyy1: The browser MUST be able to send in real-time to a third party
>media that are being transmitted to and received from remote
>participants.
>
>Ayy1: The web application MUST be able to ask the browser to transmit
in
>real-time to a third party media that are being transmitted to and
>received from remote participants and, in the case of audio at least,
>ask for the two directions of transmission to be transmitted to the
>remote recording device mixed or separately.
>=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
>
>I think it desirable that the application have control over whether to
>send mixed or unmixed, but we can certainly discuss that. It also has
>implications for the RTP model, because mixing implies an RTP mixer or
>endpoint model, rather than just relaying RTP packets. Other aspects
>such as whether a different security context is needed will also have
>impact.
>
>>
>> In PeerConnection API terms, that means that some construct like this
>> should work:
>>
>>     mainConn =3D new PeerConnection(...)
>> <connect mainConn to remote participant, producing remoteStream and
>> localStream>
>>     recorderConn =3D new PeerConnection(...)
>> <connect recorderConn to remote recorder>
>>     recorderConn.addStream(remoteStream)
>>     recorderConn.addStream(localStream)
>>
>> The important part of this, implementation-wise, is that one stream
>> needs to be able to have several destinations. What that
>> means for codec
>> negotiations, transcoding-or-not and so on may perhaps best be kept
>> under the covers.
>[JRE] By "under covers", I assume you mean not exposed at the API. That
>might be sufficient, but the implications need thinking about.
>
>John
>
>> >
>> >     Thanks,
>> >     Paul
>> >
>> >> Regards
>> >> Andy
>> >>
>> >>
>> >>
>> >>> -----Original Message-----
>> >>> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
>> >>> Behalf Of Elwell, John
>> >>> Sent: 24 August 2011 09:24
>> >>> To: Randell Jesup; rtcweb@ietf.org
>> >>> Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
>> >>> SIPREC session recording client
>> >>>
>> >>> Randell,
>> >>>
>> >>>> -----Original Message-----
>> >>>> From: rtcweb-bounces@ietf.org
>> >>>> [mailto:rtcweb-bounces@ietf.org] On Behalf Of Randell Jesup
>> >>>> Sent: 24 August 2011 08:43
>> >>>> To: rtcweb@ietf.org
>> >>>> Subject: Re: [rtcweb] Remote recording - RTC-Web client
>> >>>> acting as SIPREC session recording client
>> >>>>
>> >>>> On 8/23/2011 1:50 PM, Hutton, Andrew wrote:
>> >>>>
>> >>>>> Hi,
>> >>>>>
>> >>>>> Also agree that implementing a full SIP Session Recording
>> >>>> Client (SRC) as defined ]
>> >>>>> by the SIPREC WG in to the browser would be a tall order
>> >>>> and would open the flood
>> >>>>> gates for a lot of other requests for SIP functionality in
>> >>>> the browser.
>> >>>>
>> >>>> Agreed
>> >>>>
>> >>>>> If however SIP is not in the browser then I think it is a
>> >>>> useful and interesting test case
>> >>>>> to look at whether we could build a decomposed SRC with the
>> >>>> browser handling the
>> >>>>> media and the web server handling the signalling. What I
>> >>>> heard a number of times at
>> >>>>> IETF81 is that what people want is to build innovative new
>> >>>> applications with a real time
>> >>>>> media enabled browser and to it seems clear to me we need a
>> >>>> browser API which is
>> >>>>> flexible and enables applications to do clever things with
>> >>>> media streams. So exploring
>> >>>>> some use cases which require the browser to duplicate and
>> >>>> copy media streams is a good idea.
>> >>>>>
>> >>>>> Therefore I think we should keep explore the remote
>> >>>> recording use case further.
>> >>>>
>> >>>> My personal (not Mozilla's) opinion is that SIPREC-style
>> recording is
>> >>>> out-of-scope,
>> >>>> and would be handled in this context by a webrtc-B2BUA (i.e.
>> >>>> something
>> >>>> that sits on the
>> >>>> signalling and media paths), or something that interacts
>> >>>> directly with
>> >>>> the app.
>> >>>>
>> >>>> What I do want to consider for in-scope are local
>> recording options.
>> >>>> This can cover the "local
>> >>>> answering machine" case, but also "I want to record my call
>> >>>> with Dad",
>> >>>> "I want to record
>> >>>> my team's chatter in the game", "I want to interview someone for
>> >>>> genealogy research", etc, etc.
>> >>>> Yes, these can be done (poorly) by using some external
>> app to capture
>> >>>> the screen and
>> >>>> speaker/mic audio - I don't consider that a replacement for
>> >>>> local recording.
>> >>>>
>> >>>> Local recording is far less problematical protocol-wise than
>> >>>> SIPREC, and
>> >>>> doesn't require
>> >>>> mandating SIP.
>> >>> [JRE] I am not advocating supporting SIP in the browser
>> just for the
>> >>> purpose of supporting SIPREC SRC functionality. What I am
>> saying is
>> >>> that if the browser is concerned only with media aspects,
>> then only the
>> >>> media aspects of SRC functionality need to be considered
>> in the browser
>> >>> (e.g., copying the media streams). On the other hand, if,
>> for other
>> >>> reasons, SIP is implemented in the browser, it would
>> require additional
>> >>> enhancements to support SIP aspects of SIPREC SRC functionality.
>> >>>
>> >>> John
>> >>>
>> >>> John Elwell
>> >>> Tel: +44 1908 817801 (office and mobile)
>> >>> Email: john.elwell@siemens-enterprise.com
>> >>> http://www.siemens-enterprise.com/uk/
>> >>>
>> >>> Siemens Enterprise Communications Limited.
>> >>> Registered office: Brickhill Street, Willen Lake, Milton
>> Keynes, MK15
>> >>> 0DJ.
>> >>> Registered No: 5903714, England.
>> >>>
>> >>> Siemens Enterprise Communications Limited is a Trademark
>> Licensee of
>> >>> Siemens AG.
>> >>>
>> >>>
>> >>>>
>> >>>> --
>> >>>> Randell Jesup
>> >>>> randell-ietf@jesup.org
>> >>>>
>> >>>> _______________________________________________
>> >>>> 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  Thu Aug 25 12:15:16 2011
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From: "Ravindran Parthasarathi" <pravindran@sonusnet.com>
To: "Timothy B. Terriberry" <tterriberry@mozilla.com>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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Hi Timothy,

In principle, mixing is not required to be done by browser because lot
of real-time media analytical tool will be interested in un-mixed media
stream. Browser shall act as RTP translator or RTP end-point to meet the
requirement of recording and You may wish to look into Real-time
Transport Protocol (RTP) Recommendations for SIPREC draft
(https://datatracker.ietf.org/doc/draft-eckel-siprec-rtp-rec/).=20

Thanks
Partha

>-----Original Message-----
>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
Behalf
>Of Timothy B. Terriberry
>Sent: Friday, August 26, 2011 12:21 AM
>Cc: rtcweb@ietf.org
>Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as
SIPREC
>session recording client
>
>Matthew Kaufman wrote:
>> But this is a bad idea. Providing APIs that let a browser send audio
>> that is being received from the other end to a third party open
>several
>> different cans of worms simultaneously. One is that there's now
>another
>
>The currently proposed MediaStream Processing API
>(http://hg.mozilla.org/users/rocallahan_mozilla.com/specs/raw-
>file/tip/StreamProcessing/StreamProcessing.html)
>essentially allows exactly this (including mixing). It doesn't make any
>distinction about whether the source of a MediaStream is a local
camera,
>a <video> tag, or a remote stream from a PeerConnection object. So if
>you want to prevent this kind of thing, you need to have an active
>method of doing so, because by default the API will allow it.
>
>> Not to mention all the protocol-level implications of being an RTP
>> mixer, if we're trying to stay true to that particular choice of
>protocols.
>
>That's a good point, and it's important to think about what the
>implications of those are, especially for a ProcessedMediaStream. This
>is not something I, personally, have thought all the way through yet.
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

From bernard_aboba@hotmail.com  Thu Aug 25 12:24:59 2011
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The ICE offerer needs to be prepared to receive media on the ports indicate=
d in the offer.  After all=2C it doesn't know whether the answer will indic=
ate support for TOGETHER or not.  As a result=2C without any a-priori knowl=
edge=2C it seems to me that normal behavior with respect to gathering candi=
dates and preparing to receive media should apply.  Once an answer is retur=
ned that indicates TOGETHER support=2C then the offerer will know that medi=
a won't be received on the ports indicated in the m-lines other than the fi=
rst.  In general=2C we'd probably like to conclude the ICE exchange as quic=
kly as possible=2C so that setting ports to zero in all but the first m-lin=
e and then going through another offer-answer exchange is probably non-opti=
mal.    One concern that hasn't been mentioned so far is the interaction of=
 TOGETHER with other multi-stream work now in progress=2C including CLUE an=
d draft-westerlund-avtcore-multistream-and-simulcast  .  Do we understand h=
ow these might interact?   Paul Kyzivat said: IIUC=2C there are some aspect=
s of ICE that MUST be done before sending the=20
offer. Specifically that involved gathering the candidate list=2C which=20
may involve assignment of multiple local ports=2C interactions with a=20
TURN server=2C etc. I don't think everything can be deferred until the=20
answer is received. (I suppose one approach would be for the initial offer =
to set the port=20
to zero in all but the first m-line. Then=2C once the first answer is=20
received=2C indicating whether TOGETHER is supported=2C another offer can=20
be generated=2C using appropriate ICE based on whether using TOGETHER or=20
not. But that does start to get complex.)  		 	   		  =

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<html>
<head>
<style><!--
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margin:0px=3B
padding:0px
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body.hmmessage
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font-size: 10pt=3B
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<body class=3D'hmmessage'><div dir=3D'ltr'>



The ICE offerer needs to be prepared to receive media on the ports indicate=
d in the offer.&nbsp=3B After all=2C it doesn't know whether the answer wil=
l indicate support for TOGETHER or not.&nbsp=3B&nbsp=3BAs a result=2C witho=
ut any a-priori knowledge=2C it seems to me that normal behavior with respe=
ct to gathering candidates and preparing to receive media should apply.&nbs=
p=3B Once an answer is returned that indicates TOGETHER support=2C then the=
 offerer will know that media won't be received on the ports indicated in t=
he m-lines other than the first. <BR>&nbsp=3B<BR>In general=2C we'd probabl=
y like to conclude the ICE exchange as quickly as possible=2C so that setti=
ng ports to zero in all but the first m-line&nbsp=3Band then going through =
another offer-answer exchange is probably non-optimal. &nbsp=3B&nbsp=3B<BR>=
&nbsp=3B<BR>One concern that hasn't been mentioned so far is the interactio=
n of TOGETHER with other multi-stream work now in progress=2C including CLU=
E and&nbsp=3Bdraft-westerlund-avtcore-multistream-and-simulcast&nbsp=3B .&n=
bsp=3B Do we understand how these might&nbsp=3Binteract? <BR>&nbsp=3B<BR>&n=
bsp=3B<BR>Paul Kyzivat said:<BR>&nbsp=3B<BR><font face=3D"Courier New">IIUC=
=2C there are some aspects of ICE that MUST be done before sending the=20
<tt>offer. Specifically that involved gathering the candidate list=2C which=
=20
</tt><tt>may involve assignment of multiple local ports=2C interactions wit=
h a=20
TURN </tt><tt>server=2C etc. I don't think everything can be deferred until=
 the=20
answer </tt><tt>is received. </tt></font><BR><pre style=3D"margin: 0em=3B">=
</pre><tt>(I suppose one approach would be for the initial offer to set the=
 port=20
</tt><tt>to zero in all but the first m-line. Then=2C once the first answer=
 is=20
</tt><tt>received=2C indicating whether TOGETHER is supported=2C another of=
fer can=20
be </tt><tt>generated=2C using appropriate ICE based on whether using TOGET=
HER or=20
not. </tt><tt>But that does start to get complex.) </tt><BR> 		 	   		  </d=
iv></body>
</html>=

--_45d92119-f67f-4b93-9e04-7ce8cf29201f_--

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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 8/25/2011 11:29 AM, Ravindran Parthasarathi wrote:
>
>> On 08/25/2011 10:14 AM, Matthew Kaufman wrote:
>>> On 8/25/2011 5:51 AM, Elwell, John wrote:
>>>> New requirements:
>>>> Fyy1: The browser MUST be able to send in real-time to a third party
>>>> media that are being transmitted to and received from remote
>>>> participants.
>>> This is a bad idea.
>>>
> [Partha] Please provide good idea if you have any.
>

Remove the part of the requirement that a browser be able to send to a 
third party media that is being received from anywhere but the 
microphone and camera.

>>>> Ayy1: The web application MUST be able to ask the browser to
> transmit
>>>> in real-time to a third party media that are being transmitted to
>>> Yes. It should be possible for the browser to send two copies of the
>>> same media to two different places, possibly even with different
>>> encoders.
>>>
>>>>    and received from remote participants
>>> But this is a bad idea. Providing APIs that let a browser send audio
>>> that is being received from the other end to a third party open
>>> several different cans of worms simultaneously. One is that there's
>>> now another path by which a user's media may be sent, possibly
> without
>>> the same security constraints, without their knowledge. Sure, a B2BUA
>>> can do this as well, but it makes doing the wrong thing easier.
> [Partha] In case your concern is "security consideration", let us work
> for it to solve the security issues. Please specific the list of
> security concerns related to this.

The primary security concern is that it allows for untrusted endpoints 
(browsers) to be trivially manipulated into sending data they receive 
onward. This is worse than manipulating my local browser, because I can 
at least observe the operation of my local browser (how much traffic it 
sends, what connections are brought up, etc.).

>>> The bigger issue is that this essentially allows you to use a browser
>>> as a media relay.
> [Partha] why it is issue?

I already enumerated this in my previous email. For some examples:

1. Browser in a high-upload-bandwidth environment (like a corporation or 
university) will be used to relay media for many other participants who 
do not have sufficient bandwidth, transferring the cost from the service 
provider to users in those environments.

2. Browser in an expensive-upload-bandwidth environment (metered 
billing) may relay media without its knowledge, increasing the bandwidth 
bill.

3. Mobile browser may relay media without its knowledge, decreasing 
battery life, possibly reaching provider bandwidth caps or charges for 
metered bandwidth.



>> Or, ... picture in picture capability.
>>
>>
>>
>>> This will be objected to in corporate (and some education)
>>> environments where there are sound reasons for disallowing user
>>> applications to take advantage of a high-bandwidth connection to
> relay
>>> media for third parties.
> [Partha] It is just a policy mechanism within the particular corporate
> or education environment. Of course, any real-time application will
> undergo the same policy constraints. For example, I know of couple of
> corporate which does not permit Skype audio application to execute
> within the corporate because it consumes lot of bandwidth. IMO, the
> bandwidth consumption should not be stopping factor for adding the
> requirement.

Do you really want certain web browsers banned from the same 
environments because of the same concerns?

Getting ubiquitous deployment will require complying with norms about 
what a browser does and doesn't do. Right now one of the things it 
*doesn't* do is relay media unexpectedly.

> It also may result in unexpected resource
>>> consumption on mobile platforms, where the user is paying in
> bandwidth
>>> charges and battery life to relay media for a third party. And unless
>>> implemented carefully, it can lead to relaying that is very
> unexpected
>>> (a banner ad that doesn't ask for permission to use your camera and
>>> microphone but does relay media for a third party while running).
> [Partha] Same banner ad may send stream from your camera and microphone
> when you are not ask for it as it is malicious script.

Disagree. These will cause your camera and microphone to be accessed, 
which will cause a permission dialog to appear. ALSO, there is no 
financial benefit to causing your own media to be sent when you're not 
on a call (even if it was possible) but a huge financial benefit to use 
nodes with excess bandwidth as relays.

>   I agree that you
> concern is related to security in browser but your concern is not
> related to recording requirement. Please clarify in case I'm missing
> something.

It is related to being able to command a browser to send media *that it 
is receiving* to another place. I have no problem with a call from A to 
B being recording by having A send a copy of its media to C and B send a 
copy of its media to C. But recording should not be accomplished by 
having B send a copy of its media *and what it receives from A* to C.

Matthew Kaufman


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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On 8/25/2011 11:50 AM, Timothy B. Terriberry wrote:
> Matthew Kaufman wrote:
>> But this is a bad idea. Providing APIs that let a browser send audio
>> that is being received from the other end to a third party open several
>> different cans of worms simultaneously. One is that there's now another
>
> The currently proposed MediaStream Processing API 
> (http://hg.mozilla.org/users/rocallahan_mozilla.com/specs/raw-file/tip/StreamProcessing/StreamProcessing.html) 
> essentially allows exactly this (including mixing). It doesn't make 
> any distinction about whether the source of a MediaStream is a local 
> camera, a <video> tag, or a remote stream from a PeerConnection 
> object. So if you want to prevent this kind of thing, you need to have 
> an active method of doing so, because by default the API will allow it.

I think you need to seriously consider the security implications here. 
Any media that originates from somewhere other than a local camera that 
has given permission or a local microphone that has given permission 
needs to be marked as not sendable elsewhere.
>
>> Not to mention all the protocol-level implications of being an RTP
>> mixer, if we're trying to stay true to that particular choice of 
>> protocols.
>
> That's a good point, and it's important to think about what the 
> implications of those are, especially for a ProcessedMediaStream. This 
> is not something I, personally, have thought all the way through yet.

Agree, though less of an issue if it isn't possible.

Matthew Kaufman


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> > Matthew Kaufman wrote:
> >> But this is a bad idea. Providing APIs that let a browser send audio
> >> that is being received from the other end to a third party open severa=
l
> >> different cans of worms simultaneously.

> I think you need to seriously consider the security implications here.=20
> Any media that originates from somewhere other than a local camera that=20
> has given permission or a local microphone that has given permission=20
> needs to be marked as not sendable elsewhere.

[BA] Right.  As an example=2C the ability to send pre-recorded audio/video =
within an emergency call could be used to launch "swatting" attacks with fr=
ightening realism.
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B &gt=3B Matthew Kaufman wrote:<br>&gt=3B &gt=3B&gt=3B But th=
is is a bad idea. Providing APIs that let a browser send audio<br>&gt=3B &g=
t=3B&gt=3B that is being received from the other end to a third party open =
several<br>&gt=3B &gt=3B&gt=3B different cans of worms simultaneously.<br><=
br>&gt=3B I think you need to seriously consider the security implications =
here. <br>&gt=3B Any media that originates from somewhere other than a loca=
l camera that <br>&gt=3B has given permission or a local microphone that ha=
s given permission <br>&gt=3B needs to be marked as not sendable elsewhere.=
<br><br>[BA] Right.&nbsp=3B As an example=2C the ability to send pre-record=
ed audio/video within an emergency call could be used to launch "swatting" =
attacks with frightening realism.<br></div> 		 	   		  </div></body>
</html>=

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On Thu, Aug 25, 2011 at 1:04 PM, Matthew Kaufman
<matthew.kaufman@skype.net>wrote:

> On 8/25/2011 9:24 AM, Randell Jesup wrote:
>
>> On 8/25/2011 11:55 AM, Matthew Kaufman wrote:
>>
>>  On 8/24/2011 4:36 AM, Harald Alvestrand wrote:
>>>
>>
>> Re: negotiate offer/answer separately in each direction
>>
>>
>>
>>>> I think that:
>>>> a) this doesn't make sense - it's a completely new SDP/RTP practice,
>>>> and we should not depart from established practice without a good
>>>> reason; it also flies against the "keep the number of RTP sessions as
>>>> low as we can" conclusion that came out of all the discussions about
>>>> ICE.
>>>> b) it's not consistent with section 4.1.2 step 7.
>>>>
>>>> I think step 16 of section 4:
>>>> "If connection's ICE started flag is still false, start the
>>>> PeerConnection ICE Agent and send the initial offer. The initial offer
>>>> must include a media description for the PeerConnection data UDP media
>>>> stream, marked as "sendrecv", and for all the streams in localStreams
>>>> (marked as "sendonly")."
>>>>
>>>> is neither correct nor complete.
>>>>
>>>
>>> I agree that "this doesn't make sense" and it is just yet another reason
>>> that I think SDP offer-answer is entirely inappropriate for WEBRTC.
>>>
>>> The web user agent should do what the web site's HTML and cooperating
>>> Javascript tell it to do. It should not be engaged in direct negotiation
>>> with the far end such that the outcome is either indeterminate or even
>>> unexpected, except where direct negotiation is explicitly required to
>>> meet a security requirement (the initial ICE handshake to determine that
>>> it is permitted to send data to that endpoint).
>>>
>>> Note that any perceived gains by doing this negotiation (like "what if
>>> my browser is on a slow connection and only wants to receive audio") are
>>> immediately negated the moment the site changes the SDP enroute to add
>>> "wants HD-resolution video" for you.
>>>
>>
>> Ok, so that's a bad web-app - don't use it.
>>
>>
>> Are you really suggesting "send video to someone who doesn't want it
>> anyways"?
>>
>
> No, I don't think that's a good idea. But one of the arguments I've heard
> *for* doing O-A between the two ends is that it somehow ensures that the
> sender won't send things that the receiver isn't prepared to receive.
>
> I was simply pointing out that this isn't true at all. Because the O-A can
> easily be modified in-transit, either end can be trivially convinced to do
> things it otherwise shouldn't be doing.
>
>
>
>>
>> "perceived gains" - sending me video at (say) 500K or more plus audio
>> at<50K,
>> when I'm on a 128K link will kill my connection until I can somehow get
>> the
>> other side to back off.  Horrid user experience.
>>
>
> Yes. And it will be possible for a "bad web site" to do this whether we use
> O-A or something else.
>
>
>  Remember people without
>> broadband or with limited broadband will be using this.  What if I have
>> 768
>> down, 128 up - but Johnny in the his bedroom is watching youtube videos or
>> downloading torrents, etc?
>>
>
> In theory the sender will back off once they get reports of massive loss,
> assuming we do congestion control for the media. If not, they you're both
> out of luck. No different really than anything else though... I can easily
> build a web site that lets you connect over HTTP and then runs a modified
> TCP that doesn't slow down when there's loss.
>
>
>
>
>>
>>  In addition, because the spec is currently written with offer-answer, a
>>> wide array of use cases that would be possible if capabilities were
>>> instead exposed via Javascript become impossible. As an example, it
>>> should be possible for me as a web site developer to create a page that
>>> can determine, without prompting you to use your camera, whether or not
>>> a camera is available and if so what codecs it supports. That way I can
>>> put "I see you have a high-resolution camera and can encode H.264
>>> video... click here to call a live agent who can help you find the exact
>>> replacement part" for users who have that, and not if they don't have a
>>> codec that works with my call center or a camera with sufficient
>>> resolution to examine the parts I sell.
>>>
>>
>> In what way is that use-case blocked by offer-answer?  Access to the video
>> needs user confirmation; access to capabilities info shouldn't.
>>
>
> I'm reading http://dev.w3.org/2011/webrtc/**editor/webrtc.html<http://dev.w3.org/2011/webrtc/editor/webrtc.html>
>
> I cannot see how to get it to generate an SDP offer before I try to open a
> media connection.
>
> I also cannot see how one could possibly know *what* SDP to offer until you
> call "getUserMedia", which prompts the user. (As an example, I have two
> cameras attached to this computer over USB. One of them has an on-board
> H.264 encoder. The other does not. If my browser can do pass-through of the
> encoded H.264 but doesn't have its own H.264 encoder, what offer do I
> generate before the camera is selected? SDP doesn't have a good way to
> encode "maybe".)
>
> What we need are a set of Javascript APIs that let us enumerate the
> available audio input devices and encoders, video input devices and
> encoders, audio output devices and decoders, video output devices and
> decoders. I hate to say it but even H.245 is probably a better model for how
> to collect the capabilities than SDP O-A.
>
>
>  And so the
>> page can generate an offer with audio and video, or just audio if they
>> have
>> no camera.  A user declining to send video doesn't necessarily mean you
>> don't negotiate it - they advertise receive-only for video (if the web app
>> wants to).
>>
>>
> I don't understand what you mean by "(if the web app wants to)"... the
> current proposed specification doesn't have a way for the web app to control
> whether the SDP offer is receive-only for video or not.
>
>
>
>> You state above there are major possibilities blocked by offer-answer (I
>> can
>> think of some minor issues that with O-A require a second O-A pass) - can
>> you
>> detail some use-cases so we can see the benefit and not just the
>> assertion?
>>
>
> It is certainly possible that a repeated set of fake offer-answer exchanges
> can be used by either Javascript or the server to determine what the
> capabilities are, and that a set of rich Javascript-exposed capabilities may
> be used to generate SDP offers and answers, so in that sense they are
> mappable from one to another.
>
> But I would argue that turning capabilities into SDP is easier than the
> reverse. This is really a question of whether we are trying to turn the
> browser into a *platform for building applications* (in which case we should
> be exposing, as an operating system does, APIs for determining what is
> possible and APIs for control) or turning the browser into *a phone*, in
> which case sure, SIP and SDP are both fine choices.
>

I think it makes sense for the browser to emit capabilities, which could
then be used by the web app to generate a SDP offer or answer. This provides
the web app with more control over the functionality; the browser just
reacts to what the web app asks it to do, rather than trying to make
decisions (such as how the answer should be generated from the offer) that
may not be possible without context.

The original problem that started this email is one specific example - if
the callee application wants to only receive audio, the application can
generate an audio-only SDP based on the offer, the browser capabilities, and
the desired app behavior - without any new APIs in the browser.

>
> I think there's a whole lot of potential applications beyond what we're
> currently thinking of if we provide a platform and not just a phone.
>
> I've already outlined one case (offering the user the ability to place a
> video call *if* there is an acceptable camera and encoder on the system).
>
> Another obvious one is where there's a dozen people already in a video call
> and one more wishes to join... if the server knows what capabilities exist,
> the server can tell the new joiner "your browser can't support video in a
> format that the other users need" or can tell the existing user browsers to
> switch to a different codec that is now compatible with everyone or
> whatever, rather than having to re-run offer-answer with every party.




>
>
>  Then we could balance that against the advantage of O-A being well-speced
>> and
>> known and implemented.
>>
>
> But it *isn't* well-speced for use cases other than one-to-one calling,
> really. It works poorly for recording, it works poorly for large multi-party
> conferences, etc.
>
> And on top of that, it is missing important attributes that we'll want to
> control (like whether the Opus codec is being forced to "music" mode or
> not).
>
>
>  (And the issues surrounding how well and how
>> interoperably web-app developers can implement their own capability
>> negotiations/etc)
>>
>>
>>
> For interoperability, the browser (in Javascript) or the server (in any
> language you wish) can generate SDP, if that's how they wish to
> interoperate.
>
> Turning capability and control APIs into SDP is exactly the same problem as
> turning operating system APIs into SDP... the browser should be the new
> operating system, not just a hardcoded phone.
>
> Matthew Kaufman
>
>
> ______________________________**_________________
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> rtcweb@ietf.org
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<br><br><div class=3D"gmail_quote">On Thu, Aug 25, 2011 at 1:04 PM, Matthew=
 Kaufman <span dir=3D"ltr">&lt;<a href=3D"mailto:matthew.kaufman@skype.net"=
>matthew.kaufman@skype.net</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-le=
ft:1ex;">

<div class=3D"HOEnZb"><div></div><div class=3D"h5">On 8/25/2011 9:24 AM, Ra=
ndell Jesup wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 8/25/2011 11:55 AM, Matthew Kaufman wrote:<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 8/24/2011 4:36 AM, Harald Alvestrand wrote:<br>
</blockquote>
<br>
Re: negotiate offer/answer separately in each direction<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
I think that:<br>
a) this doesn&#39;t make sense - it&#39;s a completely new SDP/RTP practice=
,<br>
and we should not depart from established practice without a good<br>
reason; it also flies against the &quot;keep the number of RTP sessions as<=
br>
low as we can&quot; conclusion that came out of all the discussions about I=
CE.<br>
b) it&#39;s not consistent with section 4.1.2 step 7.<br>
<br>
I think step 16 of section 4:<br>
&quot;If connection&#39;s ICE started flag is still false, start the<br>
PeerConnection ICE Agent and send the initial offer. The initial offer<br>
must include a media description for the PeerConnection data UDP media<br>
stream, marked as &quot;sendrecv&quot;, and for all the streams in localStr=
eams<br>
(marked as &quot;sendonly&quot;).&quot;<br>
<br>
is neither correct nor complete.<br>
</blockquote>
<br>
I agree that &quot;this doesn&#39;t make sense&quot; and it is just yet ano=
ther reason<br>
that I think SDP offer-answer is entirely inappropriate for WEBRTC.<br>
<br>
The web user agent should do what the web site&#39;s HTML and cooperating<b=
r>
Javascript tell it to do. It should not be engaged in direct negotiation<br=
>
with the far end such that the outcome is either indeterminate or even<br>
unexpected, except where direct negotiation is explicitly required to<br>
meet a security requirement (the initial ICE handshake to determine that<br=
>
it is permitted to send data to that endpoint).<br>
<br>
Note that any perceived gains by doing this negotiation (like &quot;what if=
<br>
my browser is on a slow connection and only wants to receive audio&quot;) a=
re<br>
immediately negated the moment the site changes the SDP enroute to add<br>
&quot;wants HD-resolution video&quot; for you.<br>
</blockquote>
<br>
Ok, so that&#39;s a bad web-app - don&#39;t use it.<br>
<br>
<br>
Are you really suggesting &quot;send video to someone who doesn&#39;t want =
it anyways&quot;?<br>
</blockquote>
<br></div></div>
No, I don&#39;t think that&#39;s a good idea. But one of the arguments I&#3=
9;ve heard *for* doing O-A between the two ends is that it somehow ensures =
that the sender won&#39;t send things that the receiver isn&#39;t prepared =
to receive.<br>


<br>
I was simply pointing out that this isn&#39;t true at all. Because the O-A =
can easily be modified in-transit, either end can be trivially convinced to=
 do things it otherwise shouldn&#39;t be doing.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
&quot;perceived gains&quot; - sending me video at (say) 500K or more plus a=
udio at&lt;50K,<br>
when I&#39;m on a 128K link will kill my connection until I can somehow get=
 the<br>
other side to back off. =C2=A0Horrid user experience.<br>
</blockquote>
<br></div>
Yes. And it will be possible for a &quot;bad web site&quot; to do this whet=
her we use O-A or something else.<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Remember people without<br>
broadband or with limited broadband will be using this. =C2=A0What if I hav=
e 768<br>
down, 128 up - but Johnny in the his bedroom is watching youtube videos or<=
br>
downloading torrents, etc?<br>
</blockquote>
<br></div>
In theory the sender will back off once they get reports of massive loss, a=
ssuming we do congestion control for the media. If not, they you&#39;re bot=
h out of luck. No different really than anything else though... I can easil=
y build a web site that lets you connect over HTTP and then runs a modified=
 TCP that doesn&#39;t slow down when there&#39;s loss.<div class=3D"im">

