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References: <4F646ACC.6030905@ericsson.com>, <4F64F466.90908@alum.mit.edu> <CD5674C3CD99574EBA7432465FC13C1B22726A0985@DC-US1MBEX4.global.avaya.com> <AEA158B0C52AEC4394D7B68A331367F46C8BF92D03@EUSAACMS0703.eamcs.ericsson.se>
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From: Dean Willis <dean.willis@softarmor.com>
Date: Mon, 2 Apr 2012 12:02:40 -0500
To: Nancy Greene <nancy.greene@ericsson.com>
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Cc: "rai@ietf.org" <rai@ietf.org>
Subject: Re: [RAI] Review of 6 new SIP header fields defined by OMA
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On Mar 19, 2012, at 8:20 PM, Nancy Greene wrote:

> Hi, here is some more information on these headers and why the =
suggested alternatives were not used:
>=20
>> From: Paul Kyzivat [pkyzivat@alum.mit.edu]
> ...
>> OMA Header                    Possible Alternative
>> --------------------------    -------------------------------------
>> Conversation-ID               Message-ID of 1st message
> There was a requirement in the OMA CPM work for more than just =
messages to be grouped together. It was meant to also cover file =
transfers and chat sessions, for example. So any standalone message, =
chat session or file transfer that was all related to the same =
"conversation" would have the same conversation-id.
>=20
>> Contribution-ID               Message-ID
> Again, it has to apply to more than messages - it has to apply to chat =
sessions and file transfers too.
>=20
>>=20

So the same Conversation-ID and Contribution-ID can occur in multiple =
INVITE-initiated dialogs? In other words, they identify logical =
sequences of communications that span more than one call? This would =
clearly differentiate them from Call-ID.

If it hasn't already been done, you may wish to have OMA consider US =
patent 7,983,240 within the OMA IPR context. You might also want to =
consider US patent application 11/646,982.

--
Dean=

From fluffy@cisco.com  Mon Apr 23 10:29:07 2012
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From: Cullen Jennings <fluffy@cisco.com>
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Subject: Re: [RAI] How to exchange metadata about the media streams established with a SIP session
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worth nothing the RTCweb folks also want to exchange metadata to do with =
labeling streams. I always find just classifying it as "metadata" sort =
of makes it hard to decide what to do with it. I'd prefer the groups to =
specifically outline what data they need to exchange and then I think it =
becomes clearer the best way to exchange the different types of =
information. Knowing if the information changes per RTP packet, or only =
one when the session is set up helps.


On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:

> Folks,
>=20
> I am sending this email to the RAI list because it is about an
> architectural issue that affects more than one WG.
>=20
> Both the CLUE and SIPREC WGs have identified the need to exchange
> metadata about media streams that have been established using SIP. =
Both
> groups need to transport such metadata between SIP UAs.
>=20
> In CLUE, they are studying how to transport their metadata, as
> documented in:
> http://tools.ietf.org/html/draft-wenger-clue-transport-01
>=20
> In SIPREC, they are planning to piggyback the metadata in SIP UPDATEs =
as
> an XML-encoded body part. When they need to send a request for full
> state (as opposed to partial state), they use an UPDATE request with a
> particular body type, as documented in:
> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
>=20
> The best way to exchange metadata within a SIP session depends on the
> type of metadata and on how the UAs need to interact with each other
> (e.g., a UA simply pushing information to the other UA or a more
> complicated protocol). Given that SIPREC and CLUE have different
> metadata and interactions, it may well be that we decide to use a
> different mechanism in each group. Nevertheless, I think both groups
> would benefit from architecturally-inclined RAI people having a look =
at
> their work. Please, continue this discussion focusing on their =
specific
> requirements on either the CLUE list or the SIPREC list.
>=20
> Somewhat related to this issue, the RTCWeb group discussed the use of
> SCTP over UDP as a reliable NAT-friendly way to exchange data between
> endpoints.
>=20
> Cheers,
>=20
> Gonzalo
>=20
> _______________________________________________
> RAI mailing list
> RAI@ietf.org
> https://www.ietf.org/mailman/listinfo/rai


From pkyzivat@alum.mit.edu  Mon Apr 23 12:22:46 2012
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Subject: Re: [RAI] How to exchange metadata about the media streams established with a SIP session
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On 4/23/12 10:53 AM, Cullen Jennings wrote:
>
> worth nothing the RTCweb folks also want to exchange metadata to do with labeling streams. I always find just classifying it as "metadata" sort of makes it hard to decide what to do with it. I'd prefer the groups to specifically outline what data they need to exchange and then I think it becomes clearer the best way to exchange the different types of information. Knowing if the information changes per RTP packet, or only one when the session is set up helps.

Cullen,

I agree that just talking about "metadata" isn't especially helpful.
OTOH, getting too far into the weeds about the specifics of the metadata 
makes it hard for people with differing contexts to talk with one 
another and find commonalities.

In CLUE we realize that not all of our metadata can be treated the same 
way. There are varying degrees of real-time-ed-ness to deal with. We 
definitely have some stuff that changes at the signaling rate or less, 
and some that changes at the media rate, and some stuff that that falls 
in between. We also have issues that some of the stuff is too big to 
reasonably transmit in sip signaling, which we are considering a 
separate "data stream" for. We do think that CLUE should be able to work 
with RTCWEB based endpoints. So its possible that the RTCWEB data stream 
mechanism might work for us. But we will need to have it even with 
non-rtcweb-endpoints.

In SIPREC, we have a lot of descriptive stuff, mostly about signaling. 
The rate is tied to the signaling rate, but possibly about multiple 
calls and hence higher in aggregate. We decided that sending in the sip 
signaling was appropriate for this stuff, though the size of the data 
precludes doing so over UDP. For this app we decided we could just 
require TCP transport for the sip signaling of the recording session. 
(It doesn't constrain the transport for the calls being recorded.) This 
could be a problem in some topologies, but seems acceptable for the 
application. If there had been a mechanism that for a separate "data 
stream" that we could have reused we *might* have found that an 
attractive direction.

