RE: AW: [Sip] Extension of conference procedures

[Peili Xu <[e-mail]>]

Hi Roni,

Your case is even "smarter" than mine ;-)  that's OK for
me.

But, I'm just wondering whether there is case that A has
multiple on going dialogs with say B, C, D and only want
to turn A-B and A-C to a conference then dialog
information maybe needed.  

I'd happy to hear clarification for this point from whom
raise the requirements.

Peili

-----Original Message-----
From: Even, Roni [mailto:[e-mail]] 
Sent: Thursday, August 30, 2007 2:09 PM
To: Peili Xu
Cc: [e-mail]
Subject: RE: AW: [Sip] Extension of conference
procedures



Hi,
I think you misunderstood what I menat by "smart". If
there is a softyswitch acting as a 3PCC, it controls the
dialogs and can redirect the media to the media server
suppling the conference function.
There is no need for A to give information about the
dialog

Roni Even

> -----Original Message-----
> From: Peili Xu [mailto:[e-mail]]
> Sent: Wednesday, August 29, 2007 5:08 PM
> To: Even, Roni
> Cc: Huelsemann, Martin; [e-mail]
> Subject: Re: AW: [Sip] Extension of conference
procedures
> 
> Hi Martin, Denis,
> 
> I agree with Roni that you may need to decide who is
"smart".
> 
> I guess you want to simulate the 3PTY services in
PSTN, A make call to 
> B, then A Hold B, then A Make Call to C, then A could
do sth like hook 
> flash to turn the call between A-B and A-C to an 3pty
conference.
> later, A could still turn the conferece back to 2
independant call.
> 
> You have some assumption that the AS who performs
conference is on the 
> path between A-B and A-C. And A could inform the AS to
turn the call 
> between A-B and A-C to a conference.
> 
> If the assumption is correct, what you want is just to
tell the AS 
> which dialogs should be turned to conference by
sending an INVITE with 
> the related dialog information.
> 
> If the above understanding is correct, I'd agree with
the initial 
> proposal from Denis.
> Just to convey the dialog information along with the
URI-List.
> 
> Peili
> 
> 
> 
> 2007/8/29, Even, Roni <[e-mail]>:
> >
> > Hi,
> > In this case, like in PSTN the switch does it. You
have to decide 
> > who is
> "smart" the network or the end device.
> > A "simple" RFC 3261 only phone relies on a 3PCC or
softswitch to 
> > manage
> telephony services (not only conferencing)
> >
> > Roni Even
> >
> > > -----Original Message-----
> > > From: Huelsemann, Martin
[mailto:[e-mail]]
> > > Sent: Wednesday, August 29, 2007 1:05 PM
> > > To: Even, Roni
> > > Cc: [e-mail]; [e-mail]
> > > Subject: AW: AW: [Sip] Extension of conference
procedures
> > >
> > > Hi,
> > >
> > > also for the scenario where A refers B to dial
into a conference, 
> > > the problem is when B has a terminal not
supporting REFER (or just 
> > > doesn't want to accept the REFER for some
reasons), A cannot use 
> > > the
> conference
> > > service.
> > >
> > > Of course these simple terminals are not the
desired use-case and
> there
> > > will be limitations. But if there is a possible
fallback solution 
> > > that
> at
> > > least increases the chance that A can use the
service despite the 
> > > fact that B does not fulfill all the requirements
for the service, 
> > > I think
> we
> > > should try to figure it out.
> > >
> > > Regards, Martin
> > >
> > >
> > >
> > >
> > >
> > > > -----Ursprüngliche Nachricht-----
> > > > Von: Even, Roni [mailto:[e-mail]]
> > > > Gesendet: Montag, 27. August 2007 12:16
> > > > An: Hülsemann, Martin; [e-mail]
> > > > Cc: [e-mail]
> > > > Betreff: RE: AW: [Sip] Extension of conference
procedures
> > > >
> > > >
> > > > Hi,
> > > > My view is that every solution where you only
have A, B and a 
> > > > conference server and B only supports RFC 3261