<br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
In addition, because the spec is currently written with offer-answer, a<br>
wide array of use cases that would be possible if capabilities were<br>
instead exposed via Javascript become impossible. As an example, it<br>
should be possible for me as a web site developer to create a page that<br>
can determine, without prompting you to use your camera, whether or not<br>
a camera is available and if so what codecs it supports. That way I can<br>
put &quot;I see you have a high-resolution camera and can encode H.264<br>
video... click here to call a live agent who can help you find the exact<br=
>
replacement part&quot; for users who have that, and not if they don&#39;t h=
ave a<br>
codec that works with my call center or a camera with sufficient<br>
resolution to examine the parts I sell.<br>
</blockquote>
<br>
In what way is that use-case blocked by offer-answer? =C2=A0Access to the v=
ideo<br>
needs user confirmation; access to capabilities info shouldn&#39;t.<br>
</blockquote>
<br></div>
I&#39;m reading <a href=3D"http://dev.w3.org/2011/webrtc/editor/webrtc.html=
" target=3D"_blank">http://dev.w3.org/2011/webrtc/<u></u>editor/webrtc.html=
</a><br>
<br>
I cannot see how to get it to generate an SDP offer before I try to open a =
media connection.<br>
<br>
I also cannot see how one could possibly know *what* SDP to offer until you=
 call &quot;getUserMedia&quot;, which prompts the user. (As an example, I h=
ave two cameras attached to this computer over USB. One of them has an on-b=
oard H.264 encoder. The other does not. If my browser can do pass-through o=
f the encoded H.264 but doesn&#39;t have its own H.264 encoder, what offer =
do I generate before the camera is selected? SDP doesn&#39;t have a good wa=
y to encode &quot;maybe&quot;.)<br>


<br>
What we need are a set of Javascript APIs that let us enumerate the availab=
le audio input devices and encoders, video input devices and encoders, audi=
o output devices and decoders, video output devices and decoders. I hate to=
 say it but even H.245 is probably a better model for how to collect the ca=
pabilities than SDP O-A.<div class=3D"im">

<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
And so the<br>
page can generate an offer with audio and video, or just audio if they have=
<br>
no camera. =C2=A0A user declining to send video doesn&#39;t necessarily mea=
n you<br>
don&#39;t negotiate it - they advertise receive-only for video (if the web =
app<br>
wants to).<br>
<br>
</blockquote>
<br></div>
I don&#39;t understand what you mean by &quot;(if the web app wants to)&quo=
t;... the current proposed specification doesn&#39;t have a way for the web=
 app to control whether the SDP offer is receive-only for video or not.<div=
 class=3D"im">

<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
You state above there are major possibilities blocked by offer-answer (I ca=
n<br>
think of some minor issues that with O-A require a second O-A pass) - can y=
ou<br>
detail some use-cases so we can see the benefit and not just the assertion?=
<br>
</blockquote>
<br></div>
It is certainly possible that a repeated set of fake offer-answer exchanges=
 can be used by either Javascript or the server to determine what the capab=
ilities are, and that a set of rich Javascript-exposed capabilities may be =
used to generate SDP offers and answers, so in that sense they are mappable=
 from one to another.<br>


<br>
But I would argue that turning capabilities into SDP is easier than the rev=
erse. This is really a question of whether we are trying to turn the browse=
r into a *platform for building applications* (in which case we should be e=
xposing, as an operating system does, APIs for determining what is possible=
 and APIs for control) or turning the browser into *a phone*, in which case=
 sure, SIP and SDP are both fine choices.<br>

</blockquote><div><br></div><div>I think it makes sense for the browser to =
emit capabilities, which could then be used by the web app to generate a SD=
P offer or answer. This provides the web app with more control over the fun=
ctionality; the browser just reacts to what the web app asks it to do, rath=
er than trying to make decisions (such as how the answer should be generate=
d from the offer) that may not be possible without context.</div>

<div><br></div><div>The original problem that started this email is one spe=
cific example - if the callee application wants to only receive audio, the =
application can generate an audio-only SDP based on the offer, the browser =
capabilities, and the desired app behavior - without any new APIs in the br=
owser.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<br>
I think there&#39;s a whole lot of potential applications beyond what we&#3=
9;re currently thinking of if we provide a platform and not just a phone.<b=
r>
<br>
I&#39;ve already outlined one case (offering the user the ability to place =
a video call *if* there is an acceptable camera and encoder on the system).=
<br>
<br>
Another obvious one is where there&#39;s a dozen people already in a video =
call and one more wishes to join... if the server knows what capabilities e=
xist, the server can tell the new joiner &quot;your browser can&#39;t suppo=
rt video in a format that the other users need&quot; or can tell the existi=
ng user browsers to switch to a different codec that is now compatible with=
 everyone or whatever, rather than having to re-run offer-answer with every=
 party.</blockquote>

<div><br></div><div><br></div><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class=3D=
"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Then we could balance that against the advantage of O-A being well-speced a=
nd<br>
known and implemented.<br>
</blockquote>
<br></div>
But it *isn&#39;t* well-speced for use cases other than one-to-one calling,=
 really. It works poorly for recording, it works poorly for large multi-par=
ty conferences, etc.<br>
<br>
And on top of that, it is missing important attributes that we&#39;ll want =
to control (like whether the Opus codec is being forced to &quot;music&quot=
; mode or not).<div class=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
(And the issues surrounding how well and how<br>
interoperably web-app developers can implement their own capability<br>
negotiations/etc)<br>
<br>
<br>
</blockquote>
<br></div>
For interoperability, the browser (in Javascript) or the server (in any lan=
guage you wish) can generate SDP, if that&#39;s how they wish to interopera=
te.<br>
<br>
Turning capability and control APIs into SDP is exactly the same problem as=
 turning operating system APIs into SDP... the browser should be the new op=
erating system, not just a hardcoded phone.<br><span class=3D"HOEnZb"><font=
 color=3D"#888888">
<br>
Matthew Kaufman</font></span><div class=3D"HOEnZb"><div></div><div class=3D=
"h5"><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>
</div></div></blockquote></div><br>

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SSRCs are the easiest thing to demux on, and everything will be much simpler
if we exchange the SSRCs in signaling. Nothing to get lost, a clear ack from
the other side, and we can perform groupings that simply are not possible
without some form of signaling. RFC 5576 details these cases, such as
grouping of SSRCs for RTX and FEC purposes. We have implementation
experience that this works quite well in point-to-point and multi-user
scenarios.

So I agree with Harald:
- PeerConnection is a "session", with 1 SDP description, describing 1...N
RTP sessions, each containing 1...N RTP sources. It is identified by some
application-specific session-id used in the application-specific signaling
protocol.
- MediaStream is a grouping of 1...N RTP sources with the same CNAME,
intended to be synchronized at playout. It is identified by CNAME, but
additional presentation information could be specified by something like
a=content (type of stream) and a=label (display name)
- MediaTrack is a single RTP source. It is identified by 1...N SSRCs
(typically 1, but could be more if SSRC multiplexing is used for RTX/FEC)

Sample SDP, using RFC 5576, illustrating a SDP for a PeerConnection with 2
MediaStreams (one for audio/video, one for presentation), and 3 MediaTracks.
It uses RTP multiplexing, RTCP multiplexing, and SSRC multiplexing for RTX.
As a result it 1 RTP session, with 2 CNAMEs, and 5 SSRCs.
The only thing I invented here was how to attach a content=  to individual
streams, where I simply added a content: attribute to the a=ssrc lines.

<non-media stuff omitted>
a=group:TOGETHER audio video
m=audio 49168 RTP/AVP 0
a=mid:audio
a=ssrc:1111 cname:ABCD1234 content:speaker
a=rtcp-mux
m=video 49170 RTP/AVP 96 98
a=mid:video
a=rtpmap:96 H264/90000

a=rtpmap:98 rtx/90000
a=fmtp:98 apt=96;rtx-time=3000
a=ssrc-group:FID 2222 2223

a=ssrc:2222 cname:ABCD1234 content:speaker
a=ssrc:2223 cname:ABCD1234 content:speaker


a=ssrc-group:FID 3333 3334

a=ssrc:3333 cname:BCDE2345 content:slides
a=ssrc:3334 cname:BCDE2345 content:slides
a=rtcp-mux

On Thu, Aug 25, 2011 at 10:39 AM, Randell Jesup <randell-ietf@jesup.org>wrote:

> On 8/25/2011 3:50 AM, Elwell, John wrote:
>
>> Randell,
>>
>> Good point - if the first RTCP packet gets dropped, it will be a long time
>> before the next one arrives, and in the meantime no CSRC to SSRC mapping
>> will be available.
>>
>
> Well, we are looking at AVPF which allows (though does not mandate) faster
> RTCP,
> and I think we're suggesting the "fast sync" rfc as well, so that might
> cover it
> - but there still can be a delay before the RTCP gets through successfully.
>
>
>
>  -----Original Message-----
>>> From: public-webrtc-request@w3.org
>>> [mailto:public-webrtc-request@**w3.org <public-webrtc-request@w3.org>]
>>> On Behalf Of Randell Jesup
>>> Sent: 24 August 2011 21:13
>>> To: public-webrtc@w3.org
>>> Subject: Re: Mapping of media streams to RTP (Re: Query: What
>>> does "context" mean in the context (sic) of requirement A15?
>>> [ACTION-6])
>>>
>>> On 8/24/2011 11:47 AM, Elwell, John wrote:
>>>
>>>> RTCP doesn't have a delay if a SR is the first thing sent on
>>>>> a session,
>>>>> as recommended in draft-perkins-avt-rapid-rtp-**sync-03
>>>>>
>>>> section 2.1.1 -
>>>
>>>> this is permitted by RFC 3550 too, it notes.
>>>>>
>>>> [JRE] OK, so CNAME can be used to map an RTP stream to a
>>>>
>>> given media description in SDP. CNAME alone does not indicate
>>> the purpose of an RTP stream - it would still need something
>>> else to map it to "game" or "agent", and that is where
>>> a=content comes in.
>>>
>>> Don't count on RTCP getting received...  Start of a session
>>> might even
>>> be the worst place for it.
>>>
>>>  --
> Randell Jesup
> randell-ietf@jesup.org
>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

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SSRCs are the easiest thing to demux on, and everything will be much simple=
r if we exchange the SSRCs in signaling. Nothing to get lost, a clear ack f=
rom the other side, and we can perform groupings that simply are not possib=
le without some form of signaling.=C2=A0RFC 5576 details these cases, such =
as grouping of SSRCs for RTX and FEC purposes. We have implementation exper=
ience that this works quite well in point-to-point and multi-user scenarios=
.<div>

<div><br></div><div>So I agree with Harald:</div><div><span class=3D"Apple-=
style-span" style=3D"color: rgb(51, 51, 51); font-family: arial, sans-serif=
; font-size: 13px; background-color: rgb(255, 255, 255); ">- PeerConnection=
 is a &quot;session&quot;, with 1 SDP description, describing 1...N RTP ses=
sions, each containing 1...N RTP sources. It is identified by some applicat=
ion-specific session-id used in the application-specific signaling protocol=
.<br>

- MediaStream is a grouping of 1...N RTP sources with the same CNAME, inten=
ded to be synchronized at playout. It is identified by CNAME, but additiona=
l presentation information could be specified by something like a=3Dcontent=
 (type of stream) and a=3Dlabel (display name)<br>

</span><span class=3D"Apple-style-span" style=3D"color: rgb(51, 51, 51); fo=
nt-family: arial, sans-serif; font-size: 13px; background-color: rgb(255, 2=
55, 255); ">- MediaTrack is a single RTP source. It is identified by 1...N =
SSRCs (typically 1, but could be more if SSRC multiplexing is used for RTX/=
FEC)</span></div>

<div><font class=3D"Apple-style-span" color=3D"#333333" face=3D"arial, sans=
-serif"><br></font></div><div><font class=3D"Apple-style-span" color=3D"#33=
3333" face=3D"arial, sans-serif">Sample SDP, using RFC 5576, illustrating a=
 SDP for a PeerConnection with 2 MediaStreams (one for audio/video, one for=
 presentation), and 3 MediaTracks. It uses RTP multiplexing, RTCP multiplex=
ing, and SSRC multiplexing for RTX. As a result it 1 RTP session, with 2 CN=
AMEs, and 5 SSRCs.</font></div>

<div><font class=3D"Apple-style-span" color=3D"#333333" face=3D"arial, sans=
-serif">The only thing I invented here was how to attach a content=3D =C2=
=A0to individual streams, where I simply added a content: attribute to the =
a=3Dssrc lines.</font></div>

<div><font class=3D"Apple-style-span" color=3D"#333333" face=3D"arial, sans=
-serif"><br></font></div><div><font class=3D"Apple-style-span" color=3D"#33=
3333"><span class=3D"Apple-style-span" style=3D"color: rgb(0, 0, 0); "><div=
 style=3D"background-color: transparent; ">

<span id=3D"internal-source-marker_0.12097633280791342" style=3D"font-famil=
y: &#39;Courier New&#39;; color: rgb(0, 0, 0); background-color: transparen=
t; font-style: normal; font-variant: normal; text-decoration: none; vertica=
l-align: baseline; white-space: pre-wrap; ">&lt;non-media stuff omitted&gt;=
</span><br>

<span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); bac=
kground-color: transparent; font-style: normal; font-variant: normal; text-=
decoration: none; vertical-align: baseline; white-space: pre-wrap; ">a=3Dgr=
oup:TOGETHER audio video</span><br>

<span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); bac=
kground-color: transparent; font-style: normal; font-variant: normal; text-=
decoration: none; vertical-align: baseline; white-space: pre-wrap; ">m=3Dau=
dio 49168 RTP/AVP 0</span><br>

<span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); bac=
kground-color: transparent; font-style: normal; font-variant: normal; text-=
decoration: none; vertical-align: baseline; white-space: pre-wrap; ">a=3Dmi=
d:audio<br class=3D"kix-line-break">

a=3Dssrc:1111 cname:ABCD1234 content:speaker</span><br><span style=3D"font-=
family: &#39;Courier New&#39;; color: rgb(0, 0, 0); background-color: trans=
parent; font-style: normal; font-variant: normal; text-decoration: none; ve=
rtical-align: baseline; white-space: pre-wrap; ">a=3Drtcp-mux</span><font c=
lass=3D"Apple-style-span" face=3D"&#39;Courier New&#39;"><span class=3D"App=
le-style-span" style=3D"white-space: pre-wrap;"><br>

</span></font><span style=3D"font-family: &#39;Courier New&#39;; color: rgb=
(0, 0, 0); background-color: transparent; font-style: normal; font-variant:=
 normal; text-decoration: none; vertical-align: baseline; white-space: pre-=
wrap; ">m=3Dvideo 49170 RTP/AVP 96 98</span><br>

<span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); bac=
kground-color: transparent; font-style: normal; font-variant: normal; text-=
decoration: none; vertical-align: baseline; white-space: pre-wrap; ">a=3Dmi=
d:video<br class=3D"kix-line-break">

</span><span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, =
0); background-color: transparent; font-style: normal; font-variant: normal=
; text-decoration: none; vertical-align: baseline; white-space: pre-wrap; "=
>a=3Drtpmap:96 H264/90000</span></div>

</span></font><span class=3D"Apple-style-span" style=3D"font-family: &#39;T=
imes New Roman&#39;; "><pre class=3D"newpage" style=3D"margin-top: 0px; mar=
gin-bottom: 0px; page-break-before: always; ">a=3Drtpmap:98 rtx/90000
a=3Dfmtp:98 apt=3D96;rtx-time=3D3000
a=3Dssrc-group:FID 2222 2223<br></pre></span><font class=3D"Apple-style-spa=
n" color=3D"#333333"><span class=3D"Apple-style-span" style=3D"color: rgb(0=
, 0, 0); "><div style=3D"background-color: transparent; "><span style=3D"fo=
nt-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); background-color: tr=
ansparent; font-style: normal; font-variant: normal; text-decoration: none;=
 vertical-align: baseline; white-space: pre-wrap; ">a=3Dssrc:2222 cname:ABC=
D1234 content:speaker</span></div>

</span></font><span class=3D"Apple-style-span" style=3D"font-family: &#39;C=
ourier New&#39;; white-space: pre-wrap; ">a=3Dssrc:2223 cname:ABCD1234 cont=
ent:speaker  </span></div><div><span class=3D"Apple-style-span" style=3D"fo=
nt-family: &#39;Times New Roman&#39;; "><pre class=3D"newpage" style=3D"mar=
gin-top: 0px; margin-bottom: 0px; page-break-before: always; ">

a=3Dssrc-group:FID 3333 3334</pre></span><font class=3D"Apple-style-span" c=
olor=3D"#333333"><span class=3D"Apple-style-span" style=3D"color: rgb(0, 0,=
 0); "><div style=3D"background-color: transparent; "><span style=3D"font-f=
amily: &#39;Courier New&#39;; color: rgb(0, 0, 0); background-color: transp=
arent; font-style: normal; font-variant: normal; text-decoration: none; ver=
tical-align: baseline; white-space: pre-wrap; ">a=3Dssrc:3333 cname:BCDE234=
5 content:slides</span></div>

</span></font><span class=3D"Apple-style-span" style=3D"font-family: &#39;C=
ourier New&#39;; white-space: pre-wrap; ">a=3Dssrc:3334 cname:BCDE2345 cont=
ent:slides</span><font class=3D"Apple-style-span" color=3D"#333333"><span c=
lass=3D"Apple-style-span" style=3D"color: rgb(0, 0, 0); "><div style=3D"bac=
kground-color: transparent; ">

<span style=3D"font-family: &#39;Courier New&#39;; color: rgb(0, 0, 0); bac=
kground-color: transparent; font-style: normal; font-variant: normal; text-=
decoration: none; vertical-align: baseline; white-space: pre-wrap; ">a=3Drt=
cp-mux</span><font class=3D"Apple-style-span" face=3D"&#39;Courier New&#39;=
"><span class=3D"Apple-style-span" style=3D"white-space: pre-wrap;"><br>

</span></font></div></span></font></div><div><div><div><font class=3D"Apple=
-style-span" color=3D"#333333" face=3D"arial, sans-serif"><br></font><div c=
lass=3D"gmail_quote">On Thu, Aug 25, 2011 at 10:39 AM, Randell Jesup <span =
dir=3D"ltr">&lt;<a href=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesu=
p.org</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"im">On 8/25/2011 3:50 AM, Elw=
ell, John wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Randell,<br>
<br>
Good point - if the first RTCP packet gets dropped, it will be a long time =
before the next one arrives, and in the meantime no CSRC to SSRC mapping wi=
ll be available.<br>
</blockquote>
<br></div>
Well, we are looking at AVPF which allows (though does not mandate) faster =
RTCP,<br>
and I think we&#39;re suggesting the &quot;fast sync&quot; rfc as well, so =
that might cover it<br>
- but there still can be a delay before the RTCP gets through successfully.=
<div class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
-----Original Message-----<br>
From: <a href=3D"mailto:public-webrtc-request@w3.org" target=3D"_blank">pub=
lic-webrtc-request@w3.org</a><br>
[mailto:<a href=3D"mailto:public-webrtc-request@w3.org" target=3D"_blank">p=
ublic-webrtc-request@<u></u>w3.org</a>] On Behalf Of Randell Jesup<br>
Sent: 24 August 2011 21:13<br>
To: <a href=3D"mailto:public-webrtc@w3.org" target=3D"_blank">public-webrtc=
@w3.org</a><br>
Subject: Re: Mapping of media streams to RTP (Re: Query: What<br>
does &quot;context&quot; mean in the context (sic) of requirement A15?<br>
[ACTION-6])<br>
<br>
On 8/24/2011 11:47 AM, Elwell, John wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
RTCP doesn&#39;t have a delay if a SR is the first thing sent on<br>
a session,<br>
as recommended in draft-perkins-avt-rapid-rtp-<u></u>sync-03<br>
</blockquote></blockquote>
section 2.1.1 -<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><blockquote class=3D"gmail_quote" style=3D"m=
argin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
this is permitted by RFC 3550 too, it notes.<br>
</blockquote>
[JRE] OK, so CNAME can be used to map an RTP stream to a<br>
</blockquote>
given media description in SDP. CNAME alone does not indicate<br>
the purpose of an RTP stream - it would still need something<br>
else to map it to &quot;game&quot; or &quot;agent&quot;, and that is where<=
br>
a=3Dcontent comes in.<br>
<br>
Don&#39;t count on RTCP getting received... =C2=A0Start of a session<br>
might even<br>
be the worst place for it.<br>
<br>
</blockquote></blockquote>
-- <br>
Randell Jesup<br>
<a href=3D"mailto:randell-ietf@jesup.org" target=3D"_blank">randell-ietf@je=
sup.org</a><br>
<br></div>
______________________________<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></div></div></div></div>

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Justin Uberti wrote:
> I think it makes sense for the browser to emit capabilities, which could

I agree there's clearly some gaps here that need to be filled.

> then be used by the web app to generate a SDP offer or answer. This

> The original problem that started this email is one specific example -
> if the callee application wants to only receive audio, the application
> can generate an audio-only SDP based on the offer, the browser

I think the Harald's original problem was the other way around: the 
_caller_ wants to only receive audio, and needs to generate an SDP 
_offer_ that says that, even if the browser is capable of receiving 
video. I don't think that invalidates your point, though.

> capabilities, and the desired app behavior - without any new APIs in the
> browser.

But I'm not sure what you mean by "without any new APIs"... in your 
approach, something has to be able to enumerate the capabilities in 
sufficient detail for the webapp to generate SDP by itself. I don't 
think there are any existing APIs that go that far.

You also need an API to tell the browser what to actually do. The 
current PeerConnection approach is passing in the offer or the answer. 
If you're generating the answer, you need some way to tell your browser 
what you answered. For the "please don't send me video" case this is not 
an issue... it'll simply never arrive. If you want to change what the 
local browser is sending out, however, then it is.

I do agree it eliminates the need for an API to tell the browser what 
kind of SDP to generate, but it also seems like it imposes a pretty big 
burden on application developers: even if you keep the 
currently-proposed PeerConnection ability to generate SDP as the "simple 
API", the moment you want to do something slightly more complex, you 
have to add code to generate the appropriate SDP, which necessarily 
involves figuring out all the capabilities of the various browsers on 
various platforms. Maybe I'm naive in thinking that seems like an awful 
lot of work just to say, "Please don't send me video."

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From: Justin Uberti <juberti@google.com>
Date: Fri, 26 Aug 2011 00:48:08 -0400
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Subject: Re: [rtcweb] Additional requirement - audio-only communication
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On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry <
tterriberry@mozilla.com> wrote:

> Justin Uberti wrote:
>
>> I think it makes sense for the browser to emit capabilities, which could
>>
>
> I agree there's clearly some gaps here that need to be filled.
>
>
>  then be used by the web app to generate a SDP offer or answer. This
>>
>
>  The original problem that started this email is one specific example -
>> if the callee application wants to only receive audio, the application
>> can generate an audio-only SDP based on the offer, the browser
>>
>
> I think the Harald's original problem was the other way around: the
> _caller_ wants to only receive audio, and needs to generate an SDP _offer_
> that says that, even if the browser is capable of receiving video. I don't
> think that invalidates your point, though.
>
>
>  capabilities, and the desired app behavior - without any new APIs in the
>> browser.
>>
>
> But I'm not sure what you mean by "without any new APIs"... in your
> approach, something has to be able to enumerate the capabilities in
> sufficient detail for the webapp to generate SDP by itself. I don't think
> there are any existing APIs that go that far.
>

I meant, without specific APIs for that specific use case (i.e. "create an
audio-only offer"). We would need some sort of GetCapabilities API that
returned a blob of all the session description options the browser
supported, which probably could be formatted as a uber SDP offer if that
made parsing simplest.

>
> You also need an API to tell the browser what to actually do. The current
> PeerConnection approach is passing in the offer or the answer. If you're
> generating the answer, you need some way to tell your browser what you
> answered. For the "please don't send me video" case this is not an issue...
> it'll simply never arrive. If you want to change what the local browser is
> sending out, however, then it is.
>

Yes, you need a "HandleLocalDescription" and "HandleRemoteDescription" API,
instead of just a single OnSignalingMessage API. The deviation from the
current flow is fairly minor, you just have 2 additional states in the state
machine.

>
> I do agree it eliminates the need for an API to tell the browser what kind
> of SDP to generate, but it also seems like it imposes a pretty big burden on
> application developers: even if you keep the currently-proposed
> PeerConnection ability to generate SDP as the "simple API", the moment you
> want to do something slightly more complex, you have to add code to generate
> the appropriate SDP, which necessarily involves figuring out all the
> capabilities of the various browsers on various platforms. Maybe I'm naive
> in thinking that seems like an awful lot of work just to say, "Please don't
> send me video."


It's a fair point, there is more complexity involved in generating the
offer, but in our experience it is manageable. I suppose it depends on how
many APIs we decide we need to control the generation of SDP  (i.e. use
cases other than no-video). If we decide we need to control crypto,
resolution preference, codec preference, we may find this approach simpler.

>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailman/listinfo/rtcweb>
>

--90e6ba5bc137aa23fb04ab6141f6
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<br><br><div class=3D"gmail_quote">On Thu, Aug 25, 2011 at 8:01 PM, Timothy=
 B. Terriberry <span dir=3D"ltr">&lt;<a href=3D"mailto:tterriberry@mozilla.=
com">tterriberry@mozilla.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">Justin Uberti wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
I think it makes sense for the browser to emit capabilities, which could<br=
>
</blockquote>
<br></div>
I agree there&#39;s clearly some gaps here that need to be filled.<div clas=
s=3D"im"><br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
then be used by the web app to generate a SDP offer or answer. This<br>
</blockquote>
<br>
</div><div class=3D"im"><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
The original problem that started this email is one specific example -<br>
if the callee application wants to only receive audio, the application<br>
can generate an audio-only SDP based on the offer, the browser<br>
</blockquote>
<br></div>
I think the Harald&#39;s original problem was the other way around: the _ca=
ller_ wants to only receive audio, and needs to generate an SDP _offer_ tha=
t says that, even if the browser is capable of receiving video. I don&#39;t=
 think that invalidates your point, though.<div class=3D"im">

<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
capabilities, and the desired app behavior - without any new APIs in the<br=
>
browser.<br>
</blockquote>
<br></div>
But I&#39;m not sure what you mean by &quot;without any new APIs&quot;... i=
n your approach, something has to be able to enumerate the capabilities in =
sufficient detail for the webapp to generate SDP by itself. I don&#39;t thi=
nk there are any existing APIs that go that far.<br>

</blockquote><div><br></div><div>I meant, without specific APIs for that sp=
ecific use case (i.e. &quot;create an audio-only offer&quot;). We would nee=
d some sort of GetCapabilities API that returned a blob of all the session =
description options the browser supported, which probably could be formatte=
d as a uber SDP offer if that made parsing simplest.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<br>
You also need an API to tell the browser what to actually do. The current P=
eerConnection approach is passing in the offer or the answer. If you&#39;re=
 generating the answer, you need some way to tell your browser what you ans=
wered. For the &quot;please don&#39;t send me video&quot; case this is not =
an issue... it&#39;ll simply never arrive. If you want to change what the l=
ocal browser is sending out, however, then it is.<br>

</blockquote><div><br></div><div>Yes, you need a &quot;HandleLocalDescripti=
on&quot; and &quot;HandleRemoteDescription&quot; API, instead of just a sin=
gle OnSignalingMessage API. The deviation from the current flow is fairly m=
inor, you just have 2 additional states in the state machine.=C2=A0</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<br>
I do agree it eliminates the need for an API to tell the browser what kind =
of SDP to generate, but it also seems like it imposes a pretty big burden o=
n application developers: even if you keep the currently-proposed PeerConne=
ction ability to generate SDP as the &quot;simple API&quot;, the moment you=
 want to do something slightly more complex, you have to add code to genera=
te the appropriate SDP, which necessarily involves figuring out all the cap=
abilities of the various browsers on various platforms. Maybe I&#39;m naive=
 in thinking that seems like an awful lot of work just to say, &quot;Please=
 don&#39;t send me video.&quot;</blockquote>

<div><br></div><div>It&#39;s a fair point, there is more complexity involve=
d in generating the offer, but in our experience it is manageable. I suppos=
e it depends on how many APIs we decide we need to control the generation o=
f SDP =C2=A0(i.e. use cases other than no-video). If we decide we need to c=
ontrol crypto, resolution preference, codec preference, we may find this ap=
proach simpler.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;"><div class=3D"HOEnZb"><div></div><div class=
=3D"h5"><br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

--90e6ba5bc137aa23fb04ab6141f6--

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From: "Robert O'Callahan" <robert@ocallahan.org>
To: Matthew Kaufman <matthew.kaufman@skype.net>
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Subject: Re: [rtcweb] Remote recording - RTC-Web client acting as SIPREC session recording client
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--0015176f085e36d36a04ab619a16
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On Fri, Aug 26, 2011 at 9:19 AM, Matthew Kaufman
<matthew.kaufman@skype.net>wrote:

> On 8/25/2011 11:50 AM, Timothy B. Terriberry wrote:
>
>> Matthew Kaufman wrote:
>>
>>> But this is a bad idea. Providing APIs that let a browser send audio
>>> that is being received from the other end to a third party open several
>>> different cans of worms simultaneously. One is that there's now another
>>>
>>
>> The currently proposed MediaStream Processing API (
>> http://hg.mozilla.org/users/**rocallahan_mozilla.com/specs/**
>> raw-file/tip/StreamProcessing/**StreamProcessing.html<http://hg.mozilla.org/users/rocallahan_mozilla.com/specs/raw-file/tip/StreamProcessing/StreamProcessing.html>)
>> essentially allows exactly this (including mixing). It doesn't make any
>> distinction about whether the source of a MediaStream is a local camera, a
>> <video> tag, or a remote stream from a PeerConnection object. So if you want
>> to prevent this kind of thing, you need to have an active method of doing
>> so, because by default the API will allow it.
>>
>
> I think you need to seriously consider the security implications here. Any
> media that originates from somewhere other than a local camera that has
> given permission or a local microphone that has given permission needs to be
> marked as not sendable elsewhere.


We could do that on top of the above API, e.g. by having a PeerConnection
examine the stream graph to determine whether a MediaStream comes directly
from a local sensor.

However, such a restriction would prevent many useful applications and does
not solve the security problems that arise from a malicious Web app setting
up a connection between you and the PSTN. If that can happen, you're in all
kinds of trouble whether the app can control the contents of your outgoing
stream or not.