ISTM that it would be helpful to have a "brainstorming session" with 
some reps from rtcweb, clue, siprec, bfcp, and mediactrl to see whether 
we can find some common machinery that could support multiples of these 
cases. I'm thinking that the sctp over udp stack being worked on for 
rtcweb might be more broadly applicable, both as a data stream transport 
negotiated with o/a, and maybe as a new sip transport as well.

	Thanks,
	Paul

> On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:
>
>> Folks,
>>
>> I am sending this email to the RAI list because it is about an
>> architectural issue that affects more than one WG.
>>
>> Both the CLUE and SIPREC WGs have identified the need to exchange
>> metadata about media streams that have been established using SIP. Both
>> groups need to transport such metadata between SIP UAs.
>>
>> In CLUE, they are studying how to transport their metadata, as
>> documented in:
>> http://tools.ietf.org/html/draft-wenger-clue-transport-01
>>
>> In SIPREC, they are planning to piggyback the metadata in SIP UPDATEs as
>> an XML-encoded body part. When they need to send a request for full
>> state (as opposed to partial state), they use an UPDATE request with a
>> particular body type, as documented in:
>> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
>>
>> The best way to exchange metadata within a SIP session depends on the
>> type of metadata and on how the UAs need to interact with each other
>> (e.g., a UA simply pushing information to the other UA or a more
>> complicated protocol). Given that SIPREC and CLUE have different
>> metadata and interactions, it may well be that we decide to use a
>> different mechanism in each group. Nevertheless, I think both groups
>> would benefit from architecturally-inclined RAI people having a look at
>> their work. Please, continue this discussion focusing on their specific
>> requirements on either the CLUE list or the SIPREC list.
>>
>> Somewhat related to this issue, the RTCWeb group discussed the use of
>> SCTP over UDP as a reliable NAT-friendly way to exchange data between
>> endpoints.
>>
>> Cheers,
>>
>> Gonzalo
>>
>> _______________________________________________
>> RAI mailing list
>> RAI@ietf.org
>> https://www.ietf.org/mailman/listinfo/rai
>
> _______________________________________________
> RAI mailing list
> RAI@ietf.org
> https://www.ietf.org/mailman/listinfo/rai
>


From andrew.hutton@siemens-enterprise.com  Tue Apr 24 01:35:29 2012
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To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rai@ietf.org" <rai@ietf.org>
Date: Tue, 24 Apr 2012 10:35:24 +0200
Thread-Topic: [RAI] How to exchange metadata about the media streams established with a SIP session
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Hi,

Whilst I of course agree that it is a good idea to look for a common soluti=
on if at all possible it is important to keep in mind the charter of the va=
rious working groups and the environment which the solutions will operate.

Both the SIPREC and CLUE working groups are chartered to produce SIP based =
solutions which is possibly open to some different interpretations but it d=
oes impose some constraints on the solution. In the SIPREC case we therefor=
e defined a SIP body based mechanism for metadata and stated that other mec=
hanism although possible were out of scope I have no problem with looking a=
t other mechanisms and would participate in any brainstorming session but I=
 don't think we should change track regarding the current SIPREC mechanism =
especially since there are quite a few implementations already.

Regards
Andy




> -----Original Message-----
> From: rai-bounces@ietf.org [mailto:rai-bounces@ietf.org] On Behalf Of
> Paul Kyzivat
> Sent: 23 April 2012 20:23
> To: rai@ietf.org
> Subject: Re: [RAI] How to exchange metadata about the media streams
> established with a SIP session
>=20
> On 4/23/12 10:53 AM, Cullen Jennings wrote:
> >
> > worth nothing the RTCweb folks also want to exchange metadata to do
> with labeling streams. I always find just classifying it as "metadata"
> sort of makes it hard to decide what to do with it. I'd prefer the
> groups to specifically outline what data they need to exchange and then
> I think it becomes clearer the best way to exchange the different types
> of information. Knowing if the information changes per RTP packet, or
> only one when the session is set up helps.
>=20
> Cullen,
>=20
> I agree that just talking about "metadata" isn't especially helpful.
> OTOH, getting too far into the weeds about the specifics of the
> metadata
> makes it hard for people with differing contexts to talk with one
> another and find commonalities.
>=20
> In CLUE we realize that not all of our metadata can be treated the same
> way. There are varying degrees of real-time-ed-ness to deal with. We
> definitely have some stuff that changes at the signaling rate or less,
> and some that changes at the media rate, and some stuff that that falls
> in between. We also have issues that some of the stuff is too big to
> reasonably transmit in sip signaling, which we are considering a
> separate "data stream" for. We do think that CLUE should be able to
> work
> with RTCWEB based endpoints. So its possible that the RTCWEB data
> stream
> mechanism might work for us. But we will need to have it even with
> non-rtcweb-endpoints.
>=20
> In SIPREC, we have a lot of descriptive stuff, mostly about signaling.
> The rate is tied to the signaling rate, but possibly about multiple
> calls and hence higher in aggregate. We decided that sending in the sip
> signaling was appropriate for this stuff, though the size of the data
> precludes doing so over UDP. For this app we decided we could just
> require TCP transport for the sip signaling of the recording session.
> (It doesn't constrain the transport for the calls being recorded.) This
> could be a problem in some topologies, but seems acceptable for the
> application. If there had been a mechanism that for a separate "data
> stream" that we could have reused we *might* have found that an
> attractive direction.
>=20
> ISTM that it would be helpful to have a "brainstorming session" with
> some reps from rtcweb, clue, siprec, bfcp, and mediactrl to see whether
> we can find some common machinery that could support multiples of these
> cases. I'm thinking that the sctp over udp stack being worked on for
> rtcweb might be more broadly applicable, both as a data stream
> transport
> negotiated with o/a, and maybe as a new sip transport as well.
>=20
> 	Thanks,
> 	Paul
>=20
> > On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:
> >
> >> Folks,
> >>
> >> I am sending this email to the RAI list because it is about an
> >> architectural issue that affects more than one WG.
> >>
> >> Both the CLUE and SIPREC WGs have identified the need to exchange
> >> metadata about media streams that have been established using SIP.
> Both
> >> groups need to transport such metadata between SIP UAs.
> >>
> >> In CLUE, they are studying how to transport their metadata, as
> >> documented in:
> >> http://tools.ietf.org/html/draft-wenger-clue-transport-01
> >>
> >> In SIPREC, they are planning to piggyback the metadata in SIP
> UPDATEs as
> >> an XML-encoded body part. When they need to send a request for full
> >> state (as opposed to partial state), they use an UPDATE request with
> a
> >> particular body type, as documented in:
> >> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
> >>
> >> The best way to exchange metadata within a SIP session depends on
> the
> >> type of metadata and on how the UAs need to interact with each other
> >> (e.g., a UA simply pushing information to the other UA or a more
> >> complicated protocol). Given that SIPREC and CLUE have different
> >> metadata and interactions, it may well be that we decide to use a
> >> different mechanism in each group. Nevertheless, I think both groups
> >> would benefit from architecturally-inclined RAI people having a look
> at
> >> their work. Please, continue this discussion focusing on their
> specific
> >> requirements on either the CLUE list or the SIPREC list.
> >>
> >> Somewhat related to this issue, the RTCWeb group discussed the use
> of
> >> SCTP over UDP as a reliable NAT-friendly way to exchange data
> between
> >> endpoints.
> >>
> >> Cheers,
> >>
> >> Gonzalo
> >>
> >> _______________________________________________
> >> RAI mailing list
> >> RAI@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rai
> >
> > _______________________________________________
> > RAI mailing list
> > RAI@ietf.org
> > https://www.ietf.org/mailman/listinfo/rai
> >
>=20
> _______________________________________________
> RAI mailing list
> RAI@ietf.org
> https://www.ietf.org/mailman/listinfo/rai