Rob
-- 
"If we claim to be without sin, we deceive ourselves and the truth is not in
us. If we confess our sins, he is faithful and just and will forgive us our
sins and purify us from all unrighteousness. If we claim we have not sinned,
we make him out to be a liar and his word is not in us." [1 John 1:8-10]

--0015176f085e36d36a04ab619a16
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

On Fri, Aug 26, 2011 at 9:19 AM, Matthew Kaufman <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a>&gt;<=
/span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"=
>
<div class=3D"im">On 8/25/2011 11:50 AM, Timothy B. Terriberry wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Matthew Kaufman wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
But this is a bad idea. Providing APIs that let a browser send audio<br>
that is being received from the other end to a third party open several<br>
different cans of worms simultaneously. One is that there&#39;s now another=
<br>
</blockquote>
<br>
The currently proposed MediaStream Processing API (<a href=3D"http://hg.moz=
illa.org/users/rocallahan_mozilla.com/specs/raw-file/tip/StreamProcessing/S=
treamProcessing.html" target=3D"_blank">http://hg.mozilla.org/users/<u></u>=
rocallahan_mozilla.com/specs/<u></u>raw-file/tip/StreamProcessing/<u></u>St=
reamProcessing.html</a>) essentially allows exactly this (including mixing)=
. It doesn&#39;t make any distinction about whether the source of a MediaSt=
ream is a local camera, a &lt;video&gt; tag, or a remote stream from a Peer=
Connection object. So if you want to prevent this kind of thing, you need t=
o have an active method of doing so, because by default the API will allow =
it.<br>

</blockquote>
<br></div>
I think you need to seriously consider the security implications here. Any =
media that originates from somewhere other than a local camera that has giv=
en permission or a local microphone that has given permission needs to be m=
arked as not sendable elsewhere.</blockquote>
<div><br>We could do that on top of the above API, e.g. by having a PeerCon=
nection examine the stream graph to determine whether a MediaStream comes d=
irectly from a local sensor.<br><br>However, such a restriction would preve=
nt many useful applications and does not solve the security problems that a=
rise from a malicious Web app setting up a connection between you and the P=
STN. If that can happen, you&#39;re in all kinds of trouble whether the app=
 can control the contents of your outgoing stream or not.<br>
<br>Rob<br></div></div>-- <br>&quot;If we claim to be without sin, we decei=
ve ourselves and the truth is not in us. If we confess our sins, he is fait=
hful and just and will forgive us our sins and purify us from all unrighteo=
usness. If we claim we have not sinned, we make him out to be a liar and hi=
s word is not in us.&quot; [1 John 1:8-10]<br>


--0015176f085e36d36a04ab619a16--

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On Fri, Aug 26, 2011 at 9:18 AM, Matthew Kaufman
<matthew.kaufman@skype.net>wrote:

> Getting ubiquitous deployment will require complying with norms about what
> a browser does and doesn't do. Right now one of the things it *doesn't* do
> is relay media unexpectedly.


Browsers *do* consume arbitrary quantities of downstream bandwidth, and
sometimes upstream bandwidth too. Opera even built BitTorrent into their
browser. Most "browsers don't do X" norms have already fallen by the
wayside. "Browsers don't act as relays" will too.

Given browsers are general-purpose application platforms, it makes more
sense to ban specific Web sites than browsers if you're worried about
excessive bandwidth usage.

Rob
-- 
"If we claim to be without sin, we deceive ourselves and the truth is not in
us. If we confess our sins, he is faithful and just and will forgive us our
sins and purify us from all unrighteousness. If we claim we have not sinned,
we make him out to be a liar and his word is not in us." [1 John 1:8-10]

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On Fri, Aug 26, 2011 at 9:18 AM, Matthew Kaufman <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:matthew.kaufman@skype.net">matthew.kaufman@skype.net</a>&gt;<=
/span> wrote:<br><div class=3D"gmail_quote"><blockquote class=3D"gmail_quot=
e" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"=
>
Getting ubiquitous deployment will require complying with norms about what =
a browser does and doesn&#39;t do. Right now one of the things it *doesn&#3=
9;t* do is relay media unexpectedly.</blockquote><div><br>Browsers *do* con=
sume arbitrary quantities of downstream bandwidth, and sometimes upstream b=
andwidth too. Opera even built BitTorrent into their browser. Most &quot;br=
owsers don&#39;t do X&quot; norms have already fallen by the wayside. &quot=
;Browsers don&#39;t act as relays&quot; will too.<br>
<br>Given browsers are general-purpose application platforms, it makes more=
 sense to ban specific Web sites than browsers if you&#39;re worried about =
excessive bandwidth usage.<br><br>Rob<br></div></div>-- <br>&quot;If we cla=
im to be without sin, we deceive ourselves and the truth is not in us. If w=
e confess our sins, he is faithful and just and will forgive us our sins an=
d purify us from all unrighteousness. If we claim we have not sinned, we ma=
ke him out to be a liar and his word is not in us.&quot; [1 John 1:8-10]<br=
>


--20cf30427260e51c5504ab61c63e--

From stefan.lk.hakansson@ericsson.com  Fri Aug 26 00:13:36 2011
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FWIW,

this is how I would do the case of the caller (initiator) wanting to 
receive only audio, but willing to send audio and video with the current 
API proposal:

1. Generate an audiovisual mediastream (getUserMedia)
2. create PeerConnection, add the above mediastream (addStream)
3. combine the SDP received from PeerConnection with the following 
fields "This is an invitation to a communication", "sending audio and 
video", "want to receive only audio" into a message sent to the peer.
4. the app in the peer (same app!) receives the message, reads the 
fields, (given user agreement to start the communication) creates a 
PeerConnection, feeds the received SDP into the PC, generates an *audio 
only* stream (getUserMedia) and adds it to the PC
5. A couple of SDP exchanges later the application is working as 
intended. (there will be onaddstream's, you will attach those streams to 
video/audio elements etc.)

Quite simple! And no need to have the application read, understand, or 
modify, the SDPs - they are opaque. (Then, of course, the mediastreams 
must be mapped to RTP sessions and such stuff, but I'm sure that is 
solvable and should not be visible in the API IMHO.)

Stefan





on 2011-08-26 06:48, Justin Uberti wrote:
>
>
> On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry
> <tterriberry@mozilla.com <mailto:tterriberry@mozilla.com>> wrote:
>
>     Justin Uberti wrote:
>
>         I think it makes sense for the browser to emit capabilities,
>         which could
>
>
>     I agree there's clearly some gaps here that need to be filled.
>
>
>         then be used by the web app to generate a SDP offer or answer. This
>
>
>         The original problem that started this email is one specific
>         example -
>         if the callee application wants to only receive audio, the
>         application
>         can generate an audio-only SDP based on the offer, the browser
>
>
>     I think the Harald's original problem was the other way around: the
>     _caller_ wants to only receive audio, and needs to generate an SDP
>     _offer_ that says that, even if the browser is capable of receiving
>     video. I don't think that invalidates your point, though.
>
>
>         capabilities, and the desired app behavior - without any new
>         APIs in the
>         browser.
>
>
>     But I'm not sure what you mean by "without any new APIs"... in your
>     approach, something has to be able to enumerate the capabilities in
>     sufficient detail for the webapp to generate SDP by itself. I don't
>     think there are any existing APIs that go that far.
>
>
> I meant, without specific APIs for that specific use case (i.e. "create
> an audio-only offer"). We would need some sort of GetCapabilities API
> that returned a blob of all the session description options the browser
> supported, which probably could be formatted as a uber SDP offer if that
> made parsing simplest.
>
>
>     You also need an API to tell the browser what to actually do. The
>     current PeerConnection approach is passing in the offer or the
>     answer. If you're generating the answer, you need some way to tell
>     your browser what you answered. For the "please don't send me video"
>     case this is not an issue... it'll simply never arrive. If you want
>     to change what the local browser is sending out, however, then it is.
>
>
> Yes, you need a "HandleLocalDescription" and "HandleRemoteDescription"
> API, instead of just a single OnSignalingMessage API. The deviation from
> the current flow is fairly minor, you just have 2 additional states in
> the state machine.
>
>
>     I do agree it eliminates the need for an API to tell the browser
>     what kind of SDP to generate, but it also seems like it imposes a
>     pretty big burden on application developers: even if you keep the
>     currently-proposed PeerConnection ability to generate SDP as the
>     "simple API", the moment you want to do something slightly more
>     complex, you have to add code to generate the appropriate SDP, which
>     necessarily involves figuring out all the capabilities of the
>     various browsers on various platforms. Maybe I'm naive in thinking
>     that seems like an awful lot of work just to say, "Please don't send
>     me video."
>
>
> It's a fair point, there is more complexity involved in generating the
> offer, but in our experience it is manageable. I suppose it depends on
> how many APIs we decide we need to control the generation of SDP  (i.e.
> use cases other than no-video). If we decide we need to control crypto,
> resolution preference, codec preference, we may find this approach simpler.
>
>
>     _________________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/rtcweb
>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>


From john.elwell@siemens-enterprise.com  Fri Aug 26 00:19:35 2011
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From: "Elwell, John" <john.elwell@siemens-enterprise.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>
Date: Fri, 26 Aug 2011 09:20:44 +0200
Thread-Topic: [rtcweb] Remote recording - RTC-Web client acting as	SIPREC session recording client
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=20

> -----Original Message-----
> From: Matthew Kaufman [mailto:matthew.kaufman@skype.net]=20
> Sent: 25 August 2011 18:14
> To: Elwell, John
> Cc: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Remote recording - RTC-Web client=20
> acting as SIPREC session recording client
>=20
> On 8/25/2011 5:51 AM, Elwell, John wrote:
> >
> > New requirements:
> > Fyy1: The browser MUST be able to send in real-time to a=20
> third party media that are being transmitted to and received=20
> from remote participants.
>=20
> This is a bad idea.
>=20
> > Ayy1: The web application MUST be able to ask the browser=20
> to transmit in real-time to a third party media that are=20
> being transmitted to
>=20
> Yes. It should be possible for the browser to send two copies of the=20
> same media to two different places, possibly even with=20
> different encoders.
>=20
> >   and received from remote participants
>=20
> But this is a bad idea. Providing APIs that let a browser send audio=20
> that is being received from the other end to a third party=20
> open several=20
> different cans of worms simultaneously.=20
[JRE] I would assume it would be possible for an application to take previo=
usly received media from a remote endpoint and use that as the source for s=
ending to a third party. Will there be anything to stop the application doi=
ng that? For example, would there be constraints in place to ensure that th=
e source of media sent is an original source (microphone, camera), as oppos=
ed to a file, say? I hope not, because this would prevent me sending a pre-=
recorded announcement to callers, e.g., when I am not present in person.

I don't have a particular preference whether it is the browser or the appli=
cation doing the relaying, provided it can be more or less in real-time.

> One is that there's=20
> now another=20
> path by which a user's media may be sent, possibly without the same=20
> security constraints, without their knowledge. Sure, a B2BUA=20
> can do this=20
> as well, but it makes doing the wrong thing easier.
[JRE] There are regulations in most places that require a user to be notifi=
ed if a conversation is being recorded. I don't believe it is possible to e=
nforce such regulations technically or prevent recording in the absence of =
such notifications.

>=20
> The bigger issue is that this essentially allows you to use a=20
> browser as=20
> a media relay. This will be objected to in corporate (and some=20
> education) environments where there are sound reasons for disallowing=20
> user applications to take advantage of a high-bandwidth connection to=20
> relay media for third parties.=20
[JRE] In such corporate environments, where they want recording, they could=
 do it centrally, thereby forcing all media through a middlebox. Doing it a=
t the endpoint removes the need for that middlebox.

> It also may result in=20
> unexpected resource=20
> consumption on mobile platforms, where the user is paying in=20
> bandwidth=20
> charges and battery life to relay media for a third party.=20
[JRE] Indeed, so in this sort of environment you wouldn't want to use this =
capability, and if recording is required it would need to be done from a mi=
ddlebox.

> And unless=20
> implemented carefully, it can lead to relaying that is very=20
> unexpected=20
> (a banner ad that doesn't ask for permission to use your camera and=20
> microphone but does relay media for a third party while running).
[JRE] Perhaps permission is needed to relay media from a third party, and l=
ikewise to relay information from a local file, etc..

>=20
> Not to mention all the protocol-level implications of being an RTP=20
> mixer, if we're trying to stay true to that particular choice=20
> of protocols.
[JRE] Even if you do mixing (which helps save uplink bandwidth), you are no=
t forced to use the RTP mixer model, you could use the endpoint model. So e=
ach PeerConnection would use RTP/RTCP in the normal manner, without interac=
ting with the other one.

John


John Elwell
Tel: +44 1908 817801 (office and mobile)
Email: john.elwell@siemens-enterprise.com
http://www.siemens-enterprise.com/uk/

Siemens Enterprise Communications Limited.
Registered office: Brickhill Street, Willen Lake, Milton Keynes, MK15 0DJ.
Registered No: 5903714, England.

Siemens Enterprise Communications Limited is a Trademark Licensee of Siemen=
s AG.


>=20
> Matthew Kaufman
> =

From harald@alvestrand.no  Fri Aug 26 04:53:22 2011
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From: Harald Alvestrand <harald@alvestrand.no>
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To: Bernard Aboba <bernard_aboba@hotmail.com>
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Subject: Re: [rtcweb] draft-alvestrand-one-rtp ICE usage (and multi-stream interactions)
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On 08/25/11 21:26, Bernard Aboba wrote:
> The ICE offerer needs to be prepared to receive media on the ports 
> indicated in the offer.  After all, it doesn't know whether the answer 
> will indicate support for TOGETHER or not.  As a result, without any 
> a-priori knowledge, it seems to me that normal behavior with respect 
> to gathering candidates and preparing to receive media should apply.  
> Once an answer is returned that indicates TOGETHER support, then the 
> offerer will know that media won't be received on the ports indicated 
> in the m-lines other than the first.
>
> In general, we'd probably like to conclude the ICE exchange as quickly 
> as possible, so that setting ports to zero in all but the first 
> m-line and then going through another offer-answer exchange is 
> probably non-optimal.
>
> One concern that hasn't been mentioned so far is the interaction of 
> TOGETHER with other multi-stream work now in progress, including CLUE 
> and draft-westerlund-avtcore-multistream-and-simulcast  .  Do we 
> understand how these might interact?

I do not know, but have sent the draft to both MMUSIC and AVTCORE, 
pointing at MMUSIC (in the absence of other advice) as the place for 
further discussion of the draft.

Magnus has been on vacation for a while, so I don't think he's seen it yet.

>
>
> Paul Kyzivat said:
>
> IIUC, there are some aspects of ICE that MUST be done before sending 
> the offer. Specifically that involved gathering the candidate list, 
> which may involve assignment of multiple local ports, interactions 
> with a TURN server, etc. I don't think everything can be deferred 
> until the answer is received.
> (I suppose one approach would be for the initial offer to set the port 
> to zero in all but the first m-line. Then, once the first answer is 
> received, indicating whether TOGETHER is supported, another offer can 
> be generated, using appropriate ICE based on whether using TOGETHER or 
> not. But that does start to get complex.)
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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    On 08/25/11 21:26, Bernard Aboba wrote:
    <blockquote cite="mid:BLU152-W2E8C02FB7B9370169BA0193100@phx.gbl"
      type="cite">
      <style><!--
.hmmessage P
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      <div dir="ltr">
        The ICE offerer needs to be prepared to receive media on the
        ports indicated in the offer.&nbsp; After all, it doesn't know
        whether the answer will indicate support for TOGETHER or
        not.&nbsp;&nbsp;As a result, without any a-priori knowledge, it seems to
        me that normal behavior with respect to gathering candidates and
        preparing to receive media should apply.&nbsp; Once an answer is
        returned that indicates TOGETHER support, then the offerer will
        know that media won't be received on the ports indicated in the
        m-lines other than the first. <br>
        &nbsp;<br>
        In general, we'd probably like to conclude the ICE exchange as
        quickly as possible, so that setting ports to zero in all but
        the first m-line&nbsp;and then going through another offer-answer
        exchange is probably non-optimal. &nbsp;&nbsp;<br>
        &nbsp;<br>
        One concern that hasn't been mentioned so far is the interaction
        of TOGETHER with other multi-stream work now in progress,
        including CLUE
        and&nbsp;draft-westerlund-avtcore-multistream-and-simulcast&nbsp; .&nbsp; Do we
        understand how these might&nbsp;interact? <br>
      </div>
    </blockquote>
    <br>
    I do not know, but have sent the draft to both MMUSIC and AVTCORE,
    pointing at MMUSIC (in the absence of other advice) as the place for
    further discussion of the draft.<br>
    <br>
    Magnus has been on vacation for a while, so I don't think he's seen
    it yet.<br>
    <br>
    <blockquote cite="mid:BLU152-W2E8C02FB7B9370169BA0193100@phx.gbl"
      type="cite">
      <div dir="ltr">&nbsp;<br>
        &nbsp;<br>
        Paul Kyzivat said:<br>
        &nbsp;<br>
        <font face="Courier New">IIUC, there are some aspects of ICE
          that MUST be done before sending the <tt>offer. Specifically
            that involved gathering the candidate list, which </tt><tt>may
            involve assignment of multiple local ports, interactions
            with a TURN </tt><tt>server, etc. I don't think everything
            can be deferred until the answer </tt><tt>is received. </tt></font><br>
        <tt>(I suppose one approach would be for the initial offer to
          set the port </tt><tt>to zero in all but the first m-line.
          Then, once the first answer is </tt><tt>received, indicating
          whether TOGETHER is supported, another offer can be </tt><tt>generated,
          using appropriate ICE based on whether using TOGETHER or not.
        </tt><tt>But that does start to get complex.) </tt><br>
      </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>

--------------050801040509070209040703--

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References: <4E539A1F.109@alvestrand.no> <4E53B80C.20304@ericsson.com> <4E53E0C8.6010304@alvestrand.no> <BBF498F2D030E84AB1179E24D1AC41D61C1BCA7F62@ESESSCMS0362.eemea.ericsson.se> <4E54ADC7.8030407@jesup.org> <4E54CC05.1040705@alvestrand.no> <4E54CE29.2060605@ericsson.com> <4E54D867.4060706@alvestrand.no> <4E54D9C2.6000205@ericsson.com> <4E54DE8E.9080207@alvestrand.no> <4E54DFCF.4000805@ericsson.com> <4E54E24F.7060906@alvestrand.no> <4E56707B.104@skype.net> <4E567737.6020101@jesup.org> <4E5680B0.6070702@skype.net> <CAOJ7v-0DnVKYD2gd4os3R-LuzN1mu4qZqjndDEJcJXwY7U-FnA@mail.gmail.com> <4E56E264.8000600@mozilla.com> <CAOJ7v-2kEByX3mq7dRFuo7wEqaEGz597aWuFpEZK9zE_eS6U4A@mail.gmail.com> <4E5747E7.2050605@ericsson.com>
From: Justin Uberti <juberti@google.com>
Date: Fri, 26 Aug 2011 10:11:31 -0400
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On Fri, Aug 26, 2011 at 3:14 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> FWIW,
>
> this is how I would do the case of the caller (initiator) wanting to
> receive only audio, but willing to send audio and video with the current =
API
> proposal:
>
> 1. Generate an audiovisual mediastream (getUserMedia)
> 2. create PeerConnection, add the above mediastream (addStream)
> 3. combine the SDP received from PeerConnection with the following fields
> "This is an invitation to a communication", "sending audio and video", "w=
ant
> to receive only audio" into a message sent to the peer.
> 4. the app in the peer (same app!) receives the message, reads the fields=
,
> (given user agreement to start the communication) creates a PeerConnectio=
n,
> feeds the received SDP into the PC, generates an *audio only* stream
> (getUserMedia) and adds it to the PC
> 5. A couple of SDP exchanges later the application is working as intended=
.
> (there will be onaddstream's, you will attach those streams to video/audi=
o
> elements etc.)
>
> Quite simple! And no need to have the application read, understand, or
> modify, the SDPs - they are opaque. (Then, of course, the mediastreams mu=
st
> be mapped to RTP sessions and such stuff, but I'm sure that is solvable a=
nd
> should not be visible in the API IMHO.)
>

Maybe I misunderstand, but it sounds like with these fields you are definin=
g
a new signaling protocol, which I think we want to avoid.

Going back to your step 3 - here the initiator can of course munge the
generated SDP and discard the m=3Dvideo section, which should work, but in
other cases this munging may cause things to get out of sync since the
initiator browser's understanding of the offer now differs from reality.
What we really want is some way to either a) an API to tell the initiator
browser to produce an offer with no m=3Dvideo, or b) a way for the applicat=
ion
to customize or generate the offer and feed it back to the browser. b) is
definitely more complex, but a) could end up causing us to add a lot of
knobs to the API.

>
> Stefan
>
>
>
>
>
>
> on 2011-08-26 06:48, Justin Uberti wrote:
>
>>
>>
>> On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry
>> <tterriberry@mozilla.com <mailto:tterriberry@mozilla.**com<tterriberry@m=
ozilla.com>>>
>> wrote:
>>
>>    Justin Uberti wrote:
>>
>>        I think it makes sense for the browser to emit capabilities,
>>        which could
>>
>>
>>    I agree there's clearly some gaps here that need to be filled.
>>
>>
>>        then be used by the web app to generate a SDP offer or answer. Th=
is
>>
>>
>>        The original problem that started this email is one specific
>>        example -
>>        if the callee application wants to only receive audio, the
>>        application
>>        can generate an audio-only SDP based on the offer, the browser
>>
>>
>>    I think the Harald's original problem was the other way around: the
>>    _caller_ wants to only receive audio, and needs to generate an SDP
>>    _offer_ that says that, even if the browser is capable of receiving
>>    video. I don't think that invalidates your point, though.
>>
>>
>>        capabilities, and the desired app behavior - without any new
>>        APIs in the
>>        browser.
>>
>>
>>    But I'm not sure what you mean by "without any new APIs"... in your
>>    approach, something has to be able to enumerate the capabilities in
>>    sufficient detail for the webapp to generate SDP by itself. I don't
>>    think there are any existing APIs that go that far.
>>
>>
>> I meant, without specific APIs for that specific use case (i.e. "create
>> an audio-only offer"). We would need some sort of GetCapabilities API
>> that returned a blob of all the session description options the browser
>> supported, which probably could be formatted as a uber SDP offer if that
>> made parsing simplest.
>>
>>
>>    You also need an API to tell the browser what to actually do. The
>>    current PeerConnection approach is passing in the offer or the
>>    answer. If you're generating the answer, you need some way to tell
>>    your browser what you answered. For the "please don't send me video"
>>    case this is not an issue... it'll simply never arrive. If you want
>>    to change what the local browser is sending out, however, then it is.
>>
>>
>> Yes, you need a "HandleLocalDescription" and "HandleRemoteDescription"
>> API, instead of just a single OnSignalingMessage API. The deviation from
>> the current flow is fairly minor, you just have 2 additional states in
>> the state machine.
>>
>>
>>    I do agree it eliminates the need for an API to tell the browser
>>    what kind of SDP to generate, but it also seems like it imposes a
>>    pretty big burden on application developers: even if you keep the
>>    currently-proposed PeerConnection ability to generate SDP as the
>>    "simple API", the moment you want to do something slightly more
>>    complex, you have to add code to generate the appropriate SDP, which
>>    necessarily involves figuring out all the capabilities of the
>>    various browsers on various platforms. Maybe I'm naive in thinking
>>    that seems like an awful lot of work just to say, "Please don't send
>>    me video."
>>
>>
>> It's a fair point, there is more complexity involved in generating the
>> offer, but in our experience it is manageable. I suppose it depends on
>> how many APIs we decide we need to control the generation of SDP  (i.e.
>> use cases other than no-video). If we decide we need to control crypto,
>> resolution preference, codec preference, we may find this approach
>> simpler.
>>
>>
>>    ______________________________**___________________
>>    rtcweb mailing list
>>    rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>
>>    https://www.ietf.org/mailman/_**_listinfo/rtcweb<https://www.ietf.org=
/mailman/__listinfo/rtcweb>
>>    <https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/=
mailman/listinfo/rtcweb>
>> >
>>
>>
>>
> ______________________________**_________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/mailm=
an/listinfo/rtcweb>
>

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<br><br><div class=3D"gmail_quote">On Fri, Aug 26, 2011 at 3:14 AM, Stefan =
H=C3=A5kansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakanss=
on@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br>=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">

FWIW,<br>
<br>
this is how I would do the case of the caller (initiator) wanting to receiv=
e only audio, but willing to send audio and video with the current API prop=
osal:<br>
<br>
1. Generate an audiovisual mediastream (getUserMedia)<br>
2. create PeerConnection, add the above mediastream (addStream)<br>
3. combine the SDP received from PeerConnection with the following fields &=
quot;This is an invitation to a communication&quot;, &quot;sending audio an=
d video&quot;, &quot;want to receive only audio&quot; into a message sent t=
o the peer.<br>


4. the app in the peer (same app!) receives the message, reads the fields, =
(given user agreement to start the communication) creates a PeerConnection,=
 feeds the received SDP into the PC, generates an *audio only* stream (getU=
serMedia) and adds it to the PC<br>


5. A couple of SDP exchanges later the application is working as intended. =
(there will be onaddstream&#39;s, you will attach those streams to video/au=
dio elements etc.)<br>
<br>
Quite simple! And no need to have the application read, understand, or modi=
fy, the SDPs - they are opaque. (Then, of course, the mediastreams must be =
mapped to RTP sessions and such stuff, but I&#39;m sure that is solvable an=
d should not be visible in the API IMHO.)<br>

</blockquote><div><br></div><div>Maybe I misunderstand, but it sounds like =
with these fields you are defining a new signaling protocol, which I think =
we want to avoid.</div><div><br></div><div>Going back to your step 3 - here=
 the initiator can of course munge the generated SDP and discard the m=3Dvi=
deo section, which should work, but in other cases this munging may cause t=
hings to get out of sync since the initiator browser&#39;s understanding of=
 the offer now differs from reality. What we really want is some way to eit=
her a) an API to tell the initiator browser to produce an offer with no m=
=3Dvideo, or b) a way for the application to customize or generate the offe=
r and feed it back to the browser. b) is definitely more complex, but a) co=
uld end up causing us to add a lot of knobs to the API.</div>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex;">
<br>
Stefan<div class=3D"im"><br>
<br>
<br>
<br>
<br>
<br>
on 2011-08-26 06:48, Justin Uberti wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
<br>
<br>
On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry<br></div><div><div><=
/div><div class=3D"h5">
&lt;<a href=3D"mailto:tterriberry@mozilla.com" target=3D"_blank">tterriberr=
y@mozilla.com</a> &lt;mailto:<a href=3D"mailto:tterriberry@mozilla.com" tar=
get=3D"_blank">tterriberry@mozilla.<u></u>com</a>&gt;&gt; wrote:<br>
<br>
 =C2=A0 =C2=A0Justin Uberti wrote:<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0I think it makes sense for the browser to emit =
capabilities,<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0which could<br>
<br>
<br>
 =C2=A0 =C2=A0I agree there&#39;s clearly some gaps here that need to be fi=
lled.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0then be used by the web app to generate a SDP o=
ffer or answer. This<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0The original problem that started this email is=
 one specific<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0example -<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0if the callee application wants to only receive=
 audio, the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0application<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0can generate an audio-only SDP based on the off=
er, the browser<br>
<br>
<br>
 =C2=A0 =C2=A0I think the Harald&#39;s original problem was the other way a=
round: the<br>
 =C2=A0 =C2=A0_caller_ wants to only receive audio, and needs to generate a=
n SDP<br>
 =C2=A0 =C2=A0_offer_ that says that, even if the browser is capable of rec=
eiving<br>
 =C2=A0 =C2=A0video. I don&#39;t think that invalidates your point, though.=
<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0capabilities, and the desired app behavior - wi=
thout any new<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0APIs in the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0browser.<br>
<br>
<br>
 =C2=A0 =C2=A0But I&#39;m not sure what you mean by &quot;without any new A=
PIs&quot;... in your<br>
 =C2=A0 =C2=A0approach, something has to be able to enumerate the capabilit=
ies in<br>
 =C2=A0 =C2=A0sufficient detail for the webapp to generate SDP by itself. I=
 don&#39;t<br>
 =C2=A0 =C2=A0think there are any existing APIs that go that far.<br>
<br>
<br>
I meant, without specific APIs for that specific use case (i.e. &quot;creat=
e<br>
an audio-only offer&quot;). We would need some sort of GetCapabilities API<=
br>
that returned a blob of all the session description options the browser<br>
supported, which probably could be formatted as a uber SDP offer if that<br=
>
made parsing simplest.<br>
<br>
<br>
 =C2=A0 =C2=A0You also need an API to tell the browser what to actually do.=
 The<br>
 =C2=A0 =C2=A0current PeerConnection approach is passing in the offer or th=
e<br>
 =C2=A0 =C2=A0answer. If you&#39;re generating the answer, you need some wa=
y to tell<br>
 =C2=A0 =C2=A0your browser what you answered. For the &quot;please don&#39;=
t send me video&quot;<br>
 =C2=A0 =C2=A0case this is not an issue... it&#39;ll simply never arrive. I=
f you want<br>
 =C2=A0 =C2=A0to change what the local browser is sending out, however, the=
n it is.<br>
<br>
<br>
Yes, you need a &quot;HandleLocalDescription&quot; and &quot;HandleRemoteDe=
scription&quot;<br>
API, instead of just a single OnSignalingMessage API. The deviation from<br=
>
the current flow is fairly minor, you just have 2 additional states in<br>
the state machine.<br>
<br>
<br>
 =C2=A0 =C2=A0I do agree it eliminates the need for an API to tell the brow=
ser<br>
 =C2=A0 =C2=A0what kind of SDP to generate, but it also seems like it impos=
es a<br>
 =C2=A0 =C2=A0pretty big burden on application developers: even if you keep=
 the<br>
 =C2=A0 =C2=A0currently-proposed PeerConnection ability to generate SDP as =
the<br>
 =C2=A0 =C2=A0&quot;simple API&quot;, the moment you want to do something s=
lightly more<br>
 =C2=A0 =C2=A0complex, you have to add code to generate the appropriate SDP=
, which<br>
 =C2=A0 =C2=A0necessarily involves figuring out all the capabilities of the=
<br>
 =C2=A0 =C2=A0various browsers on various platforms. Maybe I&#39;m naive in=
 thinking<br>
 =C2=A0 =C2=A0that seems like an awful lot of work just to say, &quot;Pleas=
e don&#39;t send<br>
 =C2=A0 =C2=A0me video.&quot;<br>
<br>
<br>
It&#39;s a fair point, there is more complexity involved in generating the<=
br>
offer, but in our experience it is manageable. I suppose it depends on<br>
how many APIs we decide we need to control the generation of SDP =C2=A0(i.e=
.<br>
use cases other than no-video). If we decide we need to control crypto,<br>
resolution preference, codec preference, we may find this approach simpler.=
<br>
<br>
<br>
 =C2=A0 =C2=A0______________________________<u></u>___________________<br>
 =C2=A0 =C2=A0rtcweb mailing list<br></div></div>
 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@i=
etf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank"=
>rtcweb@ietf.org</a>&gt;<div class=3D"im"><br>
 =C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/__listinfo/rtcweb" ta=
rget=3D"_blank">https://www.ietf.org/mailman/_<u></u>_listinfo/rtcweb</a><b=
r>
 =C2=A0 =C2=A0&lt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a>&g=
t;<br>
<br>
<br>
</div></blockquote><div class=3D"HOEnZb"><div></div><div class=3D"h5">
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br>

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From stefan.lk.hakansson@ericsson.com  Fri Aug 26 07:22:12 2011
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To: Justin Uberti <juberti@google.com>
References: <4E539A1F.109@alvestrand.no> <4E53B80C.20304@ericsson.com> <4E53E0C8.6010304@alvestrand.no> <BBF498F2D030E84AB1179E24D1AC41D61C1BCA7F62@ESESSCMS0362.eemea.ericsson.se> <4E54ADC7.8030407@jesup.org> <4E54CC05.1040705@alvestrand.no> <4E54CE29.2060605@ericsson.com> <4E54D867.4060706@alvestrand.no> <4E54D9C2.6000205@ericsson.com> <4E54DE8E.9080207@alvestrand.no> <4E54DFCF.4000805@ericsson.com> <4E54E24F.7060906@alvestrand.no> <4E56707B.104@skype.net> <4E567737.6020101@jesup.org> <4E5680B0.6070702@skype.net> <CAOJ7v-0DnVKYD2gd4os3R-LuzN1mu4qZqjndDEJcJXwY7U-FnA@mail.gmail.com> <4E56E264.8000600@mozilla.com> <CAOJ7v-2kEByX3mq7dRFuo7wEqaEGz597aWuFpEZK9zE_eS6U4A@mail.gmail.com> <4E5747E7.2050605@ericsson.com> <CAOJ7v-1ccLPZhqCDW0ngFkm23TeDSLjNCMUJj-k7wwdg+tTnhg@mail.gmail.com>
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Subject: Re: [rtcweb] Additional requirement - audio-only communication
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On 2011-08-26 16:11, Justin Uberti wrote:
>
>
> On Fri, Aug 26, 2011 at 3:14 AM, Stefan HÃ¥kansson LK
> <stefan.lk.hakansson@ericsson.com
> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>
>     FWIW,
>
>     this is how I would do the case of the caller (initiator) wanting to
>     receive only audio, but willing to send audio and video with the
>     current API proposal:
>
>     1. Generate an audiovisual mediastream (getUserMedia)
>     2. create PeerConnection, add the above mediastream (addStream)
>     3. combine the SDP received from PeerConnection with the following
>     fields "This is an invitation to a communication", "sending audio
>     and video", "want to receive only audio" into a message sent to the
>     peer.
>     4. the app in the peer (same app!) receives the message, reads the
>     fields, (given user agreement to start the communication) creates a
>     PeerConnection, feeds the received SDP into the PC, generates an
>     *audio only* stream (getUserMedia) and adds it to the PC
>     5. A couple of SDP exchanges later the application is working as
>     intended. (there will be onaddstream's, you will attach those
>     streams to video/audio elements etc.)
>
>     Quite simple! And no need to have the application read, understand,
>     or modify, the SDPs - they are opaque. (Then, of course, the
>     mediastreams must be mapped to RTP sessions and such stuff, but I'm
>     sure that is solvable and should not be visible in the API IMHO.)
>
>
> Maybe I misunderstand, but it sounds like with these fields you are
> defining a new signaling protocol, which I think we want to avoid.