From pravindran@sonusnet.com  Tue Apr 24 02:34:40 2012
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From: "Ravindran, Parthasarathi" <pravindran@sonusnet.com>
To: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>, "rai@ietf.org" <rai@ietf.org>
Thread-Topic: [RAI] How to exchange metadata about the media streams established with a SIP session
Thread-Index: AQHNIYZ6YIVMz/h1yUC22aEQ7zGiuZapSyIAgABlU8A=
Date: Tue, 24 Apr 2012 09:34:31 +0000
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Subject: Re: [RAI] How to exchange metadata about the media streams established with a SIP session
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I agree with Andy as defining the metadata for SIPREC itself took=20
nearly 2 years by now, slowly folks understand SIPREC metadata model=20
and now we are discussing about SIPREC XML usage with individual=20
implementation.

In case we are aiming at come up with the common metadata model=20
for RTCWeb and SIPREC, then it takes ages as recording=20
is outside the scope of RTCWeb 1.0 usecases and XML vs. JSON will be
just a debate.

I understand the importance of CLUE signaling metadata & SIPREC=20
metadata to have commonality in the usage as both are based on SIP and
also SIPREC RTP usage with CLUE media & media-signaling metadata as=20
some of the telepresence entities may wish to record telepresence=20
session using SIPREC mechanism. I guess that it will be better to forward=20
SIPREC metadata draft, SIPREC RTP usage draft to CLUE folks to =20
understand any CLUE specific attributes has to be added. But I'm not sure=20
whether CLUE is in the state to provide those inputs at this moment.

Also, SIPREC may not require SCTP over UDP transport as most of the=20
deployed recording is based on UDP or TCP transport and there is no real
recording requirement to change SIP transport. =20

Thanks
Partha =20

>-----Original Message-----
>From: rai-bounces@ietf.org [mailto:rai-bounces@ietf.org] On Behalf Of
>Hutton, Andrew
>Sent: Tuesday, April 24, 2012 2:05 PM
>To: Paul Kyzivat; rai@ietf.org
>Subject: Re: [RAI] How to exchange metadata about the media streams
>established with a SIP session
>
>Hi,
>
>Whilst I of course agree that it is a good idea to look for a common
>solution if at all possible it is important to keep in mind the charter
>of the various working groups and the environment which the solutions
>will operate.
>
>Both the SIPREC and CLUE working groups are chartered to produce SIP
>based solutions which is possibly open to some different interpretations
>but it does impose some constraints on the solution. In the SIPREC case
>we therefore defined a SIP body based mechanism for metadata and stated
>that other mechanism although possible were out of scope I have no
>problem with looking at other mechanisms and would participate in any
>brainstorming session but I don't think we should change track regarding
>the current SIPREC mechanism especially since there are quite a few
>implementations already.
>
>Regards
>Andy
>
>
>
>
>> -----Original Message-----
>> From: rai-bounces@ietf.org [mailto:rai-bounces@ietf.org] On Behalf Of
>> Paul Kyzivat
>> Sent: 23 April 2012 20:23
>> To: rai@ietf.org
>> Subject: Re: [RAI] How to exchange metadata about the media streams
>> established with a SIP session
>>
>> On 4/23/12 10:53 AM, Cullen Jennings wrote:
>> >
>> > worth nothing the RTCweb folks also want to exchange metadata to do
>> with labeling streams. I always find just classifying it as "metadata"
>> sort of makes it hard to decide what to do with it. I'd prefer the
>> groups to specifically outline what data they need to exchange and
>> then I think it becomes clearer the best way to exchange the different
>> types of information. Knowing if the information changes per RTP
>> packet, or only one when the session is set up helps.
>>
>> Cullen,
>>
>> I agree that just talking about "metadata" isn't especially helpful.
>> OTOH, getting too far into the weeds about the specifics of the
>> metadata makes it hard for people with differing contexts to talk with
>> one another and find commonalities.
>>
>> In CLUE we realize that not all of our metadata can be treated the
>> same way. There are varying degrees of real-time-ed-ness to deal with.
>> We definitely have some stuff that changes at the signaling rate or
>> less, and some that changes at the media rate, and some stuff that
>> that falls in between. We also have issues that some of the stuff is
>> too big to reasonably transmit in sip signaling, which we are
>> considering a separate "data stream" for. We do think that CLUE should
>> be able to work with RTCWEB based endpoints. So its possible that the
>> RTCWEB data stream mechanism might work for us. But we will need to
>> have it even with non-rtcweb-endpoints.
>>
>> In SIPREC, we have a lot of descriptive stuff, mostly about signaling.
>> The rate is tied to the signaling rate, but possibly about multiple
>> calls and hence higher in aggregate. We decided that sending in the
>> sip signaling was appropriate for this stuff, though the size of the
>> data precludes doing so over UDP. For this app we decided we could
>> just require TCP transport for the sip signaling of the recording
>session.
>> (It doesn't constrain the transport for the calls being recorded.)
>> This could be a problem in some topologies, but seems acceptable for
>> the application. If there had been a mechanism that for a separate
>> "data stream" that we could have reused we *might* have found that an
>> attractive direction.
>>
>> ISTM that it would be helpful to have a "brainstorming session" with
>> some reps from rtcweb, clue, siprec, bfcp, and mediactrl to see
>> whether we can find some common machinery that could support multiples
>> of these cases. I'm thinking that the sctp over udp stack being worked
>> on for rtcweb might be more broadly applicable, both as a data stream
>> transport negotiated with o/a, and maybe as a new sip transport as
>> well.
>>
>> 	Thanks,
>> 	Paul
>>
>> > On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:
>> >
>> >> Folks,
>> >>
>> >> I am sending this email to the RAI list because it is about an
>> >> architectural issue that affects more than one WG.
>> >>
>> >> Both the CLUE and SIPREC WGs have identified the need to exchange
>> >> metadata about media streams that have been established using SIP.
>> Both
>> >> groups need to transport such metadata between SIP UAs.
>> >>
>> >> In CLUE, they are studying how to transport their metadata, as
>> >> documented in:
>> >> http://tools.ietf.org/html/draft-wenger-clue-transport-01
>> >>
>> >> In SIPREC, they are planning to piggyback the metadata in SIP
>> UPDATEs as
>> >> an XML-encoded body part. When they need to send a request for full
>> >> state (as opposed to partial state), they use an UPDATE request
>> >> with
>> a
>> >> particular body type, as documented in:
>> >> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
>> >>
>> >> The best way to exchange metadata within a SIP session depends on
>> the
>> >> type of metadata and on how the UAs need to interact with each
>> >> other (e.g., a UA simply pushing information to the other UA or a
>> >> more complicated protocol). Given that SIPREC and CLUE have
>> >> different metadata and interactions, it may well be that we decide
>> >> to use a different mechanism in each group. Nevertheless, I think
>> >> both groups would benefit from architecturally-inclined RAI people
>> >> having a look
>> at
>> >> their work. Please, continue this discussion focusing on their
>> specific
>> >> requirements on either the CLUE list or the SIPREC list.
>> >>
>> >> Somewhat related to this issue, the RTCWeb group discussed the use
>> of
>> >> SCTP over UDP as a reliable NAT-friendly way to exchange data
>> between
>> >> endpoints.
>> >>
>> >> Cheers,
>> >>
>> >> Gonzalo
>> >>
>> >> _______________________________________________
>> >> RAI mailing list
>> >> RAI@ietf.org
>> >> https://www.ietf.org/mailman/listinfo/rai
>> >
>> > _______________________________________________
>> > RAI mailing list
>> > RAI@ietf.org
>> > https://www.ietf.org/mailman/listinfo/rai
>> >
>>
>> _______________________________________________
>> RAI mailing list
>> RAI@ietf.org
>> https://www.ietf.org/mailman/listinfo/rai
>_______________________________________________
>RAI mailing list
>RAI@ietf.org
>https://www.ietf.org/mailman/listinfo/rai

From pkyzivat@alum.mit.edu  Tue Apr 24 06:36:01 2012
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Subject: Re: [RAI] How to exchange metadata about the media streams established with a SIP session
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On 4/24/12 4:35 AM, Hutton, Andrew wrote:
> Hi,
>
> Whilst I of course agree that it is a good idea to look for a common solution if at all possible it is important to keep in mind the charter of the various working groups and the environment which the solutions will operate.
>
> Both the SIPREC and CLUE working groups are chartered to produce SIP based solutions which is possibly open to some different interpretations but it does impose some constraints on the solution. In the SIPREC case we therefore defined a SIP body based mechanism for metadata and stated that other mechanism although possible were out of scope I have no problem with looking at other mechanisms and would participate in any brainstorming session but I don't think we should change track regarding the current SIPREC mechanism especially since there are quite a few implementations already.

Andy,

I don't see how anything that might be done about a general mechanism 
now would affect the output of SIPREC under its current charter. We are 
simply too far down the road for that.

*Conceivably* some general mechanism might eventually become a candidate 
for some future evolution/revision to siprec, but I wouldn't hold my 
breath for that to happen.

I see info about SIPREC as simply providing some perspective about the 
kinds of things a general mechanism might help with.