That's not how I see it. It is the same web app in both peers, and they 
can communicate inbetween them in any way the app developer see fit 
(using e.g. XHR via the web server). There is already a need for a 
mechanism to pass the SDPs between the browsers, and that mechanism 
could be used to pass other data.

So no new protocol needed!


>
> Going back to your step 3 - here the initiator can of course munge the
> generated SDP and discard the m=video section, which should work, but in
> other cases this munging may cause things to get out of sync since the
> initiator browser's understanding of the offer now differs from reality.

I would really like to avoid that the web app has to read and modify the 
SDP except for in exceptional cases.

In the current API there is also no need (at least in this case) since 
all streams are set up unidirectional - all that is needed is that the 
application in the browser of the callee gets to know it should add an 
audio-only stream (and it can be told that as I outlined above).

> What we really want is some way to either a) an API to tell the
> initiator browser to produce an offer with no m=video, or b) a way for
> the application to customize or generate the offer and feed it back to
> the browser. b) is definitely more complex, but a) could end up causing
> us to add a lot of knobs to the API.
>
>
>     Stefan
>
>
>
>
>
>
>     on 2011-08-26 06:48, Justin Uberti wrote:
>
>
>
>         On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry
>         <tterriberry@mozilla.com <mailto:tterriberry@mozilla.com>
>         <mailto:tterriberry@mozilla.__com
>         <mailto:tterriberry@mozilla.com>>> wrote:
>
>             Justin Uberti wrote:
>
>                 I think it makes sense for the browser to emit capabilities,
>                 which could
>
>
>             I agree there's clearly some gaps here that need to be filled.
>
>
>                 then be used by the web app to generate a SDP offer or
>         answer. This
>
>
>                 The original problem that started this email is one specific
>                 example -
>                 if the callee application wants to only receive audio, the
>                 application
>                 can generate an audio-only SDP based on the offer, the
>         browser
>
>
>             I think the Harald's original problem was the other way
>         around: the
>             _caller_ wants to only receive audio, and needs to generate
>         an SDP
>             _offer_ that says that, even if the browser is capable of
>         receiving
>             video. I don't think that invalidates your point, though.
>
>
>                 capabilities, and the desired app behavior - without any new
>                 APIs in the
>                 browser.
>
>
>             But I'm not sure what you mean by "without any new APIs"...
>         in your
>             approach, something has to be able to enumerate the
>         capabilities in
>             sufficient detail for the webapp to generate SDP by itself.
>         I don't
>             think there are any existing APIs that go that far.
>
>
>         I meant, without specific APIs for that specific use case (i.e.
>         "create
>         an audio-only offer"). We would need some sort of
>         GetCapabilities API
>         that returned a blob of all the session description options the
>         browser
>         supported, which probably could be formatted as a uber SDP offer
>         if that
>         made parsing simplest.
>
>
>             You also need an API to tell the browser what to actually
>         do. The
>             current PeerConnection approach is passing in the offer or the
>             answer. If you're generating the answer, you need some way
>         to tell
>             your browser what you answered. For the "please don't send
>         me video"
>             case this is not an issue... it'll simply never arrive. If
>         you want
>             to change what the local browser is sending out, however,
>         then it is.
>
>
>         Yes, you need a "HandleLocalDescription" and
>         "HandleRemoteDescription"
>         API, instead of just a single OnSignalingMessage API. The
>         deviation from
>         the current flow is fairly minor, you just have 2 additional
>         states in
>         the state machine.
>
>
>             I do agree it eliminates the need for an API to tell the browser
>             what kind of SDP to generate, but it also seems like it
>         imposes a
>             pretty big burden on application developers: even if you
>         keep the
>             currently-proposed PeerConnection ability to generate SDP as the
>         "simple API", the moment you want to do something slightly more
>             complex, you have to add code to generate the appropriate
>         SDP, which
>             necessarily involves figuring out all the capabilities of the
>             various browsers on various platforms. Maybe I'm naive in
>         thinking
>             that seems like an awful lot of work just to say, "Please
>         don't send
>             me video."
>
>
>         It's a fair point, there is more complexity involved in
>         generating the
>         offer, but in our experience it is manageable. I suppose it
>         depends on
>         how many APIs we decide we need to control the generation of SDP
>           (i.e.
>         use cases other than no-video). If we decide we need to control
>         crypto,
>         resolution preference, codec preference, we may find this
>         approach simpler.
>
>
>             ___________________________________________________
>             rtcweb mailing list
>         rtcweb@ietf.org <mailto:rtcweb@ietf.org> <mailto:rtcweb@ietf.org
>         <mailto:rtcweb@ietf.org>>
>
>         https://www.ietf.org/mailman/____listinfo/rtcweb
>         <https://www.ietf.org/mailman/__listinfo/rtcweb>
>         <https://www.ietf.org/mailman/__listinfo/rtcweb
>         <https://www.ietf.org/mailman/listinfo/rtcweb>>
>
>
>
>     _________________________________________________
>     rtcweb mailing list
>     rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/__listinfo/rtcweb
>     <https://www.ietf.org/mailman/listinfo/rtcweb>
>
>


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References: <4E539A1F.109@alvestrand.no> <4E53B80C.20304@ericsson.com> <4E53E0C8.6010304@alvestrand.no> <BBF498F2D030E84AB1179E24D1AC41D61C1BCA7F62@ESESSCMS0362.eemea.ericsson.se> <4E54ADC7.8030407@jesup.org> <4E54CC05.1040705@alvestrand.no> <4E54CE29.2060605@ericsson.com> <4E54D867.4060706@alvestrand.no> <4E54D9C2.6000205@ericsson.com> <4E54DE8E.9080207@alvestrand.no> <4E54DFCF.4000805@ericsson.com> <4E54E24F.7060906@alvestrand.no> <4E56707B.104@skype.net> <4E567737.6020101@jesup.org> <4E5680B0.6070702@skype.net> <CAOJ7v-0DnVKYD2gd4os3R-LuzN1mu4qZqjndDEJcJXwY7U-FnA@mail.gmail.com> <4E56E264.8000600@mozilla.com> <CAOJ7v-2kEByX3mq7dRFuo7wEqaEGz597aWuFpEZK9zE_eS6U4A@mail.gmail.com> <4E5747E7.2050605@ericsson.com> <CAOJ7v-1ccLPZhqCDW0ngFkm23TeDSLjNCMUJj-k7wwdg+tTnhg@mail.gmail.com> <4E57AC5C.1020406@ericsson.com>
From: Justin Uberti <juberti@google.com>
Date: Fri, 26 Aug 2011 10:56:26 -0400
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Additional requirement - audio-only communication
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On Fri, Aug 26, 2011 at 10:23 AM, Stefan H=C3=A5kansson LK <
stefan.lk.hakansson@ericsson.com> wrote:

> On 2011-08-26 16:11, Justin Uberti wrote:
>
>>
>>
>> On Fri, Aug 26, 2011 at 3:14 AM, Stefan H=C3=A5kansson LK
>> <stefan.lk.hakansson@ericsson.**com <stefan.lk.hakansson@ericsson.com>
>> <mailto:stefan.lk.hakansson@**ericsson.com<stefan.lk.hakansson@ericsson.=
com>>>
>> wrote:
>>
>>    FWIW,
>>
>>    this is how I would do the case of the caller (initiator) wanting to
>>    receive only audio, but willing to send audio and video with the
>>    current API proposal:
>>
>>    1. Generate an audiovisual mediastream (getUserMedia)
>>    2. create PeerConnection, add the above mediastream (addStream)
>>    3. combine the SDP received from PeerConnection with the following
>>    fields "This is an invitation to a communication", "sending audio
>>    and video", "want to receive only audio" into a message sent to the
>>    peer.
>>    4. the app in the peer (same app!) receives the message, reads the
>>    fields, (given user agreement to start the communication) creates a
>>    PeerConnection, feeds the received SDP into the PC, generates an
>>    *audio only* stream (getUserMedia) and adds it to the PC
>>    5. A couple of SDP exchanges later the application is working as
>>    intended. (there will be onaddstream's, you will attach those
>>    streams to video/audio elements etc.)
>>
>>    Quite simple! And no need to have the application read, understand,
>>    or modify, the SDPs - they are opaque. (Then, of course, the
>>    mediastreams must be mapped to RTP sessions and such stuff, but I'm
>>    sure that is solvable and should not be visible in the API IMHO.)
>>
>>
>> Maybe I misunderstand, but it sounds like with these fields you are
>> defining a new signaling protocol, which I think we want to avoid.
>>
>
> That's not how I see it. It is the same web app in both peers, and they c=
an
> communicate inbetween them in any way the app developer see fit (using e.=
g.
> XHR via the web server). There is already a need for a mechanism to pass =
the
> SDPs between the browsers, and that mechanism could be used to pass other
> data.
>
> So no new protocol needed!


How will this work if it is not the same web app in both peers, i.e. an
interop scenario? Clearly if it is the same app we can pass along whatever
meta-info we want to instruct the remote side on what to do.


>
>
>
>> Going back to your step 3 - here the initiator can of course munge the
>> generated SDP and discard the m=3Dvideo section, which should work, but =
in
>> other cases this munging may cause things to get out of sync since the
>> initiator browser's understanding of the offer now differs from reality.
>>
>
> I would really like to avoid that the web app has to read and modify the
> SDP except for in exceptional cases.
>
> In the current API there is also no need (at least in this case) since al=
l
> streams are set up unidirectional - all that is needed is that the
> application in the browser of the callee gets to know it should add an
> audio-only stream (and it can be told that as I outlined above).
>
>  What we really want is some way to either a) an API to tell the
>> initiator browser to produce an offer with no m=3Dvideo, or b) a way for
>> the application to customize or generate the offer and feed it back to
>> the browser. b) is definitely more complex, but a) could end up causing
>> us to add a lot of knobs to the API.
>>
>>
>>    Stefan
>>
>>
>>
>>
>>
>>
>>    on 2011-08-26 06:48, Justin Uberti wrote:
>>
>>
>>
>>        On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Terriberry
>>        <tterriberry@mozilla.com <mailto:tterriberry@mozilla.**com<tterri=
berry@mozilla.com>
>> >
>>        <mailto:tterriberry@mozilla.__**com
>>
>>        <mailto:tterriberry@mozilla.**com <tterriberry@mozilla.com>>>>
>> wrote:
>>
>>            Justin Uberti wrote:
>>
>>                I think it makes sense for the browser to emit
>> capabilities,
>>                which could
>>
>>
>>            I agree there's clearly some gaps here that need to be filled=
.
>>
>>
>>                then be used by the web app to generate a SDP offer or
>>        answer. This
>>
>>
>>                The original problem that started this email is one
>> specific
>>                example -
>>                if the callee application wants to only receive audio, th=
e
>>                application
>>                can generate an audio-only SDP based on the offer, the
>>        browser
>>
>>
>>            I think the Harald's original problem was the other way
>>        around: the
>>            _caller_ wants to only receive audio, and needs to generate
>>        an SDP
>>            _offer_ that says that, even if the browser is capable of
>>        receiving
>>            video. I don't think that invalidates your point, though.
>>
>>
>>                capabilities, and the desired app behavior - without any
>> new
>>                APIs in the
>>                browser.
>>
>>
>>            But I'm not sure what you mean by "without any new APIs"...
>>        in your
>>            approach, something has to be able to enumerate the
>>        capabilities in
>>            sufficient detail for the webapp to generate SDP by itself.
>>        I don't
>>            think there are any existing APIs that go that far.
>>
>>
>>        I meant, without specific APIs for that specific use case (i.e.
>>        "create
>>        an audio-only offer"). We would need some sort of
>>        GetCapabilities API
>>        that returned a blob of all the session description options the
>>        browser
>>        supported, which probably could be formatted as a uber SDP offer
>>        if that
>>        made parsing simplest.
>>
>>
>>            You also need an API to tell the browser what to actually
>>        do. The
>>            current PeerConnection approach is passing in the offer or th=
e
>>            answer. If you're generating the answer, you need some way
>>        to tell
>>            your browser what you answered. For the "please don't send
>>        me video"
>>            case this is not an issue... it'll simply never arrive. If
>>        you want
>>            to change what the local browser is sending out, however,
>>        then it is.
>>
>>
>>        Yes, you need a "HandleLocalDescription" and
>>        "HandleRemoteDescription"
>>        API, instead of just a single OnSignalingMessage API. The
>>        deviation from
>>        the current flow is fairly minor, you just have 2 additional
>>        states in
>>        the state machine.
>>
>>
>>            I do agree it eliminates the need for an API to tell the
>> browser
>>            what kind of SDP to generate, but it also seems like it
>>        imposes a
>>            pretty big burden on application developers: even if you
>>        keep the
>>            currently-proposed PeerConnection ability to generate SDP as
>> the
>>        "simple API", the moment you want to do something slightly more
>>            complex, you have to add code to generate the appropriate
>>        SDP, which
>>            necessarily involves figuring out all the capabilities of the
>>            various browsers on various platforms. Maybe I'm naive in
>>        thinking
>>            that seems like an awful lot of work just to say, "Please
>>        don't send
>>            me video."
>>
>>
>>        It's a fair point, there is more complexity involved in
>>        generating the
>>        offer, but in our experience it is manageable. I suppose it
>>        depends on
>>        how many APIs we decide we need to control the generation of SDP
>>          (i.e.
>>        use cases other than no-video). If we decide we need to control
>>        crypto,
>>        resolution preference, codec preference, we may find this
>>        approach simpler.
>>
>>
>>            ______________________________**_____________________
>>            rtcweb mailing list
>>        rtcweb@ietf.org <mailto:rtcweb@ietf.org> <mailto:rtcweb@ietf.org
>>
>>        <mailto:rtcweb@ietf.org>>
>>
>>        https://www.ietf.org/mailman/_**___listinfo/rtcweb<https://www.ie=
tf.org/mailman/____listinfo/rtcweb>
>>        <https://www.ietf.org/mailman/**__listinfo/rtcweb<https://www.iet=
f.org/mailman/__listinfo/rtcweb>
>> >
>>        <https://www.ietf.org/mailman/**__listinfo/rtcweb<https://www.iet=
f.org/mailman/__listinfo/rtcweb>
>>        <https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.=
org/mailman/listinfo/rtcweb>
>> >>
>>
>>
>>
>>    ______________________________**___________________
>>    rtcweb mailing list
>>    rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>>    https://www.ietf.org/mailman/_**_listinfo/rtcweb<https://www.ietf.org=
/mailman/__listinfo/rtcweb>
>>    <https://www.ietf.org/mailman/**listinfo/rtcweb<https://www.ietf.org/=
mailman/listinfo/rtcweb>
>> >
>>
>>
>>
>

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<br><br><div class=3D"gmail_quote">On Fri, Aug 26, 2011 at 10:23 AM, Stefan=
 H=C3=A5kansson LK <span dir=3D"ltr">&lt;<a href=3D"mailto:stefan.lk.hakans=
son@ericsson.com">stefan.lk.hakansson@ericsson.com</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex;">

<div class=3D"im">On 2011-08-26 16:11, Justin Uberti wrote:<br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"im">
<br>
<br>
On Fri, Aug 26, 2011 at 3:14 AM, Stefan H=C3=A5kansson LK<br>
&lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_blank">s=
tefan.lk.hakansson@ericsson.<u></u>com</a><br></div><div class=3D"im">
&lt;mailto:<a href=3D"mailto:stefan.lk.hakansson@ericsson.com" target=3D"_b=
lank">stefan.lk.hakansson@<u></u>ericsson.com</a>&gt;&gt; wrote:<br>
<br>
 =C2=A0 =C2=A0FWIW,<br>
<br>
 =C2=A0 =C2=A0this is how I would do the case of the caller (initiator) wan=
ting to<br>
 =C2=A0 =C2=A0receive only audio, but willing to send audio and video with =
the<br>
 =C2=A0 =C2=A0current API proposal:<br>
<br>
 =C2=A0 =C2=A01. Generate an audiovisual mediastream (getUserMedia)<br>
 =C2=A0 =C2=A02. create PeerConnection, add the above mediastream (addStrea=
m)<br>
 =C2=A0 =C2=A03. combine the SDP received from PeerConnection with the foll=
owing<br>
 =C2=A0 =C2=A0fields &quot;This is an invitation to a communication&quot;, =
&quot;sending audio<br>
 =C2=A0 =C2=A0and video&quot;, &quot;want to receive only audio&quot; into =
a message sent to the<br>
 =C2=A0 =C2=A0peer.<br>
 =C2=A0 =C2=A04. the app in the peer (same app!) receives the message, read=
s the<br>
 =C2=A0 =C2=A0fields, (given user agreement to start the communication) cre=
ates a<br>
 =C2=A0 =C2=A0PeerConnection, feeds the received SDP into the PC, generates=
 an<br>
 =C2=A0 =C2=A0*audio only* stream (getUserMedia) and adds it to the PC<br>
 =C2=A0 =C2=A05. A couple of SDP exchanges later the application is working=
 as<br>
 =C2=A0 =C2=A0intended. (there will be onaddstream&#39;s, you will attach t=
hose<br>
 =C2=A0 =C2=A0streams to video/audio elements etc.)<br>
<br>
 =C2=A0 =C2=A0Quite simple! And no need to have the application read, under=
stand,<br>
 =C2=A0 =C2=A0or modify, the SDPs - they are opaque. (Then, of course, the<=
br>
 =C2=A0 =C2=A0mediastreams must be mapped to RTP sessions and such stuff, b=
ut I&#39;m<br>
 =C2=A0 =C2=A0sure that is solvable and should not be visible in the API IM=
HO.)<br>
<br>
<br>
Maybe I misunderstand, but it sounds like with these fields you are<br>
defining a new signaling protocol, which I think we want to avoid.<br>
</div></blockquote>
<br>
That&#39;s not how I see it. It is the same web app in both peers, and they=
 can communicate inbetween them in any way the app developer see fit (using=
 e.g. XHR via the web server). There is already a need for a mechanism to p=
ass the SDPs between the browsers, and that mechanism could be used to pass=
 other data.<br>


<br>
So no new protocol needed!</blockquote><div><br></div><div>How will this wo=
rk if it is not the same web app in both peers, i.e. an interop scenario? C=
learly if it is the same app we can pass along whatever meta-info we want t=
o instruct the remote side on what to do.</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 class=3D"im"><br>
<br>
<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<br>
Going back to your step 3 - here the initiator can of course munge the<br>
generated SDP and discard the m=3Dvideo section, which should work, but in<=
br>
other cases this munging may cause things to get out of sync since the<br>
initiator browser&#39;s understanding of the offer now differs from reality=
.<br>
</blockquote>
<br></div>
I would really like to avoid that the web app has to read and modify the SD=
P except for in exceptional cases.<br>
<br>
In the current API there is also no need (at least in this case) since all =
streams are set up unidirectional - all that is needed is that the applicat=
ion in the browser of the callee gets to know it should add an audio-only s=
tream (and it can be told that as I outlined above).<br>


<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"im">
What we really want is some way to either a) an API to tell the<br>
initiator browser to produce an offer with no m=3Dvideo, or b) a way for<br=
>
the application to customize or generate the offer and feed it back to<br>
the browser. b) is definitely more complex, but a) could end up causing<br>
us to add a lot of knobs to the API.<br>
<br>
<br>
 =C2=A0 =C2=A0Stefan<br>
<br>
<br>
<br>
<br>
<br>
<br>
 =C2=A0 =C2=A0on 2011-08-26 06:48, Justin Uberti wrote:<br>
<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0On Thu, Aug 25, 2011 at 8:01 PM, Timothy B. Ter=
riberry<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;<a href=3D"mailto:tterriberry@mozilla.com" =
target=3D"_blank">tterriberry@mozilla.com</a> &lt;mailto:<a href=3D"mailto:=
tterriberry@mozilla.com" target=3D"_blank">tterriberry@mozilla.<u></u>com</=
a>&gt;<br></div>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;mailto:<a href=3D"mailto:tterriberry@mozill=
a." target=3D"_blank">tterriberry@mozilla.</a>__<u></u>com<div><div></div><=
div class=3D"h5"><br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;mailto:<a href=3D"mailto:tterriberry@mozill=
a.com" target=3D"_blank">tterriberry@mozilla.<u></u>com</a>&gt;&gt;&gt; wro=
te:<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0Justin Uberti wrote:<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0I think it makes se=
nse for the browser to emit capabilities,<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0which could<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0I agree there&#39;s clearly some =
gaps here that need to be filled.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0then be used by the=
 web app to generate a SDP offer or<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0answer. This<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0The original proble=
m that started this email is one specific<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0example -<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0if the callee appli=
cation wants to only receive audio, the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0application<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0can generate an aud=
io-only SDP based on the offer, the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0browser<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0I think the Harald&#39;s original=
 problem was the other way<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0around: the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0_caller_ wants to only receive au=
dio, and needs to generate<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0an SDP<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0_offer_ that says that, even if t=
he browser is capable of<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0receiving<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0video. I don&#39;t think that inv=
alidates your point, though.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0capabilities, and t=
he desired app behavior - without any new<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0APIs in the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0browser.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0But I&#39;m not sure what you mea=
n by &quot;without any new APIs&quot;...<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0in your<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0approach, something has to be abl=
e to enumerate the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0capabilities in<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0sufficient detail for the webapp =
to generate SDP by itself.<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0I don&#39;t<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0think there are any existing APIs=
 that go that far.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0I meant, without specific APIs for that specifi=
c use case (i.e.<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;create<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0an audio-only offer&quot;). We would need some =
sort of<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0GetCapabilities API<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0that returned a blob of all the session descrip=
tion options the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0browser<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0supported, which probably could be formatted as=
 a uber SDP offer<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0if that<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0made parsing simplest.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0You also need an API to tell the =
browser what to actually<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0do. The<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0current PeerConnection approach i=
s passing in the offer or the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0answer. If you&#39;re generating =
the answer, you need some way<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0to tell<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0your browser what you answered. F=
or the &quot;please don&#39;t send<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0me video&quot;<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0case this is not an issue... it&#=
39;ll simply never arrive. If<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0you want<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0to change what the local browser =
is sending out, however,<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0then it is.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0Yes, you need a &quot;HandleLocalDescription&qu=
ot; and<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;HandleRemoteDescription&quot;<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0API, instead of just a single OnSignalingMessag=
e API. The<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0deviation from<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0the current flow is fairly minor, you just have=
 2 additional<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0states in<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0the state machine.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0I do agree it eliminates the need=
 for an API to tell the browser<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0what kind of SDP to generate, but=
 it also seems like it<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0imposes a<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0pretty big burden on application =
developers: even if you<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0keep the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0currently-proposed PeerConnection=
 ability to generate SDP as the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&quot;simple API&quot;, the moment you want to =
do something slightly more<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0complex, you have to add code to =
generate the appropriate<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0SDP, which<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0necessarily involves figuring out=
 all the capabilities of the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0various browsers on various platf=
orms. Maybe I&#39;m naive in<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0thinking<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0that seems like an awful lot of w=
ork just to say, &quot;Please<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0don&#39;t send<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0me video.&quot;<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0It&#39;s a fair point, there is more complexity=
 involved in<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0generating the<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0offer, but in our experience it is manageable. =
I suppose it<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0depends on<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0how many APIs we decide we need to control the =
generation of SDP<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0(i.e.<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0use cases other than no-video). If we decide we=
 need to control<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0crypto,<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0resolution preference, codec preference, we may=
 find this<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0approach simpler.<br>
<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0______________________________<u>=
</u>_____________________<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0rtcweb mailing list<br></div></di=
v>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf.org" target=3D"_b=
lank">rtcweb@ietf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" tar=
get=3D"_blank">rtcweb@ietf.org</a>&gt; &lt;mailto:<a href=3D"mailto:rtcweb@=
ietf.org" target=3D"_blank">rtcweb@ietf.org</a><div class=3D"im">

<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" t=
arget=3D"_blank">rtcweb@ietf.org</a>&gt;&gt;<br>
<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/____lis=
tinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/_<u></u>___lis=
tinfo/rtcweb</a><br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;<a href=3D"https://www.ietf.org/mailman/__l=
istinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/<u></u>__lis=
tinfo/rtcweb</a>&gt;<br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;<a href=3D"https://www.ietf.org/mailman/__l=
istinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/<u></u>__lis=
tinfo/rtcweb</a><br>
 =C2=A0 =C2=A0 =C2=A0 =C2=A0&lt;<a href=3D"https://www.ietf.org/mailman/lis=
tinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinf=
o/rtcweb</a>&gt;&gt;<br>
<br>
<br>
<br>
 =C2=A0 =C2=A0______________________________<u></u>___________________<br>
 =C2=A0 =C2=A0rtcweb mailing list<br>
 =C2=A0 =C2=A0<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@i=
etf.org</a> &lt;mailto:<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank"=
>rtcweb@ietf.org</a>&gt;<br>
 =C2=A0 =C2=A0<a href=3D"https://www.ietf.org/mailman/__listinfo/rtcweb" ta=
rget=3D"_blank">https://www.ietf.org/mailman/_<u></u>_listinfo/rtcweb</a><b=
r>
 =C2=A0 =C2=A0&lt;<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
target=3D"_blank">https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a>&g=
t;<br>
<br>
<br>
</div></blockquote>
<br>
</blockquote></div><br>

--90e6ba5bc13714fe4604ab69c1ea--

From stefan.lk.hakansson@ericsson.com  Fri Aug 26 08:03:57 2011
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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Justin Uberti <juberti@google.com>
Date: Fri, 26 Aug 2011 17:05:11 +0200
Thread-Topic: [rtcweb] Additional requirement - audio-only communication
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References: <4E539A1F.109@alvestrand.no> <4E53B80C.20304@ericsson.com> <4E53E0C8.6010304@alvestrand.no> <BBF498F2D030E84AB1179E24D1AC41D61C1BCA7F62@ESESSCMS0362.eemea.ericsson.se> <4E54ADC7.8030407@jesup.org> <4E54CC05.1040705@alvestrand.no> <4E54CE29.2060605@ericsson.com> <4E54D867.4060706@alvestrand.no> <4E54D9C2.6000205@ericsson.com> <4E54DE8E.9080207@alvestrand.no> <4E54DFCF.4000805@ericsson.com> <4E54E24F.7060906@alvestrand.no> <4E56707B.104@skype.net> <4E567737.6020101@jesup.org> <4E5680B0.6070702@skype.net> <CAOJ7v-0DnVKYD2gd4os3R-LuzN1mu4qZqjndDEJcJXwY7U-FnA@mail.gmail.com> <4E56E264.8000600@mozilla.com> <CAOJ7v-2kEByX3mq7dRFuo7wEqaEGz597aWuFpEZK9zE_eS6U4A@mail.gmail.com> <4E5747E7.2050605@ericsson.com> <CAOJ7v-1ccLPZhqCDW0ngFkm23TeDSLjNCMUJj-k7wwdg+tTnhg@mail.gmail.com> <4E57AC5C.1020406@ericsson.com>, <CAOJ7v-1r5stCamNxgMZg9M0sNqB_aKz6T9H2JgDjQwYXBuEFgQ@mail.gmail.com>
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On 2011-08-26, Justin Uberti wrote:

>How will this work if it is not the same web app in both peers, i.e. an in=
terop scenario? Clearly if it is the same app we can pass along whatever me=
ta-info we want to instruct the remote side on what to do.
I agree that this specific case would not work without defining some kind p=
rotocol. Then, on the other hand, do all 'advanced' features have to be sup=
ported also in interop? Perhaps "you get what you send" or "you get audio a=
nd video (unless the user at the sending side opts out of one component)" i=
s OK in interop cases. I guess we need to discuss this.

Br,
Stefan

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--_3a3c0eba-0b45-4a71-a920-8e57d8fc3aea_
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[BA] Why can't the initiating application delete the m=3Dvideo line from th=
e SDP blobthat the signaling callback receives=2C before sending it to the =
callee?   Are we assumingthat the SDP blob is opaque and can't be edited? =
=20
  Stefan said: "What we really want is some way to either a) an API to tell=
 the
initiator browser to produce an offer with no m=3Dvideo=2C or b) a way for
the application to customize or generate the offer and feed it back to
the browser. b) is definitely more complex=2C but a) could end up causing
us to add a lot of knobs to the API.


   Stefan
"
  		 	   		  =

--_3a3c0eba-0b45-4a71-a920-8e57d8fc3aea_
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<html>
<head>
<style><!--
.hmmessage P
{
margin:0px=3B
padding:0px
}
body.hmmessage
{
font-size: 10pt=3B
font-family:Tahoma
}
--></style>
</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
[BA] Why can't the initiating application delete the m=3Dvideo line from th=
e SDP blob<BR>that the signaling callback receives=2C before sending it to =
the callee?&nbsp=3B&nbsp=3B Are we assuming<BR>that the SDP blob is opaque =
and can't be edited?&nbsp=3B <br><BR>&nbsp=3B<BR>&nbsp=3BStefan said:<BR>&n=
bsp=3B<BR>"What we really want is some way to either a) an API to tell the<=
br>initiator browser to produce an offer with no m=3Dvideo=2C or b) a way f=
or<br>the application to customize or generate the offer and feed it back t=
o<br>the browser. b) is definitely more complex=2C but a) could end up caus=
ing<br>us to add a lot of knobs to the API.<br><br><br>   Stefan<br>"<BR><d=
iv><br>&nbsp=3B</div> 		 	   		  </div></body>
</html>=

--_3a3c0eba-0b45-4a71-a920-8e57d8fc3aea_--

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References: <4E539A1F.109@alvestrand.no>	<4E53B80C.20304@ericsson.com>, <4E53E0C8.6010304@alvestrand.no>, <BBF498F2D030E84AB1179E24D1AC41D61C1BCA7F62@ESESSCMS0362.eemea.ericsson.se>, <4E54ADC7.8030407@jesup.org>	<4E54CC05.1040705@alvestrand.no>, <4E54CE29.2060605@ericsson.com>	<4E54D867.4060706@alvestrand.no>, <4E54D9C2.6000205@ericsson.com>	<4E54DE8E.9080207@alvestrand.no>, <4E54DFCF.4000805@ericsson.com>	<4E54E24F.7060906@alvestrand.no>, <4E56707B.104@skype.net>	<4E567737.6020101@jesup.org>, <4E5680B0.6070702@skype.net>, <CAOJ7v-0DnVKYD2gd4os3R-LuzN1mu4qZqjndDEJcJXwY7U-FnA@mail.gmail.com>, <4E56E264.8000600@mozilla.com>, <CAOJ7v-2kEByX3mq7dRFuo7wEqaEGz597aWuFpEZK9zE_eS6U4A@mail.gmail.com>, <4E5747E7.2050605@ericsson.com>, <CAOJ7v-1ccLPZhqCDW0ngFkm23TeDSLjNCMUJj-k7wwdg+tTnhg@mail.gmail.com>, <4E57AC5C.1020406@ericsson.com>, <CAOJ7v-1r5stCamNxgMZg9M0sNqB_aKz6T9H2JgDjQwYXBuEFgQ@mail.gmail.com> <BLU152-W4528C66BBF7B1C9A76E1BA93130@phx.gbl>
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Subject: Re: [rtcweb] Additional requirement - audio-only communication
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One less complex path forward is the question "is this audio meant for 
this geolocation" rather than "is this audio meant for this user". In 
the ubiquitous environment is the implication of user recognition by 
gestures and the assumption of relative vehicle location (that carries 
the audio out) to specific geolocation.

Instead of 'the phone' in the browser, tabs with geolocation data may 
clarify "mobile tabs". For example, two different web-browsers, user 
drags tab from one browser to the other, and it "just works". If the 
web-browsers works like an O/S, then this should be simple based on 
"offers" that originate at the paravirtual level. If each web-browser 
virtualizes such ability then you already know (a) and tabs as the 
semantic vehicle.

On another note, I wonder why the concept is so hard to open tab X and 
tab Y and connect the two related sites together through those tabs if I 
set X as server and Y as client for each other. Of course, some keep the 
motto that the web-browser does not enter web-server mode because 
corporate firewalls are more acceptable for them.

Based on capabilities available ("tethering included"), which endpoint 
is client and which endpoint is server SHOULD BE negotiated in the 
connection phase. The example above taken one step further, the users 
drags the mobile X tab already in server mode (for Y client-tab) to 
another web-browser... still complex? I think that clearly reveals 
legacy systems.

On 08/26/2011 10:06 AM, Bernard Aboba wrote:
> [BA] Why can't the initiating application delete the m=video line from 
> the SDP blob
> that the signaling callback receives, before sending it to the 
> callee?   Are we assuming
> that the SDP blob is opaque and can't be edited?
>
>
>  Stefan said:
>
> "What we really want is some way to either a) an API to tell the
> initiator browser to produce an offer with no m=video, or b) a way for
> the application to customize or generate the offer and feed it back to
> the browser. b) is definitely more complex, but a) could end up causing
> us to add a lot of knobs to the API.
>
>
> Stefan
> "
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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Subject: [rtcweb] IW10 Considered Harmful
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I just submitted draft-gettys-iw10-considered-harmful-00

http://www.ietf.org/id/draft-gettys-iw10-considered-harmful-00.txt

It is commenting on the proposal in
http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/
which would change the TCP initial window from 4 packets to 10.

I believe this would be a serious mistake compounding problems already
present
for low latency applications, for the reasons outlined in my draft.
                    Best regards,
                                Jim Gettys



                        Abstract

The proposed change to the initial window to 10 indraft-ietf-tcpm-
initcwnd must be considered deeply harmful; not because it is the
proposed change is evil taken in isolation, but that other changes in
web browsers and web sites that have occurred over the last decade,
it makes the problem of transient congestion at a user's broadband
connection two and a half times worse. This result has been hidden
by the already widespread bufferbloat present in broadband
connections. Packet loss in isolation is no longer a useful metric
of a path's quality. The very drive to improve latency of web page
rendering is already destroying other low latency applications, such
as VOIP and gaming, and will prevent reliable rich web real time web
applications such as those contemplated by the IETF rtcweb working
group.

--------------030706020008030602050405
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<html>
  <head>

    <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
  </head>
  <body bgcolor="#FFFFFF" text="#000000">
    I just submitted draft-gettys-iw10-considered-harmful-00<span
      class="Apple-style-span" style="color: rgb(0, 0, 0); font-family:
      'Times New Roman'; font-style: normal; font-variant: normal;
      font-weight: normal; letter-spacing: normal; line-height: normal;
      orphans: 2; text-align: -webkit-auto; text-indent: 0px;
      text-transform: none; white-space: normal; widows: 2;
      word-spacing: 0px; -webkit-text-decorations-in-effect: none;
      -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px;
      font-size: medium; "></span><br>
    <br>
    <a
href="http://www.ietf.org/id/draft-gettys-iw10-considered-harmful-00.txt">http://www.ietf.org/id/draft-gettys-iw10-considered-harmful-00.txt</a><br>
    <br>
    It is commenting on the proposal in<br>
    <a href="http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/">http://datatracker.ietf.org/doc/draft-ietf-tcpm-initcwnd/</a><br>
    which would change the TCP initial window from 4 packets to 10.<br>
    <br>
    I believe this would be a serious mistake compounding problems
    already present<br>
    for low latency applications, for the reasons outlined in my draft.<br>
    &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; Best regards,<br>
    &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; Jim Gettys<br>
    <br>
    <br>
    <br>
    &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; Abstract<br>
    <br>
    The proposed change to the initial window to 10 indraft-ietf-tcpm-<br>
    initcwnd must be considered deeply harmful; not because it is the<br>
    proposed change is evil taken in isolation, but that other changes
    in<br>
    web browsers and web sites that have occurred over the last decade,<br>
    it makes the problem of transient congestion at a user's broadband<br>
    connection two and a half times worse. This result has been hidden<br>
    by the already widespread bufferbloat present in broadband<br>
    connections. Packet loss in isolation is no longer a useful metric<br>
    of a path's quality. The very drive to improve latency of web page<br>
    rendering is already destroying other low latency applications, such<br>
    as VOIP and gaming, and will prevent reliable rich web real time web<br>
    applications such as those contemplated by the IETF rtcweb working<br>
    group.
  </body>
</html>

--------------030706020008030602050405--

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On 8/26/2011 10:23 AM, Stefan HÃ¥kansson LK wrote:
> On 2011-08-26 16:11, Justin Uberti wrote:
>>
>>
>> On Fri, Aug 26, 2011 at 3:14 AM, Stefan HÃ¥kansson LK
>> <stefan.lk.hakansson@ericsson.com
>> <mailto:stefan.lk.hakansson@ericsson.com>> wrote:
>>
>>     FWIW,
>>
>>     this is how I would do the case of the caller (initiator) wanting to
>>     receive only audio, but willing to send audio and video with the
>>     current API proposal:
>>
>>     1. Generate an audiovisual mediastream (getUserMedia)
>>     2. create PeerConnection, add the above mediastream (addStream)
>>     3. combine the SDP received from PeerConnection with the following
>>     fields "This is an invitation to a communication", "sending audio
>>     and video", "want to receive only audio" into a message sent to the
>>     peer.
>>     4. the app in the peer (same app!) receives the message, reads the
>>     fields, (given user agreement to start the communication) creates a
>>     PeerConnection, feeds the received SDP into the PC, generates an
>>     *audio only* stream (getUserMedia) and adds it to the PC
>>     5. A couple of SDP exchanges later the application is working as
>>     intended. (there will be onaddstream's, you will attach those
>>     streams to video/audio elements etc.)
>>
>>     Quite simple! And no need to have the application read, understand,
>>     or modify, the SDPs - they are opaque. (Then, of course, the
>>     mediastreams must be mapped to RTP sessions and such stuff, but I'm
>>     sure that is solvable and should not be visible in the API IMHO.)
>>
>>
>> Maybe I misunderstand, but it sounds like with these fields you are
>> defining a new signaling protocol, which I think we want to avoid.
>
> That's not how I see it. It is the same web app in both peers, and 
> they can communicate inbetween them in any way the app developer see 
> fit (using e.g. XHR via the web server). There is already a need for a 
> mechanism to pass the SDPs between the browsers, and that mechanism 
> could be used to pass other data.
>
> So no new protocol needed!

This is a new (application-specific) protocol.  It may be a simple one, 
but it is one, overlaid on top of the SDP saying "ignore this part of 
the SDP" effectively.  As mentioned, you then have more problems when 
two different domains/apps want to talk to each other.  We've left the 
protocol between the domains unspecified here, but it will likely need 
to be translated to some external standard such as SIP, H.323 (ugh) or 
Jingle/XMPP.


-- 
Randell Jesup
randell-ietf@jesup.org


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From: Justin Uberti <juberti@google.com>
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It can, and in this case it would probably work fine. The browser will still
instantiate the machinery needs to receive video, since it doesn't know the
m=video line was removed, but assuming the remote side functions correctly,
this should not result in a noticeable problem.

However, other adjustments to the SDP will not be so benign, such as
changing the a=crypto lines, or payload ids, although they could be if the
offerer was informed of the updated description.

On Fri, Aug 26, 2011 at 1:06 PM, Bernard Aboba <bernard_aboba@hotmail.com>wrote:

>  [BA] Why can't the initiating application delete the m=video line from the
> SDP blob
> that the signaling callback receives, before sending it to the callee?
> Are we assuming
> that the SDP blob is opaque and can't be edited?
>
>
>  Stefan said:
>
> "What we really want is some way to either a) an API to tell the
> initiator browser to produce an offer with no m=video, or b) a way for
> the application to customize or generate the offer and feed it back to
> the browser. b) is definitely more complex, but a) could end up causing
> us to add a lot of knobs to the API.
>
>
> Stefan
> "
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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It can, and in this case it would probably work fine. The browser will stil=
l instantiate the machinery needs to receive video, since it doesn&#39;t kn=
ow the m=3Dvideo line was removed, but assuming the remote side functions c=
orrectly, this should not result in a noticeable problem.<div>

<br></div><div>However, other adjustments to the SDP will not be so benign,=
 such as changing the a=3Dcrypto lines, or payload ids, although they could=
 be if the offerer was informed of the updated description.<br><br><div cla=
ss=3D"gmail_quote">

On Fri, Aug 26, 2011 at 1:06 PM, Bernard Aboba <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.com</a>&gt;</s=
pan> 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">
[BA] Why can&#39;t the initiating application delete the m=3Dvideo line fro=
m the SDP blob<br>that the signaling callback receives, before sending it t=
o the callee?=C2=A0=C2=A0 Are we assuming<br>that the SDP blob is opaque an=
d can&#39;t be edited?=C2=A0 <br>

<div class=3D"im"><br>=C2=A0<br>=C2=A0Stefan said:<br>=C2=A0<br>&quot;What =
we really want is some way to either a) an API to tell the<br>initiator bro=
wser to produce an offer with no m=3Dvideo, or b) a way for<br>the applicat=
ion to customize or generate the offer and feed it back to<br>

the browser. b) is definitely more complex, but a) could end up causing<br>=
us to add a lot of knobs to the API.<br><br><br>   Stefan<br>&quot;<br><div=
><br>=C2=A0</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></div>

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Subject: [rtcweb] Draft agenda RTCWeb call on September 8th
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We need to start working on an agenda for the Sept 8th - bellow is  very =
rough straw man. Please suggest ideas for things to add, delete, change =
...


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


Use Case - John (60 min)=20

    Identity in multiuser calls

    Recording=20


Terminology Mapping RTP to W3C ( 30 min)=20
    - what's a track, session, flow, connection and how do they relate=20=



Just One Port - (AKA Multiplexing) -  Harald  (30 min)


Non Media Data Transport (30 min)=20


Security and User Interface (90 min)=20
     What do we expect the see and do in relation to security and =
authorization around access to microphone/camera


Other things that I don't see as being ready to spend time on ...


NAT=20



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From: =?iso-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>
To: Randell Jesup <randell-ietf@jesup.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Sat, 27 Aug 2011 06:50:36 +0200
Thread-Topic: [rtcweb] Additional requirement - audio-only communication
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Randell wrote:
>Stefan wrote
>>Justing wrote
>>>
>>> Maybe I misunderstand, but it sounds like with these fields you are
>>> defining a new signaling protocol, which I think we want to avoid.
>>
>> That's not how I see it. It is the same web app in both peers, and
>> they can communicate inbetween them in any way the app developer see
>> fit (using e.g. XHR via the web server). There is already a need for a
>> mechanism to pass the SDPs between the browsers, and that mechanism
>> could be used to pass other data.
>>
>> So no new protocol needed!
>
>This is a new (application-specific) protocol.=20

At least it is an application specific way to exchange information.

> It may be a simple one,
>but it is one, overlaid on top of the SDP saying "ignore this part of
>the SDP" effectively. =20

I don't think it says anything along those lines. What the extra info says =
is what A expects B to set up. The SDP o/a initiated by A (caller) deals _o=
nly_ with the stream(s) flowing from A to B, it contains no info about what=
 B should send to A (and then there is no need to say anything like "ignore=
 part of SDP").

>As mentioned, you then have more problems when
>two different domains/apps want to talk to each other. =20

Yes, that is right. Another problem Harald pointed out in another thread wa=
s the mapping to RTP sessions.

For the record, I'm not trying to say that the current API proposal is the =
final one that should not be changed. I think discussions will show that th=
ere will be a need to make adjustments and enhancements to it. I'm just try=
ing to show how you could/would solve things with the current API proposal =
(as there seem to be some misconceptions).

Stefan

From bernard_aboba@hotmail.com  Sat Aug 27 08:53:51 2011
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[BA] Agreed that there seems to be some misconceptions about what the curre=
nt API proposal can do.=20

To clear this up=2C would it make sense to have a (brief) presentation on t=
he API at the interim?

 		 	   		  =

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[BA] Agreed that there seems to be some misconceptions about what the curre=
nt API proposal can do. <br><br>To clear this up=2C would it make sense to =
have a (brief) presentation on the API at the interim?<br><br> 		 	   		  <=
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From stewe@stewe.org  Sat Aug 27 09:18:01 2011
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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: [rtcweb] Additional requirement - audio-only communication
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Or perhaps pointers on this list to the newest version every now and then
(whenever a major update has happened)?
Specifically, is Harald's draft from March 24 still the one to look at?
(The companion lists have quite a bit of traffic, and there are only so many
cycles per day to monitor those, especially if API design is not in your
core interest or competency).
Thanks,
Stephan


From:  Bernard Aboba <bernard_aboba@hotmail.com>
Date:  Sat, 27 Aug 2011 08:55:06 -0700
To:  <stefan.lk.hakansson@ericsson.com>, <rtcweb@ietf.org>
Subject:  Re: [rtcweb] Additional requirement - audio-only communication

[BA] Agreed that there seems to be some misconceptions about what the
current API proposal can do.

To clear this up, would it make sense to have a (brief) presentation on the
API at the interim?

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


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<html><head></head><body style=3D"word-wrap: break-word; -webkit-nbsp-mode: s=
pace; -webkit-line-break: after-white-space; color: rgb(0, 0, 0); font-size:=
 14px; font-family: Calibri, sans-serif; "><div>Or perhaps pointers on this =
list to the newest version every now and then (whenever a major update has h=
appened)?</div><div>Specifically, is Harald's draft from March 24 still the =
one to look at?</div><div>(The companion lists have quite a bit of traffic, =
and there are only so many cycles per day to monitor those, especially if AP=
I design is not in your core interest or competency).</div><div>Thanks,</div=
><div>Stephan</div><div><br></div><div><br></div><span id=3D"OLK_SRC_BODY_SECT=
ION"><div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color=
:black; 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> Bernard Aboba &lt;<a href=3D"mailto:bernard_aboba@hotmail.com">=
bernard_aboba@hotmail.com</a>&gt;<br><span style=3D"font-weight:bold">Date: </=
span> Sat, 27 Aug 2011 08:55:06 -0700<br><span style=3D"font-weight:bold">To: =
</span> &lt;<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.haka=
nsson@ericsson.com</a>&gt;, &lt;<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a>&gt;<br><span style=3D"font-weight:bold">Subject: </span> Re: [rtcweb]=
 Additional requirement - audio-only communication<br></div><div><br></div><=
div><style><!--
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--></style><div class=3D"hmmessage"><div dir=3D"ltr">
[BA] Agreed that there seems to be some misconceptions about what the curre=
nt API proposal can do. <br><br>To clear this up, would it make sense to hav=
e a (brief) presentation on the API at the interim?<br><br> 		 	   		  </div=
></div></div>_______________________________________________
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.org=
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The W3C WEBRTC wiki (which includes links to archives=2C tracker=2C editor'=
s draft=2C etc.) is available here: http://www.w3.org/2011/04/webrtc/wiki/M=
ain_Page

Currently a link points to a W3C editor's draft dated 23 August 2011: http:=
//dev.w3.org/2011/webrtc/editor/webrtc.html
Date: Sat=2C 27 Aug 2011 09:19:09 -0400
From: stewe@stewe.org
To: rtcweb@ietf.org
Subject: Re: [rtcweb] Additional requirement - audio-only communication



Or perhaps pointers on this list to the newest version every now and then (=
whenever a major update has happened)?Specifically=2C is Harald's draft fro=
m March 24 still the one to look at?(The companion lists have quite a bit o=
f traffic=2C and there are only so many cycles per day to monitor those=2C =
especially if API design is not in your core interest or competency).Thanks=
=2CStephan

From:  Bernard Aboba <bernard_aboba@hotmail.com>
Date:  Sat=2C 27 Aug 2011 08:55:06 -0700
To:  <stefan.lk.hakansson@ericsson.com>=2C <rtcweb@ietf.org>
Subject:  Re: [rtcweb] Additional requirement - audio-only communication


[BA] Agreed that there seems to be some misconceptions about what the curre=
nt API proposal can do.=20

To clear this up=2C would it make sense to have a (brief) presentation on t=
he API at the interim?

 		 	   		  _______________________________________________
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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</head>
<body class=3D'hmmessage'><div dir=3D'ltr'>
The W3C WEBRTC wiki (which includes links to archives=2C tracker=2C editor'=
s draft=2C etc.) is available here: http://www.w3.org/2011/04/webrtc/wiki/M=
ain_Page<br><br>Currently a link points to a W3C editor's draft dated 23 Au=
gust 2011: http://dev.w3.org/2011/webrtc/editor/webrtc.html<br><div><hr id=
=3D"stopSpelling">Date: Sat=2C 27 Aug 2011 09:19:09 -0400<br>From: stewe@st=
ewe.org<br>To: rtcweb@ietf.org<br>Subject: Re: [rtcweb] Additional requirem=
ent - audio-only communication<br><br>
<meta http-equiv=3D"Content-Type" content=3D"text/html=3B charset=3Dunicode=
">
<meta name=3D"Generator" content=3D"Microsoft SafeHTML"><div>Or perhaps poi=
nters on this list to the newest version every now and then (whenever a maj=
or update has happened)?</div><div>Specifically=2C is Harald's draft from M=
arch 24 still the one to look at?</div><div>(The companion lists have quite=
 a bit of traffic=2C and there are only so many cycles per day to monitor t=
hose=2C especially if API design is not in your core interest or competency=
).</div><div>Thanks=2C</div><div>Stephan</div><div><br></div><div><br></div=
><span id=3D"ecxOLK_SRC_BODY_SECTION"><div style=3D"font-family:Calibri=3Bf=
ont-size:11pt=3Btext-align:left=3Bcolor:black=3Bborder-bottom:medium none=
=3Bborder-left:medium none=3Bpadding-bottom:0in=3Bpadding-left:0in=3Bpaddin=
g-right:0in=3Bborder-top:#b5c4df 1pt solid=3Bborder-right:medium none=3Bpad=
ding-top:3pt"><span style=3D"font-weight:bold">From: </span> Bernard Aboba =
&lt=3B<a href=3D"mailto:bernard_aboba@hotmail.com">bernard_aboba@hotmail.co=
m</a>&gt=3B<br><span style=3D"font-weight:bold">Date: </span> Sat=2C 27 Aug=
 2011 08:55:06 -0700<br><span style=3D"font-weight:bold">To: </span> &lt=3B=
<a href=3D"mailto:stefan.lk.hakansson@ericsson.com">stefan.lk.hakansson@eri=
csson.com</a>&gt=3B=2C &lt=3B<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf=
.org</a>&gt=3B<br><span style=3D"font-weight:bold">Subject: </span> Re: [rt=
cweb] Additional requirement - audio-only communication<br></div><div><br><=
/div><div><style>
.ExternalClass .ecxhmmessage P
{padding:0px=3B}
.ExternalClass body.ecxhmmessage
{font-size:10pt=3Bfont-family:Tahoma=3B}

</style><div class=3D"ecxhmmessage"><div dir=3D"ltr">
[BA] Agreed that there seems to be some misconceptions about what the curre=
nt API proposal can do. <br><br>To clear this up=2C would it make sense to =
have a (brief) presentation on the API at the interim?<br><br> 		 	   		  <=
/div></div></div>_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</span>
<br>_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb</div> 		 	   		  </div></body>
</html>=

--_eb913a86-aa96-44b2-9637-85db1f139037_--

From hoang.su.tk@gmail.com  Sun Aug 28 02:15:43 2011
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From: Nguyen Duong Tuan <hoang.su.tk@gmail.com>
To: Ravindran Parthasarathi <pravindran@sonusnet.com>
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Some misunderstandings about <Usecase & architecture: Browser application with separate webserver & voipserver>
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I'm sorry for late reply due to my vacation. Thanks Partha, I've just
begin with RTCWeb so there're quite a few thing I'm really no clear.

Thanks,
TuanND

2011/8/26 Ravindran Parthasarathi <pravindran@sonusnet.com>:
> Hi TuanND,
>
> <snip>
> b) Browser application interacts with VoIP server (use HTTP)
> =C2=A0 =C2=A0 =C2=A0 =C2=A0browser<------>RTCweb server & SIP client (new=
 GW)<------>SIP
> server
>
> New Gateway is outside the scope of RTCWeb and IETF which provides the
> opportunity for innovative way of building gateways but may leads to
> interop issue due to mapping between RTCWebserver & SIP client w.r.t
> mid-call services.
> </snip>
>
> The issue in case webserver acts as new RTCWEB & SIP GW for VoIP
> communication, there is a need of protocol mapping between the protocol
> used between browser & RTCWebserver (say RTCweb protocol) to SIP. In my
> experience with signaling protocol interworking, the interop between two
> vendors fails lot of times in mid-call services like Hold, resume,
> transfer and IETF BLISS WG is good example for such failures. My
> question was why RTCWeb was looking into that approach.
>
> From http://www.ietf.org/mail-archive/web/rtcweb/current/msg00730.html
> mail thread, I understand that more discussion will occur in RTCWeb to
> clarify this doubt. Please let me know in case you need more
> explanation.
>
> Thanks
> Partha
>
> PS: I have changed my company recently
>
>>-----Original Message-----
>>From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf
>>Of Nguyen Duong Tuan
>>Sent: Thursday, August 18, 2011 7:40 AM
>>To: rtcweb@ietf.org
>>Subject: [rtcweb] Some misunderstandings about <Usecase & architecture:
>>Browser application with separate webserver & voipserver>
>>
>>Hello everyone,
>>
>>I read about Partha's question at RTCWeb mailing list at
>>http://www.ietf.org/mail-archive/web/rtcweb/current/msg00563.html
>>
>>He said that there's interoprating issues due to mapping between
>>RTCWebserver and SIP client w.r.t mid-call services in the
>>introduction of new gateway. Could anyone please explain more about
>>those issues? I have a little bit difficult to understand his
>>meanings.
>>
>>I couldn't send an email to Partha directly by partr@cisco.com, so I
>>decided to send to this email. Sorry for any inconvenience.
>>
>>Thanks,
>>TuanND
>>_______________________________________________
>>rtcweb mailing list
>>rtcweb@ietf.org
>>https://www.ietf.org/mailman/listinfo/rtcweb
>

From internet-drafts@ietf.org  Sun Aug 28 08:28:36 2011
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A New Internet-Draft is available from the on-line Internet-Drafts director=
ies. This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.

	Title           : Web Real-Time Communication Use-cases and Requirements
	Author(s)       : Christer Holmberg
                          Stefan Hakansson
                          Goran AP Eriksson
	Filename        : draft-ietf-rtcweb-use-cases-and-requirements-03.txt
	Pages           : 16
	Date            : 2011-08-28

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


A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-require=
ments-03.txt

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

This Internet-Draft can be retrieved at:
ftp://ftp.ietf.org/internet-drafts/draft-ietf-rtcweb-use-cases-and-requirem=
ents-03.txt

From csp@csperkins.org  Sun Aug 28 10:58:33 2011
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Begin forwarded message:
> From: internet-drafts@ietf.org
> Date: 28 August 2011 18:54:41 GMT+01:00
> Subject: New Version Notification for =
draft-perkins-rtcweb-rtp-usage-03.txt
>=20
> A new version of I-D, draft-perkins-rtcweb-rtp-usage-03.txt has been =
successfully submitted by Colin Perkins and posted to the IETF =
repository.
>=20
> Filename:	 draft-perkins-rtcweb-rtp-usage
> Revision:	 03
> Title:		 RTP Requirements for RTC-Web
> Creation date:	 2011-08-28
> WG ID:		 Individual Submission
> Number of pages: 22

This version makes the following changes relative to the -02 draft:

- Removed discussion of multiplexing for RTP sessions entirely, pending =
submission of a proposal for multiplexing audio and video within an RTP =
session (as discussed at IETF 81).

- Added a reference to draft-ietf-avtcore-srtp-vbr-audio-03 in Security =
Considerations.

- Added a recommendation that =
draft-ietf-avtcore-srtp-encrypted-header-ext-00 be used when the =
client-to-mixer and mixer-to-client audio level indication RTP header =
extensions are in use in SRTP encrypted sessions.

- Added Slice Loss Indication and Reference Picture Selection Indication =
messages as OPTIONAL loss tolerance mechanisms.

Comments and feedback are welcome.

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




From stewe@stewe.org  Sun Aug 28 16:06:22 2011
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Hi,

I have a couple of comments related to
draft-ietf-rtcweb-use-cases-and-requirements-03.txt.

1. There are a number of editorial issues that need to be addressed; I
will communicate those to the editors in private.

2. Section 4.2.1.1: in a typical system, self-view is enabled well before
connection establishment, not after.  For obvious reasons: you want to
check your grooming before establishing a videoconference session (or
taking a video call).

3. Section 4.2.4.1.: QoS related issues do not only occur in "roaming"
scenarios; therefore, I would suggest to make the description of this use
case independent from the "previous one" (4.2.3), and rather refer to
4.2.1.

4. Section 4.2.6.1: This use case is not described in sufficient detail.
At least two scenarios are possible.  First, both front and rear camera
send individual video streams (potentially at different resolutions), and
the PIP mixing happens in the receiving browser.  This would be a user
interface issue and no mechanisms need to be specified in the IETF beyond
being bale to send more than one video stream (though there may be need
for related API work).  Second, the PIP mixing happens in the sending
phone, and only one stream is being sent.  In this case, I believe not
even API work is necessary.
Suggest to reconsider whether this use case is relevant enough for being
kept.  Multi-camera systems being able to send coded samples from both
cameras simultaneously are rather exotic today (only telepresence rooms
come to my mind).

5. Section 4.2.7.1.: Why are the sending peers restricted to mono audio?
Spatial arrangement is not very complex for stereo as well...

6. Section 4.2.9.1.: What's really necessary here is a mechanism that
allows a user to tell a browser that VERY tight cross-signal sync and VERY
low delay is required, which may trigger different jitter buffer handling
and such.  Beyond that, I believe that audio codec negotiation may be
helpful.  Audio professionals (like musicians) are somewhat more picky
when it comes to these technology selections than normal users.  I would
not be surprised if we would learn that there is a real market requirement
for uncompressed or lossless audio if this use case takes off.

Section 4.3.3.1: I have a number of issues with this use case.

First, in contrast to most other use cases, this one enters solution space
quite prominently.  That wouldn't be an issue for me if the solution my
employer is favoring were mentioned here, but it is not :-(.  To cure my
immediate concern, one suggestion would be to remove references to
simulcast and/or add references to spatial scalability.  However, perhaps
it's better to describe the behavior of the multipoint system in terms of
user experience rather than technology choice.

Second, why is audio mixed to stereo and not to something else, such as
5.1?  

Third, the security stuff is not in any way technically bound to the rest
of the use case, so I would farm it out into its own use case, and/or
mention it as a "generic" feature... Remarks like "it is essential that
the communication cannot be eavesdropped" would apply to pretty much all
use cases, right?

7. Missing use case:

It is my understanding that for regulatory compliance, in many developed
countries, there will be a need for an E911 type of service *IF* the
solution allows to "dial" an E.164 phone number.  I remember a controversy
involving SKYPE in ca. 2005, and also having read about recent FCC
hearings about this issue; for example,
http://www.broadbandlawadvisor.com/2011/07/articles/voip/fcc-seeks-comment-
on-extending-e911-rules-to-oneway-outboundonly-voip-improve-location-capabi
lity-of-inteconnected-voip/.
If there is a reasonable expectation that a webrtc service with outbound
dialing capability in E.164 number-space requires E911 handling, then it
does not make sense to stick our collective heads in the sand and ignore
the issue.  I believe there is such an expectation; surely during the
lifetime of an webrtc solution, but probably even during its introducer
phase.
If E911 is relevant in this sense, then this issue needs to be addressed
in section 4.3.1, 4.3.2, and perhaps 4.2.5.
I understand that the editors did not address those use cases yet based on
(presumably) lack of consensus, but I fear that IETF consensus is not the
only relevant factor here.

(I could mention "legal intercept" in the same context, but suggest to
focus on emergency calls first, because a) they are easier to handle, b)
more widely applicable, and c) generally agreed to be a useful thing, and
therefore not quite as politically loaded.)