	Thanks,
	Paul

> Regards
> Andy
>
>
>
>
>> -----Original Message-----
>> From: rai-bounces@ietf.org [mailto:rai-bounces@ietf.org] On Behalf Of
>> Paul Kyzivat
>> Sent: 23 April 2012 20:23
>> To: rai@ietf.org
>> Subject: Re: [RAI] How to exchange metadata about the media streams
>> established with a SIP session
>>
>> On 4/23/12 10:53 AM, Cullen Jennings wrote:
>>>
>>> worth nothing the RTCweb folks also want to exchange metadata to do
>> with labeling streams. I always find just classifying it as "metadata"
>> sort of makes it hard to decide what to do with it. I'd prefer the
>> groups to specifically outline what data they need to exchange and then
>> I think it becomes clearer the best way to exchange the different types
>> of information. Knowing if the information changes per RTP packet, or
>> only one when the session is set up helps.
>>
>> Cullen,
>>
>> I agree that just talking about "metadata" isn't especially helpful.
>> OTOH, getting too far into the weeds about the specifics of the
>> metadata
>> makes it hard for people with differing contexts to talk with one
>> another and find commonalities.
>>
>> In CLUE we realize that not all of our metadata can be treated the same
>> way. There are varying degrees of real-time-ed-ness to deal with. We
>> definitely have some stuff that changes at the signaling rate or less,
>> and some that changes at the media rate, and some stuff that that falls
>> in between. We also have issues that some of the stuff is too big to
>> reasonably transmit in sip signaling, which we are considering a
>> separate "data stream" for. We do think that CLUE should be able to
>> work
>> with RTCWEB based endpoints. So its possible that the RTCWEB data
>> stream
>> mechanism might work for us. But we will need to have it even with
>> non-rtcweb-endpoints.
>>
>> In SIPREC, we have a lot of descriptive stuff, mostly about signaling.
>> The rate is tied to the signaling rate, but possibly about multiple
>> calls and hence higher in aggregate. We decided that sending in the sip
>> signaling was appropriate for this stuff, though the size of the data
>> precludes doing so over UDP. For this app we decided we could just
>> require TCP transport for the sip signaling of the recording session.
>> (It doesn't constrain the transport for the calls being recorded.) This
>> could be a problem in some topologies, but seems acceptable for the
>> application. If there had been a mechanism that for a separate "data
>> stream" that we could have reused we *might* have found that an
>> attractive direction.
>>
>> ISTM that it would be helpful to have a "brainstorming session" with
>> some reps from rtcweb, clue, siprec, bfcp, and mediactrl to see whether
>> we can find some common machinery that could support multiples of these
>> cases. I'm thinking that the sctp over udp stack being worked on for
>> rtcweb might be more broadly applicable, both as a data stream
>> transport
>> negotiated with o/a, and maybe as a new sip transport as well.
>>
>> 	Thanks,
>> 	Paul
>>
>>> On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:
>>>
>>>> Folks,
>>>>
>>>> I am sending this email to the RAI list because it is about an
>>>> architectural issue that affects more than one WG.
>>>>
>>>> Both the CLUE and SIPREC WGs have identified the need to exchange
>>>> metadata about media streams that have been established using SIP.
>> Both
>>>> groups need to transport such metadata between SIP UAs.
>>>>
>>>> In CLUE, they are studying how to transport their metadata, as
>>>> documented in:
>>>> http://tools.ietf.org/html/draft-wenger-clue-transport-01
>>>>
>>>> In SIPREC, they are planning to piggyback the metadata in SIP
>> UPDATEs as
>>>> an XML-encoded body part. When they need to send a request for full
>>>> state (as opposed to partial state), they use an UPDATE request with
>> a
>>>> particular body type, as documented in:
>>>> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
>>>>
>>>> The best way to exchange metadata within a SIP session depends on
>> the
>>>> type of metadata and on how the UAs need to interact with each other
>>>> (e.g., a UA simply pushing information to the other UA or a more
>>>> complicated protocol). Given that SIPREC and CLUE have different
>>>> metadata and interactions, it may well be that we decide to use a
>>>> different mechanism in each group. Nevertheless, I think both groups
>>>> would benefit from architecturally-inclined RAI people having a look
>> at
>>>> their work. Please, continue this discussion focusing on their
>> specific
>>>> requirements on either the CLUE list or the SIPREC list.
>>>>
>>>> Somewhat related to this issue, the RTCWeb group discussed the use
>> of
>>>> SCTP over UDP as a reliable NAT-friendly way to exchange data
>> between
>>>> endpoints.
>>>>
>>>> Cheers,
>>>>
>>>> Gonzalo
>>>>
>>>> _______________________________________________
>>>> RAI mailing list
>>>> RAI@ietf.org
>>>> https://www.ietf.org/mailman/listinfo/rai
>>>
>>> _______________________________________________
>>> RAI mailing list
>>> RAI@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rai
>>>
>>
>> _______________________________________________
>> RAI mailing list
>> RAI@ietf.org
>> https://www.ietf.org/mailman/listinfo/rai
>


From andrew.hutton@siemens-enterprise.com  Tue Apr 24 08:57:11 2012
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From: "Hutton, Andrew" <andrew.hutton@siemens-enterprise.com>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>
Date: Tue, 24 Apr 2012 17:57:07 +0200
Thread-Topic: [RAI] How to exchange metadata about the media streams established with a SIP session
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Subject: Re: [RAI] How to exchange metadata about the media streams established with a SIP session
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Hi Paul,

I think we are in sync.