Stephan




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> It is my understanding that for regulatory compliance=2C in many develope=
d
> countries=2C there will be a need for an E911 type of service *IF* the
> solution allows to "dial" an E.164 phone number. =20

[BA] The following FAQ describes some of the current obligations of "interc=
onnected VoIP" services within the US:
http://transition.fcc.gov/voip/

While the current definition requires the ability to make outbound calls as=
 well as receive inbound calls=2C there is a proceeding in progress to amen=
d the definition:
http://www.fcc.gov/document/amending-definition-interconnected-voip-service=
-section-93-commissions-rules-wireless-e911-

>From a technical perspective=2C it isn't clear to me that the difference=20
between the existing and proposed definitions of "interconnected VoIP"=20
will have much impact on the RTCWEB requirements. =20

> If there is a reasonable expectation that a webrtc service with outbound
> dialing capability in E.164 number-space requires E911 handling=2C then i=
t
> does not make sense to stick our collective heads in the sand and ignore =
the issue. =20

I don't think that the WG is ignoring the issue.  It's just that there are =
quite a few proceedings in progress in the US=2C EU and other places around=
 the globe right now=2C and it's likely that we'll have a better picture of=
 the regulatory requirements within a year (or even 6 months) compared to w=
here we are today.=20
 		 	   		  =

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<body class=3D'hmmessage'><div dir=3D'ltr'>
<br><div>&gt=3B It is my understanding that for regulatory compliance=2C in=
 many developed<br>&gt=3B countries=2C there will be a need for an E911 typ=
e of service *IF* the<br>&gt=3B solution allows to "dial" an E.164 phone nu=
mber.  <br><br>[BA] The following FAQ describes some of the current obligat=
ions of "interconnected VoIP" services within the US:<br>http://transition.=
fcc.gov/voip/<br><br>While the current definition requires the ability to m=
ake outbound calls as well as receive inbound calls=2C there is a proceedin=
g in progress to amend the definition:<br>http://www.fcc.gov/document/amend=
ing-definition-interconnected-voip-service-section-93-commissions-rules-wir=
eless-e911-<br><br>From a technical perspective=2C it isn't clear to me tha=
t the difference=20
between the existing and proposed definitions of "interconnected VoIP"=20
will have much impact on the RTCWEB requirements.&nbsp=3B <br><br>&gt=3B If=
 there is a reasonable expectation that a webrtc service with outbound<br>&=
gt=3B dialing capability in E.164 number-space requires E911 handling=2C th=
en it<br>&gt=3B does not make sense to stick our collective heads in the sa=
nd and ignore the issue.  <br><br>I don't think that the WG is ignoring the=
 issue.&nbsp=3B It's just that there are quite a few proceedings in progres=
s in the US=2C EU and other places around the globe right now=2C and it's l=
ikely that we'll have a better picture of the regulatory requirements withi=
n a year (or even 6 months) compared to where we are today. <br></div> 		 	=
   		  </div></body>
</html>=

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Hi,

I have a couple of comments on the use-case sections 4.2 and 4.3 of
draft-ietf-rtcweb-use-cases-and-requirements-03

=20

Section 4.2.1.1

I suppose when you say "mute incoming media" you are giving the
flexibility to mute audio or video or both. You may want to explicitly
spell out that similar to the way it is done for sending media.

=20

Section 4.2.3

When a session moves across different network adapters is the assumption
the same user device is being used ? or can the user device also change
? if the user device can change then the devices for audio, video
input/output may also change. Is this covered ? or not in scope ?

=20

=20

Section 4.2.7.1

I see that it is mentioned in the draft that mixing of audio is locally
done in the browser. Do we also want to consider a case where mixing can
be done by a network element rather than a browser ? I see there is a
use-case in Browser/Server that has this in section 4.3.3.1. Is this
covering the cases where a browser is not capable of mixing and a server
is involved between a Browser to browser call ?

=20

Regards,

Ram


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</o:shapelayout></xml><![endif]--></head><body lang=3DEN-US link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal>Hi,<o:p></o:p></p><p class=3DMsoNormal>I have a couple =
of comments on the use-case sections 4.2 and 4.3 of =
draft-ietf-rtcweb-use-cases-and-requirements-03<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Section =
4.2.1.1<o:p></o:p></p><p class=3DMsoNormal>I suppose when you say =
&quot;mute incoming media&quot; you are giving the flexibility to mute =
audio or video or both. You may want to explicitly spell out that =
similar to the way it is done for sending media.<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Section =
4.2.3<o:p></o:p></p><p class=3DMsoNormal>When a session moves across =
different network adapters is the assumption the same user device is =
being used ? or can the user device also change ? if the user device can =
change then the devices for audio, video input/output may also change. =
Is this covered ? or not in scope ?<o:p></o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><p class=3DMsoNormal>Section =
4.2.7.1<o:p></o:p></p><p class=3DMsoNormal>I see that it is mentioned in =
the draft that mixing of audio is locally done in the browser. Do we =
also want to consider a case where mixing can be done by a network =
element rather than a browser ? I see there is a use-case in =
Browser/Server that has this in section 4.3.3.1. Is this covering the =
cases where a browser is not capable of mixing and a server is involved =
between a Browser to browser call ?<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>Ram<o:p></o:p></p></div></body></html>
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From harald@alvestrand.no  Mon Aug 29 02:44:28 2011
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Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Draft agenda RTCWeb call on September 8th
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Given that draft-perkins-rtcweb-rtp-usage-03 is just out, should we 
include that on the agenda?

On 08/27/11 01:09, Cullen Jennings wrote:
> We need to start working on an agenda for the Sept 8th - bellow is  very rough straw man. Please suggest ideas for things to add, delete, change ...
>
>
> --------------------
>
>
> Use Case - John (60 min)
>
>      Identity in multiuser calls
>
>      Recording
could you add a little more text here (in particular as John is a 
multicast address)?
I think I know where "recording" comes from, but what is the "identity" 
point?
>
> Terminology Mapping RTP to W3C ( 30 min)
>      - what's a track, session, flow, connection and how do they relate
This is a discussion that's been happening on the W3C list, for those of 
you who don't read that.
>
> Just One Port - (AKA Multiplexing) -  Harald  (30 min)
>
>
> Non Media Data Transport (30 min)
>
>
> Security and User Interface (90 min)
>       What do we expect the see and do in relation to security and authorization around access to microphone/camera
This has previously been said to be a W3C discussion - and the W3C WG is 
having a meeting on Aug 31.
Don't know if 90 minutes of IETF time is the Right Thing here.
>
> Other things that I don't see as being ready to spend time on ...
>
>
> NAT
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From harald@alvestrand.no  Mon Aug 29 06:31:47 2011
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Subject: [rtcweb] Review of draft-perkins-rtcweb-usage-03 (Re: Fwd: New Version Notification for	draft-perkins-rtcweb-rtp-usage-03.txt)
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Colin, I really like this version!

There might want to be a placeholder about multiplexing, saying "we will 
return to this issue after more discussion". Otherwise, it seems to me 
we're setting up the expectation that this will be handled entirely in 
some other document - I think the end product of the WG needs to discuss 
it, and it seems logical that some aspect of it should go here.

Some detailed comments:

- In Figure 2, section 1.1, I think you're making the assumption that 
multi-unicast topologies will use a single shared RTP session (SSRC 
number space) for all the links. This is not obvious; it's possible to 
build this topology on point-to-point RTP sessions too.

- In section 3, you make the point that improper signalling of bandwidth 
can cause failure to interoperate (because of differing RTCP timings). 
Is there a (possibly theoretical) problem with interoperation between 
RTP/AVP and RTP/AVPF too, or have we verified that the AVPF profile 
always sends enough RTCP packets that AVP-conformant endpoints don't 
time out?

- In section 6, I would recommend removing the point about scenarios 
with mixers using point-to-point RTP sessions are "not well utilizing 
the mechanisms of RTP" - I think people's engineering tradeoffs should 
be respected.
I'm fine with leaving in the comment about protocol violations (although 
I'd like to be more specific about what they are - protocols shouldn't 
be violated; if people "have" to do that, there's something wrong with 
the protocol).

- In the list of other extensions, section 6.2, you say that two 
extensions are "not recommended" - is this a recommendation against 
(like NOT RECOMMENDED would be), or simply a declaration that their use 
or non-use is of no concern?

- there are some sections with garbled grammar (the last paragraph of 
section 7.2, before 7.2.1, is particularly irksome to the eye), but I 
assume that this will be reviewed in later versions; the meaning comes 
through.



On 08/28/11 19:59, Colin Perkins wrote:
> Begin forwarded message:
>> From: internet-drafts@ietf.org
>> Date: 28 August 2011 18:54:41 GMT+01:00
>> Subject: New Version Notification for draft-perkins-rtcweb-rtp-usage-03.txt
>>
>> A new version of I-D, draft-perkins-rtcweb-rtp-usage-03.txt has been successfully submitted by Colin Perkins and posted to the IETF repository.
>>
>> Filename:	 draft-perkins-rtcweb-rtp-usage
>> Revision:	 03
>> Title:		 RTP Requirements for RTC-Web
>> Creation date:	 2011-08-28
>> WG ID:		 Individual Submission
>> Number of pages: 22
> This version makes the following changes relative to the -02 draft:
>
> - Removed discussion of multiplexing for RTP sessions entirely, pending submission of a proposal for multiplexing audio and video within an RTP session (as discussed at IETF 81).
>
> - Added a reference to draft-ietf-avtcore-srtp-vbr-audio-03 in Security Considerations.
>
> - Added a recommendation that draft-ietf-avtcore-srtp-encrypted-header-ext-00 be used when the client-to-mixer and mixer-to-client audio level indication RTP header extensions are in use in SRTP encrypted sessions.
>
> - Added Slice Loss Indication and Reference Picture Selection Indication messages as OPTIONAL loss tolerance mechanisms.
>
> Comments and feedback are welcome.
>


From Christian.Groves@nteczone.com  Mon Aug 29 18:14:25 2011
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Subject: Re: [rtcweb] connection event notification
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Hello Matthew,

You may also want to take a look at:
http://www.iana.org/assignments/megaco-h248

There's plenty of Megaco packages that may have notification events and 
statistics that would also be relevant.

Regards, Christian

On 6/08/2011 11:11 AM, Matthew Kaufman wrote:
> While doing the research for the previous thread, I saw that MGCP 
> actually has some things we might wish to borrow for RTCWEB. 
> Specifically, some of the notification events... would sure be nice to 
> get Javascript events when we saw "packet loss exceeded" or "RTP/RTCP 
> timeout" for instance so that the web application could take an 
> appropriate action.
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

From stefan.lk.hakansson@ericsson.com  Tue Aug 30 03:48:18 2011
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Subject: Re: [rtcweb] Comments on use case draft
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Stephan,

thanks for very good input! I have made some answers below, and look 
forward to your, and the group's, feedback to them.

Br,
Stefan

On 2011-08-28 22:07, Stephan Wenger wrote:
> Hi,
>
> I have a couple of comments related to
> draft-ietf-rtcweb-use-cases-and-requirements-03.txt.
>
> 1. There are a number of editorial issues that need to be addressed; I
> will communicate those to the editors in private.
>
> 2. Section 4.2.1.1: in a typical system, self-view is enabled well before
> connection establishment, not after.  For obvious reasons: you want to
> check your grooming before establishing a videoconference session (or
> taking a video call).

That is right. We might re-phrase it a bit. But what we really was after 
was the flexibility for the web app to display, or not, a self view (and 
to add and remove it at any time).

>
> 3. Section 4.2.4.1.: QoS related issues do not only occur in "roaming"
> scenarios; therefore, I would suggest to make the description of this use
> case independent from the "previous one" (4.2.3), and rather refer to
> 4.2.1.

The reason for pointing at 4.2.3 was to say that QoS capabilities for 
different accesses should be possible to use. If we refer to 4.2.1 this 
is much more unclear. The users in 4.2.1 may very well be in a situation 
where there is no possibility to offer QoS.

>
> 4. Section 4.2.6.1: This use case is not described in sufficient detail.
> At least two scenarios are possible.  First, both front and rear camera
> send individual video streams (potentially at different resolutions), and
> the PIP mixing happens in the receiving browser.  This would be a user
> interface issue and no mechanisms need to be specified in the IETF beyond
> being bale to send more than one video stream (though there may be need
> for related API work).  Second, the PIP mixing happens in the sending
> phone, and only one stream is being sent.  In this case, I believe not
> even API work is necessary.

We were after the first case (two independent video streams). That is 
also the requirement derived from this use case.

> Suggest to reconsider whether this use case is relevant enough for being
> kept.  Multi-camera systems being able to send coded samples from both
> cameras simultaneously are rather exotic today (only telepresence rooms
> come to my mind).

I have a different view. The average device (smartphone, tablet) is to a 
higher and higher degree equipped with two cameras. I think it should be 
possible to use both.

>
> 5. Section 4.2.7.1.: Why are the sending peers restricted to mono audio?
> Spatial arrangement is not very complex for stereo as well...

That is a very valid comment. Should we just remove 'mono' and be silent 
on that aspect?

>
> 6. Section 4.2.9.1.: What's really necessary here is a mechanism that
> allows a user to tell a browser that VERY tight cross-signal sync and VERY
> low delay is required, which may trigger different jitter buffer handling
> and such.  Beyond that, I believe that audio codec negotiation may be
> helpful.  Audio professionals (like musicians) are somewhat more picky
> when it comes to these technology selections than normal users.  I would
> not be surprised if we would learn that there is a real market requirement
> for uncompressed or lossless audio if this use case takes off.

I think there is a lot to be investigated for this use case really. I 
got the tip that this research group as Stanford: 
https://ccrma.stanford.edu/groups/soundwire/
has done a lot and have a lot of real-life experience. Probably we 
should get in contact with them. As you suspect, they seem to use 
uncompressed audio.

>
> Section 4.3.3.1: I have a number of issues with this use case.
>
> First, in contrast to most other use cases, this one enters solution space
> quite prominently.  That wouldn't be an issue for me if the solution my
> employer is favoring were mentioned here, but it is not :-(.  To cure my
> immediate concern, one suggestion would be to remove references to
> simulcast and/or add references to spatial scalability.  However, perhaps
> it's better to describe the behavior of the multipoint system in terms of
> user experience rather than technology choice.

I think this is a valid point. Let's see if we can come up with something.

>
> Second, why is audio mixed to stereo and not to something else, such as
> 5.1?

Right. It should not say "stereo", but something like "stereo, 5.1, or 
similar".

>
> Third, the security stuff is not in any way technically bound to the rest
> of the use case, so I would farm it out into its own use case, and/or
> mention it as a "generic" feature... Remarks like "it is essential that
> the communication cannot be eavesdropped" would apply to pretty much all
> use cases, right?

I guess that depends. If you have a leisure conversation, maybe you do 
not care that much. This was a case where we could clearly say that "not 
being eavesdropped" is essential.

But we could say something about it already in the very first use case 
(the most basic one), and derive the requirement already there. Perhaps 
we could say that the conversation in that use case is about planning a 
bank robbery:)

>
> 7. Missing use case:
>
> It is my understanding that for regulatory compliance, in many developed
> countries, there will be a need for an E911 type of service *IF* the
> solution allows to "dial" an E.164 phone number.  I remember a controversy
> involving SKYPE in ca. 2005, and also having read about recent FCC
> hearings about this issue; for example,
> http://www.broadbandlawadvisor.com/2011/07/articles/voip/fcc-seeks-comment-
> on-extending-e911-rules-to-oneway-outboundonly-voip-improve-location-capabi
> lity-of-inteconnected-voip/.
> If there is a reasonable expectation that a webrtc service with outbound
> dialing capability in E.164 number-space requires E911 handling, then it
> does not make sense to stick our collective heads in the sand and ignore
> the issue.  I believe there is such an expectation; surely during the
> lifetime of an webrtc solution, but probably even during its introducer
> phase.
> If E911 is relevant in this sense, then this issue needs to be addressed
> in section 4.3.1, 4.3.2, and perhaps 4.2.5.
> I understand that the editors did not address those use cases yet based on
> (presumably) lack of consensus, but I fear that IETF consensus is not the
> only relevant factor here.

Use cases 4.3.1 and 4.3.2 do say that you should be able to dial E.164 
using the rtcweb standard IMO. But we never really discussed to what 
extent this means E911 must be supported, and in turn what requirements 
it puts on the rtcweb standard. One aspect is of course location; but 
this is in a web browser context handled by the W3C geolocation WG 
(http://www.w3.org/2008/geolocation/).

>
> (I could mention "legal intercept" in the same context, but suggest to
> focus on emergency calls first, because a) they are easier to handle, b)
> more widely applicable, and c) generally agreed to be a useful thing, and
> therefore not quite as politically loaded.)
>
> Stephan
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From christer.holmberg@ericsson.com  Tue Aug 30 04:20:07 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Tue, 30 Aug 2011 13:21:31 +0200
Thread-Topic: Multiplexing using the same port number for multiple media descritions
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Hi,

One possible alternative solution for SDP multiplex negotiation could be ba=
sed on the assumption of using the same port number in multiple SDP m- line=
s (yes, I know SDP does not allow it, and I will come back to that).

Something like:

SDP offer:

m=3Daudio 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...



MAYBE there is also a need to use some kind of grouping, in which case it c=
ould look something like (borrowing some terminology from Harald):



SDP offer:

a=3Dgroup:TOGETHER foo bar
m=3Daudio 10000 ...
a=3Dmid:foo
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
a=3Dmid:foo
m=3Dvideo 20000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...


An ISSUE with this solution is of course that SDP does not allow for it.

However, we could always say that browsers must support it, in which case i=
t should work fine in direct browser-to-browser cases.


When interworking with legacy, I guess two things can happen:

1. The offer is acctepted, with different port number in the answer, and mu=
ltiplex won't be used (see example above)

2. The offer is rejected. In this case, the fallback would be that the brow=
ser sends a new offer, with different port numbers, and multiplex won't be =
used.

Regards,

Christer




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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
<meta name=3D"Generator" content=3D"Microsoft Exchange Server">
<!-- converted from rtf -->
<style><!-- .EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left:=
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</head>
<body>
<font face=3D"Arial, sans-serif" size=3D"2">
<div>Hi,</div>
<div>&nbsp;</div>
<div>One possible alternative solution for SDP multiplex negotiation could =
be based on the assumption of using the same port number in multiple SDP m-=
 lines (yes, I know SDP does not allow it, and I will come back to that).</=
div>
<div>&nbsp;</div>
<div>Something like:</div>
<div>&nbsp;</div>
<div>SDP offer:</div>
<div>&nbsp;</div>
<div>m=3Daudio 10000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 10000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>SDP answer (multiplex supported/accepted):</div>
<div>&nbsp;</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>SDP answer (multiplex not-supported/rejected):</div>
<div>&nbsp;</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 30000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>MAYBE there is also a need to use some kind of grouping, in which case=
 it could look something like (borrowing some terminology from Harald):</di=
v>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>SDP offer:</div>
<div>&nbsp;</div>
<div>a=3Dgroup:TOGETHER foo bar</div>
<div>m=3Daudio 10000 ...</div>
<div>a=3Dmid:foo</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 10000 ...</div>
<div>a=3Dmid:bar</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>SDP answer (multiplex supported/accepted):</div>
<div>&nbsp;</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Dmid:foo</div>
<div>m=3Dvideo 20000 ...</div>
<div>a=3Dmid:bar</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>SDP answer (multiplex not-supported/rejected):</div>
<div>&nbsp;</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 30000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>An ISSUE with this solution is of course that SDP does not allow for i=
t.</div>
<div>&nbsp;</div>
<div>However, we could always say that browsers must support it, in which c=
ase it should work fine in direct browser-to-browser cases.</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>When interworking with legacy, I guess two things can happen:</div>
<div>&nbsp;</div>
<div>1. The offer is acctepted, with different port number in the answer, a=
nd multiplex won't be used (see example above)</div>
<div>&nbsp;</div>
<div>2. The offer is rejected. In this case, the fallback would be that the=
 browser sends a new offer, with different port numbers, and multiplex won'=
t be used.</div>
<div>&nbsp;</div>
<div>Regards,</div>
<div>&nbsp;</div>
<div>Christer</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
<div>&nbsp;</div>
</font>
</body>
</html>

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I think Harald's approach is cleaner, since the fallback does not require a
new offer. What do you see as problematic with Harald's suggestion?

On Tue, Aug 30, 2011 at 7:21 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

>  Hi,
>
> One possible alternative solution for SDP multiplex negotiation could be
> based on the assumption of using the same port number in multiple SDP m-
> lines (yes, I know SDP does not allow it, and I will come back to that).
>
> Something like:
>
> SDP offer:
>
> m=audio 10000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 10000 ...
> a=rtpmap ...
> a=rtpmap ...
>
>
> SDP answer (multiplex supported/accepted):
>
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 20000 ...
> a=rtpmap ...
> a=rtpmap ...
>
>
> SDP answer (multiplex not-supported/rejected):
>
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 30000 ...
> a=rtpmap ...
> a=rtpmap ...
>
>
>
> MAYBE there is also a need to use some kind of grouping, in which case it
> could look something like (borrowing some terminology from Harald):
>
>
>
> SDP offer:
>
> a=group:TOGETHER foo bar
> m=audio 10000 ...
> a=mid:foo
> a=rtpmap ...
> a=rtpmap ...
> m=video 10000 ...
> a=mid:bar
> a=rtpmap ...
> a=rtpmap ...
>
>
> SDP answer (multiplex supported/accepted):
>
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> a=mid:foo
> m=video 20000 ...
> a=mid:bar
> a=rtpmap ...
> a=rtpmap ...
>
>
> SDP answer (multiplex not-supported/rejected):
>
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 30000 ...
> a=rtpmap ...
> a=rtpmap ...
>
>
> An ISSUE with this solution is of course that SDP does not allow for it.
>
> However, we could always say that browsers must support it, in which case
> it should work fine in direct browser-to-browser cases.
>
>
> When interworking with legacy, I guess two things can happen:
>
> 1. The offer is acctepted, with different port number in the answer, and
> multiplex won't be used (see example above)
>
> 2. The offer is rejected. In this case, the fallback would be that the
> browser sends a new offer, with different port numbers, and multiplex won't
> be used.
>
> Regards,
>
> Christer
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--0015176f09d8ba030904abba04be
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

I think Harald&#39;s approach is cleaner, since the fallback does not requi=
re a new offer. What do you see as problematic with Harald&#39;s suggestion=
?<div><br><div class=3D"gmail_quote">On Tue, Aug 30, 2011 at 7:21 AM, Chris=
ter Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@eric=
sson.com">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>
<font face=3D"Arial, sans-serif" size=3D"2">
<div>Hi,</div>
<div>=C2=A0</div>
<div>One possible alternative solution for SDP multiplex negotiation could =
be based on the assumption of using the same port number in multiple SDP m-=
 lines (yes, I know SDP does not allow it, and I will come back to that).</=
div>


<div>=C2=A0</div>
<div>Something like:</div>
<div>=C2=A0</div>
<div>SDP offer:</div>
<div>=C2=A0</div>
<div>m=3Daudio 10000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 10000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>SDP answer (multiplex supported/accepted):</div>
<div>=C2=A0</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>SDP answer (multiplex not-supported/rejected):</div>
<div>=C2=A0</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 30000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>MAYBE there is also a need to use some kind of grouping, in which case=
 it could look something like (borrowing some terminology from Harald):</di=
v>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>SDP offer:</div>
<div>=C2=A0</div>
<div>a=3Dgroup:TOGETHER foo bar</div>
<div>m=3Daudio 10000 ...</div>
<div>a=3Dmid:foo</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 10000 ...</div>
<div>a=3Dmid:bar</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>SDP answer (multiplex supported/accepted):</div>
<div>=C2=A0</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Dmid:foo</div>
<div>m=3Dvideo 20000 ...</div>
<div>a=3Dmid:bar</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>SDP answer (multiplex not-supported/rejected):</div>
<div>=C2=A0</div>
<div>m=3Daudio 20000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>m=3Dvideo 30000 ...</div>
<div>a=3Drtpmap ...</div>
<div>a=3Drtpmap ...</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>An ISSUE with this solution is of course that SDP does not allow for i=
t.</div>
<div>=C2=A0</div>
<div>However, we could always say that browsers must support it, in which c=
ase it should work fine in direct browser-to-browser cases.</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>When interworking with legacy, I guess two things can happen:</div>
<div>=C2=A0</div>
<div>1. The offer is acctepted, with different port number in the answer, a=
nd multiplex won&#39;t be used (see example above)</div>
<div>=C2=A0</div>
<div>2. The offer is rejected. In this case, the fallback would be that the=
 browser sends a new offer, with different port numbers, and multiplex won&=
#39;t be used.</div>
<div>=C2=A0</div>
<div>Regards,</div>
<div>=C2=A0</div><span class=3D"HOEnZb"><font color=3D"#888888">
<div>Christer</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
<div>=C2=A0</div>
</font></span></font>
</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>

--0015176f09d8ba030904abba04be--

From randell-ietf@jesup.org  Tue Aug 30 08:07:48 2011
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On 8/28/2011 4:07 PM, Stephan Wenger wrote:
> 4. Section 4.2.6.1: This use case is not described in sufficient detail.
> At least two scenarios are possible.  First, both front and rear camera
> send individual video streams (potentially at different resolutions), and
> the PIP mixing happens in the receiving browser.  This would be a user
> interface issue and no mechanisms need to be specified in the IETF beyond
> being bale to send more than one video stream (though there may be need
> for related API work).  Second, the PIP mixing happens in the sending
> phone, and only one stream is being sent.  In this case, I believe not
> even API work is necessary.
> Suggest to reconsider whether this use case is relevant enough for being
> kept.  Multi-camera systems being able to send coded samples from both
> cameras simultaneously are rather exotic today (only telepresence rooms
> come to my mind).
I think this directly talks to a common case that would interest many news
organizations: personal news reporting.  CNN iReport, even local news channels -
they love having users act as reporters-on-the spot, and having the video from
both cameras is a big win for them (and for the producers working from such
footage, who could swap the PIPs or suppress one or the other on the fly).
And most newer Android phones and iPhones have two cameras.

Could I live with losing this use-case?  Yes, with pain.  I do want to support
multiple streams so you can have the user and a local video (either a file, a
camera, or a video encoding of a desktop or window).  I'll note this usecase
only hits the two-cameras part of what I'd like to see, but I don't know
we need to go into that detail here (i.e. where other than a camera a stream
can come from is not something we need to mandate here - I think we just need
to call out the need for two streams).

> 5. Section 4.2.7.1.: Why are the sending peers restricted to mono audio?
> Spatial arrangement is not very complex for stereo as well...
>
> 6. Section 4.2.9.1.: What's really necessary here is a mechanism that
> allows a user to tell a browser that VERY tight cross-signal sync and VERY
> low delay is required, which may trigger different jitter buffer handling
> and such.  Beyond that, I believe that audio codec negotiation may be
> helpful.  Audio professionals (like musicians) are somewhat more picky
> when it comes to these technology selections than normal users.  I would
> not be surprised if we would learn that there is a real market requirement
> for uncompressed or lossless audio if this use case takes off.
Distributed music band - that needs TIGHT N-browser N-stream sync and
VERY LOW (and virtually constant i.e. lan) delay.  I do not believe this is
likely to be technically feasible such that users would accept it.

It would likely need ultra-low delay jitter buffers, much lower packetization
sizes, uncompressed audio, etc.

Distribution of music to multiple playback stations: more possible; the sync
requirements are somewhat relaxed, and the ultra-low delay is relaxed much more.

Let's drop this one, please.


>
> Section 4.3.3.1: I have a number of issues with this use case.
>
> First, in contrast to most other use cases, this one enters solution space
> quite prominently.  That wouldn't be an issue for me if the solution my
> employer is favoring were mentioned here, but it is not :-(.  To cure my
> immediate concern, one suggestion would be to remove references to
> simulcast and/or add references to spatial scalability.  However, perhaps
> it's better to describe the behavior of the multipoint system in terms of
> user experience rather than technology choice.
Generally I agree.  The use case can be more user-oriented.  What are we targeting
for the space here: a conference system tightly tied to an application on the
browser, or a generic conference system which would allow better operation when
you have "interop" calls between services - i.e. can someone on Facebook join
a rtcweb Hangout hosted on Google?  These choices would drive different requirements
to be derived from the usecase.

Also, we're describing things an application *could* do with the system, not the only
or even preferred way to do a conference.  This usecase states that someone *could*
build a "dumb" conference server that does no re-encoding, just stream selection and
forwarding.  It doesn't prevent a conference server that re-encodes, nor a conference
system using SVC or equivalent that subsets the incoming stream.

The application could request that a second smaller stream be sent, though obviously
this presumes the implementation, so it would be more tied to the conference server
implementation.  I'm wondering if there would be a good way to say "find a way to
deliver a low and high resolution image", and let the system (rtcweb) figure out how
to do it given the shared codecs available.  (I.e. SVC in a particular config if both
the conf server and browser support it, simulcast streams if they don't.)


> Second, why is audio mixed to stereo and not to something else, such as
> 5.1?
Remove reference to the number of channels.  That's handled via negotiation as
normal.


>
> Third, the security stuff is not in any way technically bound to the rest
> of the use case, so I would farm it out into its own use case, and/or
> mention it as a "generic" feature... Remarks like "it is essential that
> the communication cannot be eavesdropped" would apply to pretty much all
> use cases, right?
Per someone else's comment: we can drop this and add a separate usecase for
planning a bank robbery.  :-)  Better, though, might be a lawyer-client
conversation or a secret agent. :-)


>
> 7. Missing use case:
>
> It is my understanding that for regulatory compliance, in many developed
> countries, there will be a need for an E911 type of service *IF* the
> solution allows to "dial" an E.164 phone number.  I remember a controversy
> involving SKYPE in ca. 2005, and also having read about recent FCC
> hearings about this issue; for example,
> http://www.broadbandlawadvisor.com/2011/07/articles/voip/fcc-seeks-comment-
> on-extending-e911-rules-to-oneway-outboundonly-voip-improve-location-capabi
> lity-of-inteconnected-voip/.
> If there is a reasonable expectation that a webrtc service with outbound
> dialing capability in E.164 number-space requires E911 handling, then it
> does not make sense to stick our collective heads in the sand and ignore
> the issue.  I believe there is such an expectation; surely during the
> lifetime of an webrtc solution, but probably even during its introducer
> phase.
Agreed.  I've dealt with FCC rulings (and I'm sure other countries will have
similar and possibly even stricter requirements).  Basically, it will be very
likely that a provider will want to connect rtcweb to the PSTN (even if through
a gateway), and once that's done they'll need to support E911 services.

I could even see future expansion of the technologies for emergency calling;
you're seeing this now with emergency centers looking to support text messages.
So you might have emergency centers with rtcweb support directly or via a
translation/forwarding service (as voip is generally done today).  There is
significant advantage to an emergency center to be able to support video calls,
for example (though obviously there would be issues surrounding that).

> If E911 is relevant in this sense, then this issue needs to be addressed
> in section 4.3.1, 4.3.2, and perhaps 4.2.5.
> I understand that the editors did not address those use cases yet based on
> (presumably) lack of consensus, but I fear that IETF consensus is not the
> only relevant factor here.
>
> (I could mention "legal intercept" in the same context, but suggest to
> focus on emergency calls first, because a) they are easier to handle, b)
> more widely applicable, and c) generally agreed to be a useful thing, and
> therefore not quite as politically loaded.)

Yes.  :-)

-- 
Randell Jesup
randell-ietf@jesup.org


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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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On 08/30/11 13:21, Christer Holmberg wrote:
> Hi,
> One possible alternative solution for SDP multiplex negotiation could 
> be based on the assumption of using the same port number in multiple 
> SDP m- lines (yes, I know SDP does not allow it, and I will come back 
> to that).
> Something like:
> SDP offer:
> m=audio 10000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 10000 ...
> a=rtpmap ...
> a=rtpmap ...
> SDP answer (multiplex supported/accepted):
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 20000 ...
> a=rtpmap ...
> a=rtpmap ...
In this case, packets flow from/to port 10000 to port 20000 for both 
media types.
> SDP answer (multiplex not-supported/rejected):
> m=audio 20000 ...
> a=rtpmap ...
> a=rtpmap ...
> m=video 30000 ...
> a=rtpmap ...
> a=rtpmap ...
In this case, packets would presumably flow between port 10000 and port 
20000 for audio, and port 10000 and port 30000 for video.
Is that a correct interpretation of how you think this would work?

In that case, we have an RTP spec problem, not just an SDP problem: RTP 
"straight" claims to identify sessions by destination address + port 
(RFC 3550 section 5.2, for instance). In normal unicast / point-to-point 
usage, we expect all packets to come from the same address + port too, 
but RFC 3550 doesn't say that.

I don't know if this is an issue. If it is an issue, sender might have 
to start over.


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    On 08/30/11 13:21, Christer Holmberg wrote:
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05852233D64F47@ESESSCMS0356.eemea.ericsson.se"
      type="cite">
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        <div>Hi,</div>
        <div>&nbsp;</div>
        <div>One possible alternative solution for SDP multiplex
          negotiation could be based on the assumption of using the same
          port number in multiple SDP m- lines (yes, I know SDP does not
          allow it, and I will come back to that).</div>
        <div>&nbsp;</div>
        <div>Something like:</div>
        <div>&nbsp;</div>
        <div>SDP offer:</div>
        <div>&nbsp;</div>
        <div>m=audio 10000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
        <div>m=video 10000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
        <div>&nbsp;</div>
        <div>&nbsp;</div>
        <div>SDP answer (multiplex supported/accepted):</div>
        <div>&nbsp;</div>
        <div>m=audio 20000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
        <div>m=video 20000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
      </font></blockquote>
    In this case, packets flow from/to port 10000 to port 20000 for both
    media types.<br>
    <blockquote
cite="mid:7F2072F1E0DE894DA4B517B93C6A05852233D64F47@ESESSCMS0356.eemea.ericsson.se"
      type="cite"><font face="Arial, sans-serif" size="2">
        <div>&nbsp;</div>
        <div>&nbsp;</div>
        <div>SDP answer (multiplex not-supported/rejected):</div>
        <div>&nbsp;</div>
        <div>m=audio 20000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
        <div>m=video 30000 ...</div>
        <div>a=rtpmap ...</div>
        <div>a=rtpmap ...</div>
        <div>&nbsp;</div>
      </font></blockquote>
    In this case, packets would presumably flow between port 10000 and
    port 20000 for audio, and port 10000 and port 30000 for video.<br>
    Is that a correct interpretation of how you think this would work?<br>
    <br>
    In that case, we have an RTP spec problem, not just an SDP problem:
    RTP "straight" claims to identify sessions by destination address +
    port (RFC 3550 section 5.2, for instance). In normal unicast /
    point-to-point usage, we expect all packets to come from the same
    address + port too, but RFC 3550 doesn't say that.<br>
    <br>
    I don't know if this is an issue. If it is an issue, sender might
    have to start over.<br>
    <br>
  </body>
</html>

--------------080504000607090606040508--

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Harald,

This looks good. I just have two very minor comments regarding Section =
5:

- The correct reference for SRTP is RFC3711, not RFC3550 (the security =
mechanism in RFC3550 is not compatible with SRTP).

- I agree with the sentiment that "SRTP is used for transport of all =
real-time media", however this leaves open the choice of RTP profile =
(there are several that use SRTP). It might be useful to reference =
draft-perkins-rtcweb-rtp-usage, since that addresses this question.

Cheers,
Colin



On 24 Aug 2011, at 16:08, Harald Alvestrand wrote:
> A few drawings, some more definitions, and a link to the W3C API spec.
>=20
> If people are quick with comments, I can get another version done =
before our September 7 meeting; version numbers are cheap.
>=20
>                Harald
>=20
> -------- Original Message --------
> Subject:	New Version Notification for =
draft-ietf-rtcweb-overview-01.txt
> Date:	Wed, 24 Aug 2011 07:37:14 -0700
> From:	internet-drafts@ietf.org
> To:	harald@alvestrand.no
> CC:	harald@alvestrand.no
>=20
> A new version of I-D, draft-ietf-rtcweb-overview-01.txt has been =
successfully submitted by Harald Alvestrand and posted to the IETF =
repository.
>=20
> Filename:	 draft-ietf-rtcweb-overview
> Revision:	 01
> Title:		 Overview: Real Time Protocols for Brower-based =
Applications
> Creation date:	 2011-08-24
> WG ID:		 rtcweb
> Number of pages: 17
>=20
> Abstract:
>    This document gives an overview and context of a protocol suite
>    intended for use with real-time applications that can be deployed =
in
>    browsers - &quot;real time communication on the Web&quot;.
>=20
>    It intends to serve as a starting and coordination point to make =
sure
>    all the parts that are needed to achieve this goal are findable, =
and
>    that the parts that belong in the Internet protocol suite are fully
>    specified and on the right publication track.
>=20
>    This work is an attempt to synthesize the input of many people, but
>    makes no claims to fully represent the views of any of them.  All
>    parts of the document should be regarded as open for discussion,
>    unless the RTCWEB chairs have declared consensus on an item.
>=20
>    This document is a candidate to become a work item of the RTCWEB
>    working group.
>=20
>=20
>                                                                        =
          =20
>=20
>=20
> The IETF Secretariat
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




--Apple-Mail-80--743355380
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; =
">Harald,<div><br></div><div>This looks good. I just have two very minor =
comments regarding Section 5:</div><div><br></div><div>- The correct =
reference for SRTP is RFC3711, not RFC3550 (the security mechanism in =
RFC3550 is not compatible with SRTP).</div><div><br></div><div>- I agree =
with the sentiment that "SRTP is used for transport of all real-time =
media", however this leaves open the choice of RTP profile (there are =
several that use SRTP). It might be useful to reference =
draft-perkins-rtcweb-rtp-usage, since that addresses this =
question.</div><div><br></div><div>Cheers,</div><div>Colin</div><div><br><=
/div><div><br></div><div><br><div><div>On 24 Aug 2011, at 16:08, Harald =
Alvestrand wrote:</div><blockquote type=3D"cite">

 =20

    <meta http-equiv=3D"content-type" content=3D"text/html; =
charset=3DUTF-8">
 =20
  <div bgcolor=3D"#ffffff" text=3D"#000000">
    A few drawings, some more definitions, and a link to the W3C API
    spec.<br>
    <br>
    If people are quick with comments, I can get another version done
    before our September 7 meeting; version numbers are cheap.<br>
    <br>
    =
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; Harald<br>
    <br>
    -------- Original Message --------
    <table class=3D"moz-email-headers-table" border=3D"0" =
cellpadding=3D"0" cellspacing=3D"0">
      <tbody>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" =
valign=3D"BASELINE">Subject: </th>
          <td>New Version Notification for
            draft-ietf-rtcweb-overview-01.txt</td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">Date: =
</th>
          <td>Wed, 24 Aug 2011 07:37:14 -0700</td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">From: =
</th>
          <td><a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a></td>=

        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">To: =
</th>
          <td><a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
        </tr>
        <tr>
          <th align=3D"RIGHT" nowrap=3D"nowrap" valign=3D"BASELINE">CC: =
</th>
          <td><a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
        </tr>
      </tbody>
    </table>
    <br>
    <br>
    <pre>A new version of I-D, draft-ietf-rtcweb-overview-01.txt has =
been successfully submitted by Harald Alvestrand and posted to the IETF =
repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 01
Title:		 Overview: Real Time Protocols for Brower-based =
Applications
Creation date:	 2011-08-24
WG ID:		 rtcweb
Number of pages: 17

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - &amp;quot;real time communication on the Web&amp;quot;.

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a candidate to become a work item of the RTCWEB
   working group.


                                                                         =
        =20


The IETF Secretariat

</pre>
  </div>

_______________________________________________<br>rtcweb mailing =
list<br><a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>https://www.ietf.or=
g/mailman/listinfo/rtcweb<br></blockquote></div><br><div>
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Arial; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; orphans: 2; text-align: auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
'Lucida Sans Typewriter'; font-size: 9px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px; "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; -webkit-border-horizontal-spacing: =
0px; -webkit-border-vertical-spacing: 0px; color: rgb(0, 0, 0); =
font-family: 'Lucida Sans Typewriter'; font-size: 9px; font-style: =
normal; font-variant: normal; font-weight: normal; letter-spacing: =
normal; line-height: normal; -webkit-text-decorations-in-effect: none; =
text-indent: 0px; -webkit-text-size-adjust: auto; text-transform: none; =
orphans: 2; white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; =
font-size: 9px; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; =
-webkit-text-decorations-in-effect: none; text-indent: 0px; =
-webkit-text-size-adjust: auto; text-transform: none; orphans: 2; =
white-space: normal; widows: 2; word-spacing: 0px; "><div><br =
class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div></span></spa=
n></span><br class=3D"Apple-interchange-newline"></span><br =
class=3D"Apple-interchange-newline"></span>
</div>
<br></div></body></html>=

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Harald,

On 29 Aug 2011, at 14:33, Harald Alvestrand wrote:
> Colin, I really like this version!
>=20
> There might want to be a placeholder about multiplexing, saying "we =
will return to this issue after more discussion". Otherwise, it seems to =
me we're setting up the expectation that this will be handled entirely =
in some other document - I think the end product of the WG needs to =
discuss it, and it seems logical that some aspect of it should go here.

Yes, I'll add this.

> Some detailed comments:
>=20
> - In Figure 2, section 1.1, I think you're making the assumption that =
multi-unicast topologies will use a single shared RTP session (SSRC =
number space) for all the links. This is not obvious; it's possible to =
build this topology on point-to-point RTP sessions too.

It's possible, but not necessarily desirable. Building this using a =
single RTP session (with a shared SSRC space) makes debugging a lot =
easier, since you can do third-party debugging (e.g., you can see from =
the RTCP that Alice can hear Bob talking, but you can't, so you know =
there's a problem somewhere, and can alert the user). It also enables =
various other features, such as third-party retransmissions.=20

> - In section 3, you make the point that improper signalling of =
bandwidth can cause failure to interoperate (because of differing RTCP =
timings). Is there a (possibly theoretical) problem with interoperation =
between RTP/AVP and RTP/AVPF too, or have we verified that the AVPF =
profile always sends enough RTCP packets that AVP-conformant endpoints =
don't time out?

I'm not aware of any problems here. They were designed to interoperate.

> - In section 6, I would recommend removing the point about scenarios =
with mixers using point-to-point RTP sessions are "not well utilizing =
the mechanisms of RTP" - I think people's engineering tradeoffs should =
be respected.
> I'm fine with leaving in the comment about protocol violations =
(although I'd like to be more specific about what they are - protocols =
shouldn't be violated; if people "have" to do that, there's something =
wrong with the protocol).

The referenced sections of RFC 5117 list specific technical problems =
with these approaches. I agree that the wording could be improved, but I =
think that the recommendation is generally appropriate.

> - In the list of other extensions, section 6.2, you say that two =
extensions are "not recommended" - is this a recommendation against =
(like NOT RECOMMENDED would be), or simply a declaration that their use =
or non-use is of no concern?

My feeling is there's no clear benefit for RTCWeb from implementing =
these other extensions. The wording is perhaps a little strong, and =
could be weakened.

> - there are some sections with garbled grammar (the last paragraph of =
section 7.2, before 7.2.1, is particularly irksome to the eye), but I =
assume that this will be reviewed in later versions; the meaning comes =
through.


Yeah, I'll review it and try to improve things in a coming version.

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




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I'm happy to present if people think it's useful, although I'd guess =
most of the discussion will relate to multiplexing, which isn't in the =
latest version of my draft.

Colin


On 29 Aug 2011, at 10:45, Harald Alvestrand wrote:
> Given that draft-perkins-rtcweb-rtp-usage-03 is just out, should we =
include that on the agenda?
>=20
> On 08/27/11 01:09, Cullen Jennings wrote:
>> We need to start working on an agenda for the Sept 8th - bellow is  =
very rough straw man. Please suggest ideas for things to add, delete, =
change ...
>>=20
>>=20
>> --------------------
>>=20
>>=20
>> Use Case - John (60 min)
>>=20
>>     Identity in multiuser calls
>>=20
>>     Recording
> could you add a little more text here (in particular as John is a =
multicast address)?
> I think I know where "recording" comes from, but what is the =
"identity" point?
>>=20
>> Terminology Mapping RTP to W3C ( 30 min)
>>     - what's a track, session, flow, connection and how do they =
relate
> This is a discussion that's been happening on the W3C list, for those =
of you who don't read that.
>>=20
>> Just One Port - (AKA Multiplexing) -  Harald  (30 min)
>>=20
>>=20
>> Non Media Data Transport (30 min)
>>=20
>>=20
>> Security and User Interface (90 min)
>>      What do we expect the see and do in relation to security and =
authorization around access to microphone/camera
> This has previously been said to be a W3C discussion - and the W3C WG =
is having a meeting on Aug 31.
> Don't know if 90 minutes of IETF time is the Right Thing here.
>>=20
>> Other things that I don't see as being ready to spend time on ...
>>=20
>>=20
>> NAT
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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




From ted.ietf@gmail.com  Tue Aug 30 09:33:45 2011
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The chairs would like to call for adoption of
draft-perkins-rtcweb-rtp-usage as the basis of a WG draft; please send
comments to the list by September 7th, 2011.

Note that we expect it to be part of the document set meeting the
following charter item: "RTCWeb protocol profiles and Media format
specification(s)".

regards,

Ted, Cullen, Magnus

From bernard_aboba@hotmail.com  Tue Aug 30 10:11:18 2011
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I'm in favor of adopting it.=20

> Date: Tue=2C 30 Aug 2011 09:35:12 -0700
> From: ted.ietf@gmail.com
> To: rtcweb@ietf.org
> Subject: [rtcweb] Call for adoption as WG draft:	draft-perkins-rtcweb-rtp=
-usage
>=20
> The chairs would like to call for adoption of
> draft-perkins-rtcweb-rtp-usage as the basis of a WG draft=3B please send
> comments to the list by September 7th=2C 2011.
>=20
> Note that we expect it to be part of the document set meeting the
> following charter item: "RTCWeb protocol profiles and Media format
> specification(s)".
>=20
> regards=2C
>=20
> Ted=2C Cullen=2C Magnus
> _______________________________________________
> 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'>
I'm in favor of adopting it. <br><br><div>&gt=3B Date: Tue=2C 30 Aug 2011 0=
9:35:12 -0700<br>&gt=3B From: ted.ietf@gmail.com<br>&gt=3B To: rtcweb@ietf.=
org<br>&gt=3B Subject: [rtcweb] Call for adoption as WG draft:	draft-perkin=
s-rtcweb-rtp-usage<br>&gt=3B <br>&gt=3B The chairs would like to call for a=
doption of<br>&gt=3B draft-perkins-rtcweb-rtp-usage as the basis of a WG dr=
aft=3B please send<br>&gt=3B comments to the list by September 7th=2C 2011.=
<br>&gt=3B <br>&gt=3B Note that we expect it to be part of the document set=
 meeting the<br>&gt=3B following charter item: "RTCWeb protocol profiles an=
d Media format<br>&gt=3B specification(s)".<br>&gt=3B <br>&gt=3B regards=2C=
<br>&gt=3B <br>&gt=3B Ted=2C Cullen=2C Magnus<br>&gt=3B ___________________=
____________________________<br>&gt=3B rtcweb mailing list<br>&gt=3B rtcweb=
@ietf.org<br>&gt=3B https://www.ietf.org/mailman/listinfo/rtcweb<br></div> =
		 	   		  </div></body>
</html>=

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From randell-ietf@jesup.org  Tue Aug 30 10:32:43 2011
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On 8/30/2011 12:22 PM, Colin Perkins wrote:
>
>> - In section 3, you make the point that improper signalling of bandwidth can cause failure to interoperate (because of differing RTCP timings). Is there a (possibly theoretical) problem with interoperation between RTP/AVP and RTP/AVPF too, or have we verified that the AVPF profile always sends enough RTCP packets that AVP-conformant endpoints don't time out?
> I'm not aware of any problems here. They were designed to interoperate.

Agreed, this should be fine and has been done a lot.

>> - In section 6, I would recommend removing the point about scenarios with mixers using point-to-point RTP sessions are "not well utilizing the mechanisms of RTP" - I think people's engineering tradeoffs should be respected.
>> I'm fine with leaving in the comment about protocol violations (although I'd like to be more specific about what they are - protocols shouldn't be violated; if people "have" to do that, there's something wrong with the protocol).
Or they're handling a usecase that isn't covered by the protocol or wasn't anticipated.
That may not be something "wrong" with the protocol - they may be using the wrong
protocol, or they may need a new protocol, or they may need an extension to the protocol
to cover this new usecase.

If it's a "covered" usecase, then there's something wrong with the protocol.


-- 
Randell Jesup
randell-ietf@jesup.org


From randell-ietf@jesup.org  Tue Aug 30 10:39:15 2011
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On 8/30/2011 12:35 PM, Ted Hardie wrote:
> The chairs would like to call for adoption of
> draft-perkins-rtcweb-rtp-usage as the basis of a WG draft; please send
> comments to the list by September 7th, 2011.
Agree


-- 
Randell Jesup
randell-ietf@jesup.org


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Hi Harald,

>>One possible alternative solution for SDP multiplex negotiation could be =
based on the assumption of using the same port number in multiple SDP m- li=
nes (yes, I know SDP does not allow it, and I will come back to that).
>>
>>Something like:
>>
>>SDP offer:
>>
>>m=3Daudio 10000 ...
>>a=3Drtpmap ...
>>a=3Drtpmap ...
>m=3Dvideo 10000 ...
>a=3Drtpmap ...
>a=3Drtpmap ...
>
>
>SDP answer (multiplex supported/accepted):
>
>m=3Daudio 20000 ...
>a=3Drtpmap ...
>a=3Drtpmap ...
>m=3Dvideo 20000 ...
>a=3Drtpmap ...
>a=3Drtpmap ...
>
>In this case, packets flow from/to port 10000 to port 20000 for both media=
 types.

Yes.

(The question is then whether we need something in addition, e.g. something=
 like the TOGETHER grouping, to actually indicate that they are multiplexed=
. But, I guess that depends e.g. on the actual multiplex mechanism we'll us=
e.)


>SDP answer (multiplex not-supported/rejected):
>
>m=3Daudio 20000 ...
>a=3Drtpmap ...
>a=3Drtpmap ...
>m=3Dvideo 30000 ...
>a=3Drtpmap ...
>a=3Drtpmap ...
>
>In this case, packets would presumably flow between port 10000 and port 20=
000 for audio, and port 10000 and port 30000 for video.
>Is that a correct interpretation of how you think this would work?

Yes.

>In that case, we have an RTP spec problem, not just an SDP problem: RTP "s=
traight" claims to identify sessions by destination address + port (RFC 355=
0 section 5.2, for instance). In normal unicast / point-to-point usage, we =
expect all packets to come from the same=20
>address + port too, but RFC 3550 doesn't say that.
>
>I don't know if this is an issue. If it is an issue, sender might have to =
start over.

I am not sure I understood the issue.

However, I am not saying there aren't any issues - I just think we should a=
llow ourselves to look into different solution alternatives :)

Regards,

Christer=

From christer.holmberg@ericsson.com  Tue Aug 30 12:26:08 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Justin Uberti <juberti@google.com>
Date: Tue, 30 Aug 2011 21:27:33 +0200
Thread-Topic: [rtcweb] Multiplexing using the same port number for multiple media descritions
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Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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Hi Justin,

>I think Harald's approach is cleaner, since the fallback does not require =
a new offer.
>
>What do you see as problematic with Harald's suggestion?

I have sent comments on Harald's draft, but my main issues are:

1. Unclear how to remove the m- line which is used for the multiplexed stre=
am.

(Maybe Harald has indicated somewhere how it would be done, and in that cas=
e I appologise for having missed it)


2. Intermediaries that do not understand the extension would still think th=
at there are individual streams, and reserve resources etc accordingly.

(Now, maybe there aren't such intermediaries in a web environment. But as C=
LUE very likely will also need some kind of a multiplex negotiation mechani=
sm I would like to see something more general).

Regards,

Christer



On Tue, Aug 30, 2011 at 7:21 AM, Christer Holmberg <christer.holmberg@erics=
son.com<mailto:christer.holmberg@ericsson.com>> wrote:
Hi,

One possible alternative solution for SDP multiplex negotiation could be ba=
sed on the assumption of using the same port number in multiple SDP m- line=
s (yes, I know SDP does not allow it, and I will come back to that).

Something like:

SDP offer:

m=3Daudio 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...



MAYBE there is also a need to use some kind of grouping, in which case it c=
ould look something like (borrowing some terminology from Harald):



SDP offer:

a=3Dgroup:TOGETHER foo bar
m=3Daudio 10000 ...
a=3Dmid:foo
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
a=3Dmid:foo
m=3Dvideo 20000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...


An ISSUE with this solution is of course that SDP does not allow for it.

However, we could always say that browsers must support it, in which case i=
t should work fine in direct browser-to-browser cases.


When interworking with legacy, I guess two things can happen:

1. The offer is acctepted, with different port number in the answer, and mu=
ltiplex won't be used (see example above)

2. The offer is rejected. In this case, the fallback would be that the brow=
ser sends a new offer, with different port numbers, and multiplex won't be =
used.

Regards,

Christer




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



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Subject: Re: [rtcweb] Comments on use case draft
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On 08/30/2011 08:06 AM, Randell Jesup wrote:
> On 8/28/2011 4:07 PM, Stephan Wenger wrote:
>> 4. Section 4.2.6.1: This use case is not described in sufficient detail.
>> At least two scenarios are possible.  First, both front and rear camera
>> send individual video streams (potentially at different resolutions), 
>> and
>> the PIP mixing happens in the receiving browser.  This would be a user
>> interface issue and no mechanisms need to be specified in the IETF 
>> beyond
>> being bale to send more than one video stream (though there may be need
>> for related API work).  Second, the PIP mixing happens in the sending
>> phone, and only one stream is being sent.  In this case, I believe not
>> even API work is necessary.
>> Suggest to reconsider whether this use case is relevant enough for being
>> kept.  Multi-camera systems being able to send coded samples from both
>> cameras simultaneously are rather exotic today (only telepresence rooms
>> come to my mind).
> I think this directly talks to a common case that would interest many 
> news
> organizations: personal news reporting.  CNN iReport, even local news 
> channels -
> they love having users act as reporters-on-the spot, and having the 
> video from
> both cameras is a big win for them (and for the producers working from 
> such
> footage, who could swap the PIPs or suppress one or the other on the 
> fly).
> And most newer Android phones and iPhones have two cameras.
>
> Could I live with losing this use-case?  Yes, with pain.  I do want to 
> support
> multiple streams so you can have the user and a local video (either a 
> file, a
> camera, or a video encoding of a desktop or window).  I'll note this 
> usecase
> only hits the two-cameras part of what I'd like to see, but I don't know
> we need to go into that detail here (i.e. where other than a camera a 
> stream
> can come from is not something we need to mandate here - I think we 
> just need
> to call out the need for two streams).
>
>> 5. Section 4.2.7.1.: Why are the sending peers restricted to mono audio?
>> Spatial arrangement is not very complex for stereo as well...
>>
>> 6. Section 4.2.9.1.: What's really necessary here is a mechanism that
>> allows a user to tell a browser that VERY tight cross-signal sync and 
>> VERY
>> low delay is required, which may trigger different jitter buffer 
>> handling
>> and such.  Beyond that, I believe that audio codec negotiation may be
>> helpful.  Audio professionals (like musicians) are somewhat more picky
>> when it comes to these technology selections than normal users.  I would
>> not be surprised if we would learn that there is a real market 
>> requirement
>> for uncompressed or lossless audio if this use case takes off.
> Distributed music band - that needs TIGHT N-browser N-stream sync and
> VERY LOW (and virtually constant i.e. lan) delay.  I do not believe 
> this is
> likely to be technically feasible such that users would accept it.
>
> It would likely need ultra-low delay jitter buffers, much lower 
> packetization
> sizes, uncompressed audio, etc.
>
> Distribution of music to multiple playback stations: more possible; 
> the sync
> requirements are somewhat relaxed, and the ultra-low delay is relaxed 
> much more.
>
> Let's drop this one, please.
>
>
>>
>> Section 4.3.3.1: I have a number of issues with this use case.
>>
>> First, in contrast to most other use cases, this one enters solution 
>> space
>> quite prominently.  That wouldn't be an issue for me if the solution my
>> employer is favoring were mentioned here, but it is not :-(.  To cure my
>> immediate concern, one suggestion would be to remove references to
>> simulcast and/or add references to spatial scalability.  However, 
>> perhaps
>> it's better to describe the behavior of the multipoint system in 
>> terms of
>> user experience rather than technology choice.
> Generally I agree.  The use case can be more user-oriented.  What are 
> we targeting
> for the space here: a conference system tightly tied to an application 
> on the
> browser, or a generic conference system which would allow better 
> operation when
> you have "interop" calls between services - i.e. can someone on 
> Facebook join
> a rtcweb Hangout hosted on Google?  These choices would drive 
> different requirements
> to be derived from the usecase.
>
> Also, we're describing things an application *could* do with the 
> system, not the only
> or even preferred way to do a conference.  This usecase states that 
> someone *could*
> build a "dumb" conference server that does no re-encoding, just stream 
> selection and
> forwarding.  It doesn't prevent a conference server that re-encodes, 
> nor a conference
> system using SVC or equivalent that subsets the incoming stream.
>
> The application could request that a second smaller stream be sent, 
> though obviously
> this presumes the implementation, so it would be more tied to the 
> conference server
> implementation.  I'm wondering if there would be a good way to say 
> "find a way to
> deliver a low and high resolution image", and let the system (rtcweb) 
> figure out how
> to do it given the shared codecs available.  (I.e. SVC in a particular 
> config if both
> the conf server and browser support it, simulcast streams if they don't.)
>
>
>> Second, why is audio mixed to stereo and not to something else, such as
>> 5.1?
> Remove reference to the number of channels.  That's handled via 
> negotiation as
> normal.
>
>
>>
>> Third, the security stuff is not in any way technically bound to the 
>> rest
>> of the use case, so I would farm it out into its own use case, and/or
>> mention it as a "generic" feature... Remarks like "it is essential that
>> the communication cannot be eavesdropped" would apply to pretty much all
>> use cases, right?
> Per someone else's comment: we can drop this and add a separate 
> usecase for
> planning a bank robbery.  :-)  Better, though, might be a lawyer-client
> conversation or a secret agent. :-)


In regards to PIP, if A signs B's security certificate, and the media 
swaps, then there should exist A's security certificate signed by B. By 
yesterday's standard, that was not done automatically. The full duplex 
automation of such process is evidence as people's source of anxiety. 
Maybe that the generic feature for that, or to enable that process, is 
available on the "ribbon-end-point".

Can we make this toy-story stack more obvious? Stacked in this order, 
hypervisor(or grub), linux, X11, Xfce (GNOME/compiz optional), 
Mono/.NET, winelib/wine/winnt(co-op), Internet Browser, Win 7/8 File 
Manager. I think that is  clearer on exactly what is "the client" 
framework. Personnally, I want to "the client" to know schema-only, SSRC 
aware, with reflection built-into iNET, which I think is given (YMMV), 
yet not so obvious what is being accomplished with such stack overall. 
We can make wider use out of device driver certificates, under such 
stack, if we consider tablets (with "pen") as client-only.

Under RTP, I can imagine the pen certify itself with the "ink"'d media, 
and continue certification of any gesture recognition. Well, I have 
imagined this for dozen years now, yet realized my homework got done and 
turned-in... do municipal banks invest in student tablets? Yes! That is 
about what is left of municipalities, so now they like (or only can 
afford) the fire-n-forget "upgrade when you break it" mode loaded with 
academic licenses. Sound familiar?

I tried to get one grant written such that one local college had 
real-time DNS of local tablets and routers, so that homework and email 
could be sent around on campus with zero-config, but the procurement 
process stopped our non-faculty vote. Dummy me: "___ ate my pen."

You know why war-dialing started...  accessibilities and solutions clean 
of plagiarism.


>
>
>>
>> 7. Missing use case:
>>
>> It is my understanding that for regulatory compliance, in many developed
>> countries, there will be a need for an E911 type of service *IF* the
>> solution allows to "dial" an E.164 phone number.  I remember a 
>> controversy
>> involving SKYPE in ca. 2005, and also having read about recent FCC
>> hearings about this issue; for example,
>> http://www.broadbandlawadvisor.com/2011/07/articles/voip/fcc-seeks-comment- 
>>
>> on-extending-e911-rules-to-oneway-outboundonly-voip-improve-location-capabi 
>>
>> lity-of-inteconnected-voip/.
>> If there is a reasonable expectation that a webrtc service with outbound
>> dialing capability in E.164 number-space requires E911 handling, then it
>> does not make sense to stick our collective heads in the sand and ignore
>> the issue.  I believe there is such an expectation; surely during the
>> lifetime of an webrtc solution, but probably even during its introducer
>> phase.
> Agreed.  I've dealt with FCC rulings (and I'm sure other countries 
> will have
> similar and possibly even stricter requirements).  Basically, it will 
> be very
> likely that a provider will want to connect rtcweb to the PSTN (even 
> if through
> a gateway), and once that's done they'll need to support E911 services.
>
> I could even see future expansion of the technologies for emergency 
> calling;
> you're seeing this now with emergency centers looking to support text 
> messages.
> So you might have emergency centers with rtcweb support directly or via a
> translation/forwarding service (as voip is generally done today).  
> There is
> significant advantage to an emergency center to be able to support 
> video calls,
> for example (though obviously there would be issues surrounding that).
>
>> If E911 is relevant in this sense, then this issue needs to be addressed
>> in section 4.3.1, 4.3.2, and perhaps 4.2.5.
>> I understand that the editors did not address those use cases yet 
>> based on
>> (presumably) lack of consensus, but I fear that IETF consensus is not 
>> the
>> only relevant factor here.
>>
>> (I could mention "legal intercept" in the same context, but suggest to
>> focus on emergency calls first, because a) they are easier to handle, b)
>> more widely applicable, and c) generally agreed to be a useful thing, 
>> and
>> therefore not quite as politically loaded.)
>
> Yes.  :-)
>


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


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To clarify a bit further on what we're thinking for next week's
agenda, the chairs are currently thinking of this time split:

1 Hour:  Use case discussion (draft available now)

1 Hour:  Security (draft available end of the week)
--Focus of the discussion will be on what we want to provide as
assurances to different actors.