Regards
Andy

> -----Original Message-----
> From: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]
> Sent: 24 April 2012 15:36
> To: Hutton, Andrew
> Cc: rai@ietf.org
> Subject: Re: [RAI] How to exchange metadata about the media streams
> established with a SIP session
>=20
> On 4/24/12 4:35 AM, Hutton, Andrew wrote:
> > Hi,
> >
> > Whilst I of course agree that it is a good idea to look for a common
> solution if at all possible it is important to keep in mind the charter
> of the various working groups and the environment which the solutions
> will operate.
> >
> > Both the SIPREC and CLUE working groups are chartered to produce SIP
> based solutions which is possibly open to some different
> interpretations but it does impose some constraints on the solution. In
> the SIPREC case we therefore defined a SIP body based mechanism for
> metadata and stated that other mechanism although possible were out of
> scope I have no problem with looking at other mechanisms and would
> participate in any brainstorming session but I don't think we should
> change track regarding the current SIPREC mechanism especially since
> there are quite a few implementations already.
>=20
> Andy,
>=20
> I don't see how anything that might be done about a general mechanism
> now would affect the output of SIPREC under its current charter. We are
> simply too far down the road for that.
>=20
> *Conceivably* some general mechanism might eventually become a
> candidate
> for some future evolution/revision to siprec, but I wouldn't hold my
> breath for that to happen.
>=20
> I see info about SIPREC as simply providing some perspective about the
> kinds of things a general mechanism might help with.
>=20
> 	Thanks,
> 	Paul
>=20
> > Regards
> > Andy
> >
> >
> >
> >
> >> -----Original Message-----
> >> From: rai-bounces@ietf.org [mailto:rai-bounces@ietf.org] On Behalf
> Of
> >> Paul Kyzivat
> >> Sent: 23 April 2012 20:23
> >> To: rai@ietf.org
> >> Subject: Re: [RAI] How to exchange metadata about the media streams
> >> established with a SIP session
> >>
> >> On 4/23/12 10:53 AM, Cullen Jennings wrote:
> >>>
> >>> worth nothing the RTCweb folks also want to exchange metadata to do
> >> with labeling streams. I always find just classifying it as
> "metadata"
> >> sort of makes it hard to decide what to do with it. I'd prefer the
> >> groups to specifically outline what data they need to exchange and
> then
> >> I think it becomes clearer the best way to exchange the different
> types
> >> of information. Knowing if the information changes per RTP packet,
> or
> >> only one when the session is set up helps.
> >>
> >> Cullen,
> >>
> >> I agree that just talking about "metadata" isn't especially helpful.
> >> OTOH, getting too far into the weeds about the specifics of the
> >> metadata
> >> makes it hard for people with differing contexts to talk with one
> >> another and find commonalities.
> >>
> >> In CLUE we realize that not all of our metadata can be treated the
> same
> >> way. There are varying degrees of real-time-ed-ness to deal with. We
> >> definitely have some stuff that changes at the signaling rate or
> less,
> >> and some that changes at the media rate, and some stuff that that
> falls
> >> in between. We also have issues that some of the stuff is too big to
> >> reasonably transmit in sip signaling, which we are considering a
> >> separate "data stream" for. We do think that CLUE should be able to
> >> work
> >> with RTCWEB based endpoints. So its possible that the RTCWEB data
> >> stream
> >> mechanism might work for us. But we will need to have it even with
> >> non-rtcweb-endpoints.
> >>
> >> In SIPREC, we have a lot of descriptive stuff, mostly about
> signaling.
> >> The rate is tied to the signaling rate, but possibly about multiple
> >> calls and hence higher in aggregate. We decided that sending in the
> sip
> >> signaling was appropriate for this stuff, though the size of the
> data
> >> precludes doing so over UDP. For this app we decided we could just
> >> require TCP transport for the sip signaling of the recording
> session.
> >> (It doesn't constrain the transport for the calls being recorded.)
> This
> >> could be a problem in some topologies, but seems acceptable for the
> >> application. If there had been a mechanism that for a separate "data
> >> stream" that we could have reused we *might* have found that an
> >> attractive direction.
> >>
> >> ISTM that it would be helpful to have a "brainstorming session" with
> >> some reps from rtcweb, clue, siprec, bfcp, and mediactrl to see
> whether
> >> we can find some common machinery that could support multiples of
> these
> >> cases. I'm thinking that the sctp over udp stack being worked on for
> >> rtcweb might be more broadly applicable, both as a data stream
> >> transport
> >> negotiated with o/a, and maybe as a new sip transport as well.
> >>
> >> 	Thanks,
> >> 	Paul
> >>
> >>> On Feb 28, 2012, at 3:20 , Gonzalo Camarillo wrote:
> >>>
> >>>> Folks,
> >>>>
> >>>> I am sending this email to the RAI list because it is about an
> >>>> architectural issue that affects more than one WG.
> >>>>
> >>>> Both the CLUE and SIPREC WGs have identified the need to exchange
> >>>> metadata about media streams that have been established using SIP.
> >> Both
> >>>> groups need to transport such metadata between SIP UAs.
> >>>>
> >>>> In CLUE, they are studying how to transport their metadata, as
> >>>> documented in:
> >>>> http://tools.ietf.org/html/draft-wenger-clue-transport-01
> >>>>
> >>>> In SIPREC, they are planning to piggyback the metadata in SIP
> >> UPDATEs as
> >>>> an XML-encoded body part. When they need to send a request for
> full
> >>>> state (as opposed to partial state), they use an UPDATE request
> with
> >> a
> >>>> particular body type, as documented in:
> >>>> http://tools.ietf.org/html/draft-ietf-siprec-protocol-02
> >>>>
> >>>> The best way to exchange metadata within a SIP session depends on
> >> the
> >>>> type of metadata and on how the UAs need to interact with each
> other
> >>>> (e.g., a UA simply pushing information to the other UA or a more
> >>>> complicated protocol). Given that SIPREC and CLUE have different
> >>>> metadata and interactions, it may well be that we decide to use a
> >>>> different mechanism in each group. Nevertheless, I think both
> groups
> >>>> would benefit from architecturally-inclined RAI people having a
> look
> >> at
> >>>> their work. Please, continue this discussion focusing on their
> >> specific
> >>>> requirements on either the CLUE list or the SIPREC list.
> >>>>
> >>>> Somewhat related to this issue, the RTCWeb group discussed the use
> >> of
> >>>> SCTP over UDP as a reliable NAT-friendly way to exchange data
> >> between
> >>>> endpoints.
> >>>>
> >>>> Cheers,
> >>>>
> >>>> Gonzalo
> >>>>
> >>>> _______________________________________________
> >>>> RAI mailing list
> >>>> RAI@ietf.org
> >>>> https://www.ietf.org/mailman/listinfo/rai
> >>>
> >>> _______________________________________________
> >>> RAI mailing list
> >>> RAI@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/rai
> >>>
> >>
> >> _______________________________________________
> >> RAI mailing list
> >> RAI@ietf.org
> >> https://www.ietf.org/mailman/listinfo/rai
> >