1 Hour: Signaling
--Focus of the discussion will be semantics & path (e.g Offer/Answer or
alternative?)

1/2 Hour: RTP concepts mapping to WebRTC API objects
--How  flows/streams/sessions map to tracks/connections/tracklists

1/2 Hour: Tentative slot for Congestion Control

Please send any agenda discussion to the as soon as possible, so
folks are prepping for the right things.

thanks,

Ted, Cullen, Magnus

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On 08/30/2011 03:02 PM, Ted Hardie wrote:
> To clarify a bit further on what we're thinking for next week's
> agenda, the chairs are currently thinking of this time split:
>
> 1 Hour:  Use case discussion (draft available now)
>
> 1 Hour:  Security (draft available end of the week)
> --Focus of the discussion will be on what we want to provide as
> assurances to different actors.
>
> 1 Hour: Signaling
> --Focus of the discussion will be semantics&  path (e.g Offer/Answer or
> alternative?)
>
> 1/2 Hour: RTP concepts mapping to WebRTC API objects
> --How  flows/streams/sessions map to tracks/connections/tracklists
>
> 1/2 Hour: Tentative slot for Congestion Control
>
> Please send any agenda discussion to the as soon as possible, so
> folks are prepping for the right things.
>    

We can't rely on STD-C library or anything out-of-scope of the stable 
toy-story of the Personal Companion Cube (PCC); that means we want any 
"congestional control" in ABI-only, not API, such that the word 
"control" is kept to the digital comprehension only. No more 
ban-hammer's needed by gTLDs because young scientists think STD-C is the 
only ABI comparison; systems that know the truth of quantum spin simply 
use physical simulation for needs and meta-physical needs. Yes, why 
should physical systems "control" meta-physical maps: earthquake? Proven.

In ray-cast systems, the client end doesn't know if the source is 
analogue or digital when not the original signal. Some read my UUID per 
cast with progressive optimization blogs, there is no API needed for 
this ABI with security assets such as this kind of atomic clock (with or 
without real peripherals). "Start with the solution."

Replication is possible in my books, so I'll pause on this evil.

> thanks,
>
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>    


-- 
--- http://twitter.com/Dzonatas_Sol ---
Web Development, Software Engineering
Ag-Biotech, Virtual Reality, Consultant


From harald@alvestrand.no  Wed Aug 31 00:39:52 2011
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Subject: Re: [rtcweb] Review of draft-perkins-rtcweb-usage-03 (Re: Fwd: New Version Notification for	draft-perkins-rtcweb-rtp-usage-03.txt)
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On 08/30/11 18:22, Colin Perkins wrote:
> Harald,
>
> On 29 Aug 2011, at 14:33, Harald Alvestrand wrote:
>> Colin, I really like this version!
>>
>> There might want to be a placeholder about multiplexing, saying "we will return to this issue after more discussion". Otherwise, it seems to me we're setting up the expectation that this will be handled entirely in some other document - I think the end product of the WG needs to discuss it, and it seems logical that some aspect of it should go here.
> Yes, I'll add this.
>
>> Some detailed comments:
>>
>> - In Figure 2, section 1.1, I think you're making the assumption that multi-unicast topologies will use a single shared RTP session (SSRC number space) for all the links. This is not obvious; it's possible to build this topology on point-to-point RTP sessions too.
> It's possible, but not necessarily desirable. Building this using a single RTP session (with a shared SSRC space) makes debugging a lot easier, since you can do third-party debugging (e.g., you can see from the RTCP that Alice can hear Bob talking, but you can't, so you know there's a problem somewhere, and can alert the user). It also enables various other features, such as third-party retransmissions.
It also means that all the sessions have to have a single bandwidth 
number, since otherwise you can't get a consistent RTCP send rate, for 
instance.

Speaking for some implementors of the WEBRTC API, it's also clear that 
there's a significant cost to implementing the multiway RTP session 
concept - both in code complexity and API complexity. I think this 
warrants more discussion, where all 3 outcomes should be on the table:

- We recommend doing multi-unicast with a shared RTP session only
- We recommend supporting both shared and split RTP sessions for 
multi-unicast
- We recommend doing multi-unicast with split RTP sessions only

We should probably do this in a new thread.
>> - In section 3, you make the point that improper signalling of bandwidth can cause failure to interoperate (because of differing RTCP timings). Is there a (possibly theoretical) problem with interoperation between RTP/AVP and RTP/AVPF too, or have we verified that the AVPF profile always sends enough RTCP packets that AVP-conformant endpoints don't time out?
> I'm not aware of any problems here. They were designed to interoperate.
Thanks for the reassurance!
>> - In section 6, I would recommend removing the point about scenarios with mixers using point-to-point RTP sessions are "not well utilizing the mechanisms of RTP" - I think people's engineering tradeoffs should be respected.
>> I'm fine with leaving in the comment about protocol violations (although I'd like to be more specific about what they are - protocols shouldn't be violated; if people "have" to do that, there's something wrong with the protocol).
> The referenced sections of RFC 5117 list specific technical problems with these approaches. I agree that the wording could be improved, but I think that the recommendation is generally appropriate.
This text, from section 3.6?

    1) Loop detection cannot be performed on the RTP level.  When
       carelessly connecting two misconfigured MCUs, a loop could be
       generated.

    2) There is no information about active media senders available in
       the RTP packet.  As this information is missing, receivers cannot
       use it.  It also deprives the client of information related to
       currently active senders in a machine-usable way, thus preventing
       clients from indicating currently active speakers in user
       interfaces, etc.

    Note that deployed MCUs (and endpoints) rely on signalling layer
    mechanisms for the identification of the contributing sources, for
    example, a SIP conferencing package [RFC4575].  This alleviates, to
    some extent, the aforementioned issues resulting from ignoring RTP's
    CSRC mechanism.

I have my opinions about these issues (I've mentioned them to you in 
another message).
There are scenarios where these issues are not problems. And no protocol 
violation is identified in this text.

I agree that the video-switching-MCU scenario described in RFC 5117 
section 3.5 has problems, because it chooses to neither be one nor the 
other; I have no problems with recommending against that. It's RFC 5117 
section 3.6 that is the one I think engineers should be free to use when 
they think the engineering tradeoffs are appropriate.
>> - In the list of other extensions, section 6.2, you say that two extensions are "not recommended" - is this a recommendation against (like NOT RECOMMENDED would be), or simply a declaration that their use or non-use is of no concern?
> My feeling is there's no clear benefit for RTCWeb from implementing these other extensions. The wording is perhaps a little strong, and could be weakened.
I don't have an opinion, and would be happy to see "not recommended" 
interpreted as "RTCWEB doesn't care whether you use them or not". A 
recommendation against them would need a justification.
>> - there are some sections with garbled grammar (the last paragraph of section 7.2, before 7.2.1, is particularly irksome to the eye), but I assume that this will be reviewed in later versions; the meaning comes through.
>
> Yeah, I'll review it and try to improve things in a coming version.
>
Thanks!



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On 08/30/11 17:39, Colin Perkins wrote:
> Harald,
>
> This looks good. I just have two very minor comments regarding Section 5:
>
> - The correct reference for SRTP is RFC3711, not RFC3550 (the security 
> mechanism in RFC3550 is not compatible with SRTP).
Oops! Will fix.
>
> - I agree with the sentiment that "SRTP is used for transport of all 
> real-time media", however this leaves open the choice of RTP profile 
> (there are several that use SRTP). It might be useful to reference 
> draft-perkins-rtcweb-rtp-usage, since that addresses this question.
Hopefully, draft-perkins-rtcweb-rtp-usage will be 
draft-ietf-rtcweb-rtp-usage soon. I'll put in the pointer.

> Cheers,
> Colin
>
>
>
> On 24 Aug 2011, at 16:08, Harald Alvestrand wrote:
>> A few drawings, some more definitions, and a link to the W3C API spec.
>>
>> If people are quick with comments, I can get another version done 
>> before our September 7 meeting; version numbers are cheap.
>>
>>                Harald
>>
>> -------- Original Message --------
>> Subject: 	New Version Notification for draft-ietf-rtcweb-overview-01.txt
>> Date: 	Wed, 24 Aug 2011 07:37:14 -0700
>> From: 	internet-drafts@ietf.org
>> To: 	harald@alvestrand.no
>> CC: 	harald@alvestrand.no
>>
>>
>>
>> A new version of I-D, draft-ietf-rtcweb-overview-01.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.
>>
>> Filename:	 draft-ietf-rtcweb-overview
>> Revision:	 01
>> Title:		 Overview: Real Time Protocols for Brower-based Applications
>> Creation date:	 2011-08-24
>> WG ID:		 rtcweb
>> Number of pages: 17
>>
>> Abstract:
>>     This document gives an overview and context of a protocol suite
>>     intended for use with real-time applications that can be deployed in
>>     browsers -&quot;real time communication on the Web&quot;.
>>
>>     It intends to serve as a starting and coordination point to make sure
>>     all the parts that are needed to achieve this goal are findable, and
>>     that the parts that belong in the Internet protocol suite are fully
>>     specified and on the right publication track.
>>
>>     This work is an attempt to synthesize the input of many people, but
>>     makes no claims to fully represent the views of any of them.  All
>>     parts of the document should be regarded as open for discussion,
>>     unless the RTCWEB chairs have declared consensus on an item.
>>
>>     This document is a candidate to become a work item of the RTCWEB
>>     working group.
>>
>>
>>
>>
>>
>> The IETF Secretariat
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> -- 
> Colin Perkins
> http://csperkins.org/
>
>
>


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    On 08/30/11 17:39, Colin Perkins wrote:
    <blockquote
      cite="mid:EC5069B1-1A33-4A69-AEA1-E62EFE177934@csperkins.org"
      type="cite">Harald,
      <div><br>
      </div>
      <div>This looks good. I just have two very minor comments
        regarding Section 5:</div>
      <div><br>
      </div>
      <div>- The correct reference for SRTP is RFC3711, not RFC3550 (the
        security mechanism in RFC3550 is not compatible with SRTP).</div>
    </blockquote>
    Oops! Will fix.<br>
    <blockquote
      cite="mid:EC5069B1-1A33-4A69-AEA1-E62EFE177934@csperkins.org"
      type="cite">
      <div><br>
      </div>
      <div>- I agree with the sentiment that "SRTP is used for transport
        of all real-time media", however this leaves open the choice of
        RTP profile (there are several that use SRTP). It might be
        useful to reference draft-perkins-rtcweb-rtp-usage, since that
        addresses this question.</div>
    </blockquote>
    Hopefully, draft-perkins-rtcweb-rtp-usage will be
    draft-ietf-rtcweb-rtp-usage soon. I'll put in the pointer.<br>
    <br>
    <blockquote
      cite="mid:EC5069B1-1A33-4A69-AEA1-E62EFE177934@csperkins.org"
      type="cite">
      <div>Cheers,</div>
      <div>Colin</div>
      <div><br>
      </div>
      <div><br>
      </div>
      <div><br>
        <div>
          <div>On 24 Aug 2011, at 16:08, Harald Alvestrand wrote:</div>
          <blockquote type="cite">
            <meta http-equiv="content-type" content="text/html;
              charset=ISO-8859-1">
            <div bgcolor="#ffffff" text="#000000"> A few drawings, some
              more definitions, and a link to the W3C API spec.<br>
              <br>
              If people are quick with comments, I can get another
              version done before our September 7 meeting; version
              numbers are cheap.<br>
              <br>
              &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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-ietf-rtcweb-overview-01.txt</td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">Date:
                    </th>
                    <td>Wed, 24 Aug 2011 07:37:14 -0700</td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">From:
                    </th>
                    <td><a moz-do-not-send="true"
                        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 moz-do-not-send="true"
                        class="moz-txt-link-abbreviated"
                        href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></td>
                  </tr>
                  <tr>
                    <th align="RIGHT" nowrap="nowrap" valign="BASELINE">CC:
                    </th>
                    <td><a moz-do-not-send="true"
                        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-ietf-rtcweb-overview-01.txt has been successfully submitted by Harald Alvestrand and posted to the IETF repository.

Filename:	 draft-ietf-rtcweb-overview
Revision:	 01
Title:		 Overview: Real Time Protocols for Brower-based Applications
Creation date:	 2011-08-24
WG ID:		 rtcweb
Number of pages: 17

Abstract:
   This document gives an overview and context of a protocol suite
   intended for use with real-time applications that can be deployed in
   browsers - &amp;quot;real time communication on the Web&amp;quot;.

   It intends to serve as a starting and coordination point to make sure
   all the parts that are needed to achieve this goal are findable, and
   that the parts that belong in the Internet protocol suite are fully
   specified and on the right publication track.

   This work is an attempt to synthesize the input of many people, but
   makes no claims to fully represent the views of any of them.  All
   parts of the document should be regarded as open for discussion,
   unless the RTCWEB chairs have declared consensus on an item.

   This document is a candidate to become a work item of the RTCWEB
   working group.


                                                                                  


The IETF Secretariat

</pre>
            </div>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
            <a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
          </blockquote>
        </div>
        <br>
        <div>
          <span class="Apple-style-span" style="border-collapse:
            separate; color: rgb(0, 0, 0); font-family: Arial;
            font-size: 9px; font-style: normal; font-variant: normal;
            font-weight: normal; letter-spacing: normal; line-height:
            normal; orphans: 2; text-indent: 0px; text-transform: none;
            white-space: normal; widows: 2; word-spacing: 0px;"><span
              class="Apple-style-span" style="border-collapse: separate;
              color: rgb(0, 0, 0); font-family: 'Lucida Sans
              Typewriter'; font-size: 9px; font-style: normal;
              font-variant: normal; font-weight: normal; letter-spacing:
              normal; line-height: normal; orphans: 2; text-indent: 0px;
              text-transform: none; white-space: normal; widows: 2;
              word-spacing: 0px;"><span class="Apple-style-span"
                style="border-collapse: separate; color: rgb(0, 0, 0);
                font-family: 'Lucida Sans Typewriter'; font-size: 9px;
                font-style: normal; font-variant: normal; font-weight:
                normal; letter-spacing: normal; line-height: normal;
                text-indent: 0px; text-transform: none; orphans: 2;
                white-space: normal; widows: 2; word-spacing: 0px;"><span
                  class="Apple-style-span" style="border-collapse:
                  separate; color: rgb(0, 0, 0); font-family: 'Lucida
                  Sans Typewriter'; font-size: 9px; font-style: normal;
                  font-variant: normal; font-weight: normal;
                  letter-spacing: normal; line-height: normal;
                  text-indent: 0px; text-transform: none; orphans: 2;
                  white-space: normal; widows: 2; word-spacing: 0px;"><span
                    class="Apple-style-span" style="border-collapse:
                    separate; color: rgb(0, 0, 0); font-family: 'Lucida
                    Sans Typewriter'; font-size: 9px; font-style:
                    normal; font-variant: normal; font-weight: normal;
                    letter-spacing: normal; line-height: normal;
                    text-indent: 0px; text-transform: none; orphans: 2;
                    white-space: normal; widows: 2; word-spacing: 0px;">
                    <div><br class="Apple-interchange-newline">
                      <br class="khtml-block-placeholder">
                    </div>
                    <div>--&nbsp;</div>
                    <div>Colin Perkins</div>
                    <div><a moz-do-not-send="true"
                        href="http://csperkins.org/">http://csperkins.org/</a></div>
                  </span></span></span><br
                class="Apple-interchange-newline">
            </span><br class="Apple-interchange-newline">
          </span>
        </div>
        <br>
      </div>
    </blockquote>
    <br>
  </body>
</html>

--------------010208020803030008030401--

From harald@alvestrand.no  Wed Aug 31 03:28:43 2011
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Subject: Re: [rtcweb] Call for adoption as WG draft:	draft-perkins-rtcweb-rtp-usage
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I support this adoption.

On 08/30/11 18:35, Ted Hardie wrote:
> The chairs would like to call for adoption of
> draft-perkins-rtcweb-rtp-usage as the basis of a WG draft; please send
> comments to the list by September 7th, 2011.
>
> Note that we expect it to be part of the document set meeting the
> following charter item: "RTCWeb protocol profiles and Media format
> specification(s)".
>
> regards,
>
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From magnus.westerlund@ericsson.com  Wed Aug 31 04:13:44 2011
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Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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On 2011-08-30 17:15, Harald Alvestrand wrote:
> On 08/30/11 13:21, Christer Holmberg wrote:

> In that case, we have an RTP spec problem, not just an SDP problem: RTP
> "straight" claims to identify sessions by destination address + port
> (RFC 3550 section 5.2, for instance). In normal unicast / point-to-point
> usage, we expect all packets to come from the same address + port too,
> but RFC 3550 doesn't say that.
> 
> I don't know if this is an issue. If it is an issue, sender might have
> to start over.
> 

Well, that is one way to id the session a flow is supposed to end up in.
 If I understand the situation you are proposing would still allow the
inviting party to use 5-tuples to identify flows coming into each
session. This as you will learn from the ICE processing each peers
visible source address for each session through the ICE plus signaling.

And I don't think we have a spec problem if we like to use a session
transport setup that looks like this.

Session 1
A:10000 <---> B:20000

Session 2
A:10000 <---> B:30000

As this provides different 5-tuples it does fulfill for example the
following from section 5.2 of RFC3550:

   For example, in a teleconference
   composed of audio and video media encoded separately, each medium
   SHOULD be carried in a separate RTP session with its own destination
   transport address.

At least if you interpret "own destination transport address" as the
full five tuple rather than a 3-tuple of destiantion address, port and
protocol.

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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Subject: Re: [rtcweb] Call for adoption as WG draft:	draft-perkins-rtcweb-rtp-usage
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I support this work.

=D0=9D=D0=B0 30.08.11 19:35, Ted Hardie =D0=BD=D0=B0=D0=BF=D0=B8=D1=81=D0=
=B0:
> The chairs would like to call for adoption of
> draft-perkins-rtcweb-rtp-usage as the basis of a WG draft; please send
> comments to the list by September 7th, 2011.
>=20
> Note that we expect it to be part of the document set meeting the
> following charter item: "RTCWeb protocol profiles and Media format
> specification(s)".
>=20
> regards,
>=20
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20

--=20
http://jitsi.org


From christer.holmberg@ericsson.com  Wed Aug 31 06:34:29 2011
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Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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Hi,

Another advantage of the same-port-number solution is that the browser does=
n't need to reserve more than one port (including candidate collection etc)=
 - no matter whether multiplexing will eventually be used or not.

Regards,

Christer

________________________________
From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On Behalf Of=
 Christer Holmberg
Sent: 30. elokuuta 2011 14:22
To: rtcweb@ietf.org
Subject: [rtcweb] Multiplexing using the same port number for multiple medi=
a descritions

Hi,

One possible alternative solution for SDP multiplex negotiation could be ba=
sed on the assumption of using the same port number in multiple SDP m- line=
s (yes, I know SDP does not allow it, and I will come back to that).

Something like:

SDP offer:

m=3Daudio 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...



MAYBE there is also a need to use some kind of grouping, in which case it c=
ould look something like (borrowing some terminology from Harald):



SDP offer:

a=3Dgroup:TOGETHER foo bar
m=3Daudio 10000 ...
a=3Dmid:foo
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 10000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex supported/accepted):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
a=3Dmid:foo
m=3Dvideo 20000 ...
a=3Dmid:bar
a=3Drtpmap ...
a=3Drtpmap ...


SDP answer (multiplex not-supported/rejected):

m=3Daudio 20000 ...
a=3Drtpmap ...
a=3Drtpmap ...
m=3Dvideo 30000 ...
a=3Drtpmap ...
a=3Drtpmap ...


An ISSUE with this solution is of course that SDP does not allow for it.

However, we could always say that browsers must support it, in which case i=
t should work fine in direct browser-to-browser cases.


When interworking with legacy, I guess two things can happen:

1. The offer is acctepted, with different port number in the answer, and mu=
ltiplex won't be used (see example above)

2. The offer is rejected. In this case, the fallback would be that the brow=
ser sends a new offer, with different port numbers, and multiplex won't be =
used.

Regards,

Christer




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<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Hi,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Another advantage of the same-port-number solution=
 is that=20
the browser doesn't need to reserve more than&nbsp;one port (including cand=
idate=20
collection etc)&nbsp;- no matter whether multiplexing will eventually be us=
ed or=20
not.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D686253213-31082011><FONT face=3DA=
rial=20
color=3D#0000ff size=3D2>Christer</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px soli=
d; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> rtcweb-bounces@ietf.org=20
  [mailto:rtcweb-bounces@ietf.org] <B>On Behalf Of </B>Christer=20
  Holmberg<BR><B>Sent:</B> 30. elokuuta 2011 14:22<BR><B>To:</B>=20
  rtcweb@ietf.org<BR><B>Subject:</B> [rtcweb] Multiplexing using the same p=
ort=20
  number for multiple media descritions<BR></FONT><BR></DIV>
  <DIV></DIV><FONT face=3D"Arial, sans-serif" size=3D2>
  <DIV>Hi,</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>One possible alternative solution for SDP multiplex negotiation coul=
d be=20
  based on the assumption of using the same port number in multiple SDP m- =
lines=20
  (yes, I know SDP does not allow it, and I will come back to that).</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Something like:</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP offer:</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>m=3Daudio 10000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>m=3Dvideo 10000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP answer (multiplex supported/accepted):</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>m=3Daudio 20000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>m=3Dvideo 20000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP answer (multiplex not-supported/rejected):</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>m=3Daudio 20000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>m=3Dvideo 30000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>MAYBE there is also a need to use some kind of grouping, in which ca=
se it=20
  could look something like (borrowing some terminology from Harald):</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP offer:</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>a=3Dgroup:TOGETHER foo bar</DIV>
  <DIV>m=3Daudio 10000 ...</DIV>
  <DIV>a=3Dmid:foo</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>m=3Dvideo 10000 ...</DIV>
  <DIV>a=3Dmid:bar</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP answer (multiplex supported/accepted):</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>m=3Daudio 20000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Dmid:foo</DIV>
  <DIV>m=3Dvideo 20000 ...</DIV>
  <DIV>a=3Dmid:bar</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>SDP answer (multiplex not-supported/rejected):</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>m=3Daudio 20000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>m=3Dvideo 30000 ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>a=3Drtpmap ...</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>An ISSUE with this solution is of course that SDP does not allow for=
=20
  it.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>However, we could always say that browsers must support it, in which=
 case=20
  it should work fine in direct browser-to-browser cases.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>When interworking with legacy, I guess two things can happen:</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>1. The offer is acctepted, with different port number in the answer,=
 and=20
  multiplex won't be used (see example above)</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>2. The offer is rejected. In this case, the fallback would be that t=
he=20
  browser sends a new offer, with different port numbers, and multiplex won=
't be=20
  used.</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Regards,</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>Christer</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV>
  <DIV>&nbsp;</DIV></BLOCKQUOTE></FONT></BODY></HTML>

--_000_7F2072F1E0DE894DA4B517B93C6A05852233DD0669ESESSCMS0356e_--

From harald@alvestrand.no  Wed Aug 31 07:54:29 2011
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Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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On 08/30/11 21:27, Christer Holmberg wrote:
> Hi Justin,
>
>> I think Harald's approach is cleaner, since the fallback does not require a new offer.
>>
>> What do you see as problematic with Harald's suggestion?
> I have sent comments on Harald's draft, but my main issues are:
>
> 1. Unclear how to remove the m- line which is used for the multiplexed stream.
>
> (Maybe Harald has indicated somewhere how it would be done, and in that case I appologise for having missed it)
>
>
> 2. Intermediaries that do not understand the extension would still think that there are individual streams, and reserve resources etc accordingly.
Would those intermediaries understand the double usage in your proposal?
> (Now, maybe there aren't such intermediaries in a web environment. But as CLUE very likely will also need some kind of a multiplex negotiation mechanism I would like to see something more general).


From christer.holmberg@ericsson.com  Wed Aug 31 10:45:03 2011
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Date: Wed, 31 Aug 2011 19:43:16 +0200
Thread-Topic: [rtcweb] Multiplexing using the same port number for multiple media descritions
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Subject: Re: [rtcweb] Multiplexing using the same port number for multiple media descritions
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Hi Harald,

>>> I think Harald's approach is cleaner, since the fallback does not requi=
re a new offer.
>>>
>>> What do you see as problematic with Harald's suggestion?
>> I have sent comments on Harald's draft, but my main issues are:
>>
>> 1. Unclear how to remove the m- line which is used for the multiplexed s=
tream.
>>
>> (Maybe Harald has indicated somewhere how it would be done, and in that =
case I appologise for having >>missed it)
>>
>>
>> 2. Intermediaries that do not understand the extension would still think=
 that there are individual streams, and >>reserve resources etc accordingly=
.
>Would those intermediaries understand the double usage in your proposal?

The currently existing products and deployments that I am aware of would re=
serve different local ports for each m- line. They don't care about that fa=
ct that identical remote ports are used. So, there would be no multiplexing=
, but at least things would work - the correct amount of resources etc woul=
d be reserved.

Regards,

Christer=

From csp@csperkins.org  Wed Aug 31 13:46:38 2011
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Subject: Re: [rtcweb] Review of draft-perkins-rtcweb-usage-03 (Re: Fwd: New Version Notification for	draft-perkins-rtcweb-rtp-usage-03.txt)
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On 31 Aug 2011, at 08:41, Harald Alvestrand wrote:
> On 08/30/11 18:22, Colin Perkins wrote:
>> On 29 Aug 2011, at 14:33, Harald Alvestrand wrote:
>>> Colin, I really like this version!
>>>=20
>>> There might want to be a placeholder about multiplexing, saying "we =
will return to this issue after more discussion". Otherwise, it seems to =
me we're setting up the expectation that this will be handled entirely =
in some other document - I think the end product of the WG needs to =
discuss it, and it seems logical that some aspect of it should go here.
>> Yes, I'll add this.
>>=20
>>> Some detailed comments:
>>>=20
>>> - In Figure 2, section 1.1, I think you're making the assumption =
that multi-unicast topologies will use a single shared RTP session (SSRC =
number space) for all the links. This is not obvious; it's possible to =
build this topology on point-to-point RTP sessions too.
>> It's possible, but not necessarily desirable. Building this using a =
single RTP session (with a shared SSRC space) makes debugging a lot =
easier, since you can do third-party debugging (e.g., you can see from =
the RTCP that Alice can hear Bob talking, but you can't, so you know =
there's a problem somewhere, and can alert the user). It also enables =
various other features, such as third-party retransmissions.
> It also means that all the sessions have to have a single bandwidth =
number, since otherwise you can't get a consistent RTCP send rate, for =
instance.

True (although this has already been somewhat broken by the decision to =
multiplex audio and video in the same RTP session).

> Speaking for some implementors of the WEBRTC API, it's also clear that =
there's a significant cost to implementing the multiway RTP session =
concept - both in code complexity and API complexity. I think this =
warrants more discussion, where all 3 outcomes should be on the table:
>=20
> - We recommend doing multi-unicast with a shared RTP session only
> - We recommend supporting both shared and split RTP sessions for =
multi-unicast
> - We recommend doing multi-unicast with split RTP sessions only
>=20
> We should probably do this in a new thread.

Indeed; I'd be interested to know what complexity you're seeing. I would =
have expected the shared RTP session model to decrease complexity, since =
the RTP layer can be oblivious to the detail of the underlying transport =
connections, and wouldn't have to worry about tracking SSRC and payload =
type mapping separately for each session.

...
>>> - In section 6, I would recommend removing the point about scenarios =
with mixers using point-to-point RTP sessions are "not well utilizing =
the mechanisms of RTP" - I think people's engineering tradeoffs should =
be respected.
>>> I'm fine with leaving in the comment about protocol violations =
(although I'd like to be more specific about what they are - protocols =
shouldn't be violated; if people "have" to do that, there's something =
wrong with the protocol).
>> The referenced sections of RFC 5117 list specific technical problems =
with these approaches. I agree that the wording could be improved, but I =
think that the recommendation is generally appropriate.
> This text, from section 3.6?
>=20
>   1) Loop detection cannot be performed on the RTP level.  When
>      carelessly connecting two misconfigured MCUs, a loop could be
>      generated.
>=20
>   2) There is no information about active media senders available in
>      the RTP packet.  As this information is missing, receivers cannot
>      use it.  It also deprives the client of information related to
>      currently active senders in a machine-usable way, thus preventing
>      clients from indicating currently active speakers in user
>      interfaces, etc.
>=20
>   Note that deployed MCUs (and endpoints) rely on signalling layer
>   mechanisms for the identification of the contributing sources, for
>   example, a SIP conferencing package [RFC4575].  This alleviates, to
>   some extent, the aforementioned issues resulting from ignoring RTP's
>   CSRC mechanism.

And in RFC 5117 section 3.5, which says:

   The video switching MCU has most of the attributes of a Translator.
   However, its stream selection is a mixing behavior.  This behavior
   has some RTP and RTCP issues associated with it.  The suppression of
   all but one media stream results in most participants seeing only a
   subset of the sent media streams at any given time, often a single
   stream per conference.  Therefore, RTCP Receiver Reports only report
   on these streams.  Consequently, the media senders that are not
   currently forwarded receive a view of the session that indicates
   their media streams disappear somewhere en route.  This makes the use
   of RTCP for congestion control, or any type of quality reporting,
   very problematic.

   To avoid the aforementioned issues, the MCU needs to implement two
   features.  First, it needs to act as a Mixer (see Section 3.4) and
   forward the selected media stream under its own SSRC and with the
   appropriate CSRC values.  Second, the MCU needs to modify the RTCP
   RRs it forwards between the domains.  As a result, it is RECOMMENDED
   that one implement a centralized video switching conference using a
   Mixer according to RFC 3550, instead of the shortcut implementation
   described here.


> I have my opinions about these issues (I've mentioned them to you in =
another message).
> There are scenarios where these issues are not problems. And no =
protocol violation is identified in this text.
>=20
> I agree that the video-switching-MCU scenario described in RFC 5117 =
section 3.5 has problems, because it chooses to neither be one nor the =
other; I have no problems with recommending against that. It's RFC 5117 =
section 3.6 that is the one I think engineers should be free to use when =
they think the engineering tradeoffs are appropriate.

I agree that there are scenarios where these issues are not problematic, =
but there are also cases where they are. We believe the RFC 3550 mixer =
and translator models are better, and not more complex to implement or =
signal, which is why our draft recommends them.=20

>>> - In the list of other extensions, section 6.2, you say that two =
extensions are "not recommended" - is this a recommendation against =
(like NOT RECOMMENDED would be), or simply a declaration that their use =
or non-use is of no concern?
>> My feeling is there's no clear benefit for RTCWeb from implementing =
these other extensions. The wording is perhaps a little strong, and =
could be weakened.
> I don't have an opinion, and would be happy to see "not recommended" =
interpreted as "RTCWEB doesn't care whether you use them or not". A =
recommendation against them would need a justification.

Sure.

>>> - there are some sections with garbled grammar (the last paragraph =
of section 7.2, before 7.2.1, is particularly irksome to the eye), but I =
assume that this will be reviewed in later versions; the meaning comes =
through.
>>=20
>> Yeah, I'll review it and try to improve things in a coming version.
>>=20
> Thanks!


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




From ron.even.tlv@gmail.com  Wed Aug 31 14:31:49 2011
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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Ted Hardie'" <ted.ietf@gmail.com>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Call for adoption as WG draft:	draft-perkins-rtcweb-rtp-usage
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I am in favor for adopting this draft
Roni Even

> -----Original Message-----
> From: rtcweb-bounces@ietf.org [mailto:rtcweb-bounces@ietf.org] On
> Behalf Of Ted Hardie
> Sent: Tuesday, August 30, 2011 7:35 PM
> To: rtcweb@ietf.org
> Subject: [rtcweb] Call for adoption as WG draft: draft-perkins-rtcweb-
> rtp-usage
> 
> The chairs would like to call for adoption of
> draft-perkins-rtcweb-rtp-usage as the basis of a WG draft; please send
> comments to the list by September 7th, 2011.
> 
> Note that we expect it to be part of the document set meeting the
> following charter item: "RTCWeb protocol profiles and Media format
> specification(s)".
> 
> regards,
> 
> Ted, Cullen, Magnus
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