From dworley@avaya.com  Fri Apr 27 15:01:29 2012
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From: "Worley, Dale R (Dale)" <dworley@avaya.com>
To: "rai@ietf.org" <rai@ietf.org>, Al Morton <acmorton@att.com>, "draft-ietf-bmwg-sip-bench-term@tools.ietf.org" <draft-ietf-bmwg-sip-bench-term@tools.ietf.org>, "bmwg@ietf.org" <bmwg@ietf.org>
Date: Fri, 27 Apr 2012 17:56:42 -0400
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Subject: [RAI] RAI-ART review of draft-ietf-bmwg-sip-bench-term-04 and draft-ietf-bmwg-sip-bench-meth-04
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I am the assigned RAI-ART reviewer for this draft.  For background on
RAI-ART, please see the FAQ at
<http://wiki.tools.ietf.org/area/rai/trac/wiki/RaiArtfaq>.  Please
resolve these comments along with any other comments you may receive.

This draft is on the right track but has open issues, described in the
review.

Please note that I am quite familiar with SIP but have little
familiarity with benchmarking.  Thus, some of the points I make might
have implicit answers in the usual practice of benchmarking.  But
since one audience of these documents is the SIP community, such
implicit answers should probably be made explicit.

I. Technical issues

A. Media

The drafts seem to be undecided as to whether they want to benchmark
media processing or not.  At one point, it says that the benchmarking
is entirely in the signaling plane.  At another point, it is specified
how to set the number of media streams per INVITE dialog.  I suspect
the conflict is that there are devices the WG wants to benchmark which
do handle the media and their media handling affects performance
significantly, but the WG is unsure how to parametrize media
benchmarking.

The simplest situation, of course, is if the DUT is a proxy and the
media bypasses it entirely.  Beyond that, even in the simplest media
relaying situations, significant processing costs can arise.  Even if
the DUT isn't transcoding, it may have to rewrite the RTP codec
numbers (as it may have modified the SDP passing through it).  In more
complex situations, the DUT may have to transcode the codecs.

Some consideration has to be made of how to specify exactly what the
media loads are.  SDP already has ways to specify the exact encoding
of media streams, including the number of samples per packet, which
implies the packet size and packet rate.  These drafts only allow
specifying the packet size as a benchmark parameter, but packet size
is notoriously ambiguous -- which layers of encapsulation of the
encoded media are counted?  But specifying the detailed encoding
side-steps that question.

The draft wants to consider the duration of a media stream to be
separately adjustable from the duration of the containing dialog, but
the draft explicitly places out of scope the re-INVITE which is
necessary to accomplish that realistically (that is, with the
signaling matching the presented media packets).

B. INVITE dialog structure

The drafts seem to want to consider the establishment (or failure
thereof) of an INVITE dialog to be instantaneous, after which the
dialog continues for a chosen length of time, and then vanishes
instantly.  Little or no consideration is given to the various
scenarios of call establishment, including the most common case:
INVITE sent -- 183 response -- significant delay -- 200 response.
Dialog teardown is not conceptualized as being a processing step that
involves significant cost and may fail:  "Session disconnect is not
considered in the scope of this work item."

This lack of consideration is enhanced in the forking cases, as the
variety of scenarios (and their durations) increases.

In addition, the drafts only consider forking which is done within the
DUT, whereas it will be common in practice for forking to be done
downstream of the DUT, presenting the DUT with a stream of 1xx
responses from multiple endpoints, with a 2xx after an extended delay.

Also, in regard to signaling benchmarking, INVITEs that ultimately
fail are likely to be as costly as INVITEs that succeed, but there
doesn't seem to be a defined parameter "fraction of attempted calls
which succeed" (which controls the callee EAs).

C. Loop detection

All discussions of loop detection needs to based on the revised loop
detection requirements in RFC 5393.

D. Authentication

In some SIP operations, authentication is commonly done.  This can
have various effects on the message flows that need to be taken into
account in the benchmarks.

For instance, a registrar may require that the registering UA
authenticate itself.  Commonly, the UA sends a REGISTER request, which
is rejected with 401 because it contains a nonce that is too old.  The
UA then immediately sends another REGISTER with the nonce provided in
the 401 response, and that request receives a 200 response.

In this scenario, the number of effective REGISTER requests is half of
the total REGISTER requests, leading to an apparent attempt failure
rate of 50%, even though the middlebox is doing the Right Thing 100%
of the time.  This suggests that the definition of "attempt failure"
needs to be updated so that a 4xx response "passed upstream" by the
DUT is not counted as an attempt failure.

In other scenarios, the DUT itself might be expected to enforce SIP
authentication, which would require a somewhat different definition of
attempt failure, and would be expected to have lower throughput.

So some thought needs to be given as to whether these scenarios are
to be benchmarked, and to document how authentication is to be handled
in whatever benchmarks are defined.

II. Editorial issues

The drafts appear to me to be, well, drafts.  That is, they generally
contain the intended technical content but the exposition is not
complete or clear, and at points the reader has to guess at the exact
meaning.  At various points the sentences are not correct English,
references are not complete or correct, and there are various points
of contradiction between different parts of the documents.

The document needs a proper final revision, with each paragraph edited
over carefully to maximize its clarity and to verify that all parts of
the documents are consistent with each other.

Examples of editorial problems are:

draft-ietf-bmwg-sip-bench-term-04

section 1

   The term Throughput is defined in RFC2544 [RFC2544].

The definition is in RFC 1242; RFC 2544 refers to RFC 1242.

   This document uses existing terminology defined in other BMWG work.
   Examples include, but are not limited to:

      Device under test (DUT) (c.f., Section 3.1.1 RFC 2285 [RFC2285]).
      System under test (SUT) (c.f., Section 3.1.2, RFC 2285 [RFC2285]).

In what way would the reader determine the relevant "other BMWG work"?
This reference needs to be made definite in some way.

      The behavior of a stateful proxy is further defined in Section
      16.

This sentence was copied directly from RFC 3261, in whose section 16
the referenced definition exists.  However, *this* document has no
section 16.

section 2.2

The figures are inconsistent in how they label the "Tester" boxes as
"EA" or not.  Is this difference meaningful?

Figures 9 and 10 show "SUT" embracing the entire test setup, including
the "Tester" boxes, whereas RFC 2285 section 3.1.2 says that SUT does
not include the tester components of the setup.

section 3.1.1

The various components of a "session" (usually "dialog" in SIP
terminology) are given quasi-mathematical names that look peculiar to
me.  Worse, they're not completely correct: since each RTP stream has
a corresponding RTCP stream, "session[x].medc" should be
"session[x].medc[y]".  (Is there any benefit of introducing this
symbolism?)

section 3.1.5

This section defines "overload" and then says "The distinction between
an overload condition and other failure scenarios is outside the scope
of this document which is blackbox testing."  If the distinction is
outside the scope, why is there a definition here?

section 3.1.6

The definition of "Session Attempt" seems to be incorrect.  Of course,
a session attempt is each sending of an INVITE/SUBSCRIBE/MESSAGE,
whether or not it is ultimately successful.  But the definition written
makes "session attempt" a time-varying property, which is true only
until a response is received by the EA.

section 3.1.8

      An IS is identified by the Call-ID, To-tag, and From-tag of the
      SIP message that establishes the session.

As written, this is incorrect, as the to-tag is present only in the
response(s) to the INVITE.

      2.  If a media session is described in the SDP body of the
          signaling message, then the media session is established by
          the end of Establishment Threshold Time (c.f.  Section 3.3.3).

This sentence is correct as far as it goes, but there is no clear
description (that I can tell) of when a media session is
"established".  Indeed, it's not clear to me what a proper way to set
up the media is -- in real SIP systems, a proxy or SBC can start
receiving early media from a callee before it has received *any* SIP
responses from the callee, and the middlebox can have some difficulty
matching the RTP to a dialog being set up.  So the exact details of
when to start the media flows are needed to make the benchmarking
process reproducible.

section 3.2.3

This section defines "SIP-Aware Stateful Firewall" as=20

      Device in test topology that provides Denial-of-Service (DoS)
      Protection to the Signaling and Media Planes for the EAs and
      Signaling Server

But a device can be a SIP-Aware Stateful Firewall (in the ordinary
sense of the words) without providing DoS protection.

section 3.3.2

"IS Media Attempt Rate" is defined as

      Configuration on the EA for number of ISs with Associated Media to
      be established at the DUT per continuous one- second time
      intervals.

However, "established" should be "attempted".

section 3.3.3

      Configuration of the EA for representing the amount of time that
      an EA will wait before declaring a Session Attempt Failure.

The discussion makes clear that there may be different Establishment
Threshold Times for IS and NS, but these times are written about in
most places as if they were the same.

draft-ietf-bmwg-sip-bench-meth-04

section 2

Refers to "Presence Rate", but draft-ietf-bmwg-sip-bench-term-04 says
that presence is out of scope.

section 4

What is the relationship between the data items described in this
draft's "Benchmarking Considerations" and
draft-ietf-bmwg-sip-bench-term-04's "Test Setup Parameters"?

section 4.2

I'm having trouble understanding the various terms that include the
word "server", including "Signaling Server", "The Server", "SIP
Server".  Are these intended to have the same meaning?  Are they
intended to cover the whole range of SIP "middleboxes" (including,
e.g., proxies)?  In section 2 is

   The DUT is a SIP Server, which
   may be any [RFC3261] conforming device.  The SUT can be any device or
   group of devices containing RFC 3261 conforming functionality along
   with Firewall and/or NAT functionality.

If read literally, that would *exclude* pure proxies, since they don't
have firewall or NAT functionality.

section 4.3

References "Figures 4 and 5", which don't exist.  Does it mean to
reference draft-ietf-bmwg-sip-bench-term-04?

section 4.9

The formatting of the pseudocode is inconsistent.  The variable "c"
should be boolean rather than 0/1.

section 6.6

      1.  If the DUT is being benchmarked as a proxy or B2BUA, and
          forking is supported in the DUT, then configure the DUT in the
          test topology shown in Figure 5 in [I-D.sip-bench-term].  If
          the DUT does not support forking, then this step can be
          skipped.
      2.  Configure a SUT according to the test topology shown in Figure
          8 of [I-D.sip-bench-term].

It's not clear to me how one can configure the DUT/SUT according to
both figures 5 and 8.  And neither figure shows two callee EAs.

The text suggests that DUTs that do not support forking can be tested,
even though this is a test specifically of performance when the DUT is
doing forking.

Dale