will have some 
> > > > limitation and will be a hack.
> > > >
> > > > The recommended way to do it is for A to send a
Refer to B to 
> > > > the focus.
> > > >
> > > > Also asking that B supporting only RFC 3261 will
support 
> > > > conference event package is contradictory. B
will not even be 
> > > > aware that it is in a conference.
> > > >
> > > > Roni Even
> > > >
> > > > !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
> > > > A consensus means that everyone agrees to say
collectively what 
> > > > no one believes individually
!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
> > > >
> > > > > -----Original Message-----
> > > > > From: Huelsemann, Martin
[mailto:[e-mail]]
> > > > > Sent: Monday, August 27, 2007 12:59 PM
> > > > > To: [e-mail]
> > > > > Cc: [e-mail]
> > > > > Subject: AW: AW: [Sip] Extension of conference
procedures
> > > > >
> > > > > Hi,
> > > > >
> > > > > the pure RFC 3261 client won't be the normal
case, of
> > > > course. But there
> > > > > might be networks with which you want to
interwork where
> > > > those simple
> > > > > clients are existing.
> > > > > Okay, you got me, it's again the PSTN
interworking. So
> > > > let's say what is
> > > > > needed is a fallback solution for this case.
> > > > > On the other hand, this fallback might be
useful for a
> > > > normal SIP client
> > > > > which supports RFC3891, too, e.g. if there are
problems with 
> > > > > authorization.
> > > > >
> > > > > The user experience of the invitee should be
exactly as you
> > > > describe.
> > > > >
> > > > > If A sends the reINVITE / UPDATE himself that
could be a
> > > > solution, too.
> > > > > The only thing is, can B then use all the
conference features (e.
> g.
> > > > > conference event package), when the focus has
no knowledge on 
> > > > > the signalling level that B is connected to
the conference bridge?
> > > > >
> > > > > Regards, Martin
> > > > >
> > > > >
> > > > >
> > > > >
> > > > >
> > > > > > -----Ursprüngliche Nachricht-----
> > > > > > Von: Jeroen van Bemmel
[mailto:[e-mail]]
> > > > > > Gesendet: Sonntag, 26. August 2007 15:24
> > > > > > An: Hülsemann, Martin
> > > > > > Cc: Alexeitsev, Denis; [e-mail]
> > > > > > Betreff: Re: AW: [Sip] Extension of
conference procedures
> > > > > >
> > > > > >
> > > > > > Martin,
> > > > > >
> > > > > > Now it becomes more clear. So the
requirement is to
> > > > enable a scenario
> > > > > > where a regular call is transformed into a
conference
> > > > call, assuming
> > > > > > that the invitee only has a "pure RFC3261"
client.
> > > > > > More specifically:
> > > > > >
> > > > > > - to get a smooth user experience, the
scenario must not 
> > > > > > cause
> the
> > > > > > invitee's phone to ring, and/or ask the
invitee for 
> > > > > > permission (acceptance is assumed to be
implied)
> > > > > >
> > > > > > What if A would send the reINVITE (or
UPDATE) itself, while 
> > > > > > filling in the SDP according to the media
provided by the 
> > > > > > conference server (i.e.
> > > > > > use codecs, media ports as received from the
focus)? (trying 
> > > > > > to get the requirements more clear here)
> > > > > >
> > > > > > In practice, it may still be a challenge to
get such "very
> simple
> > > > > > terminals" to properly handle a change of
media (i.e. new
> > > > destination
> > > > > > ip/port for sending, new source ip/port and
RTP src id,
> > > > and possibly
> > > > > > different codecs)
> > > > > >
> > > > > > Regards,
> > > > > > Jeroen
> > > > > >
> > > > > >
> > > > > > Huelsemann, Martin wrote:
> > > > > > > Hi Jeroen,
> > > > > > >
> > > > > > > the usage of the Replaces header is of
course the best
> > > > > > solution, it's also described in the
regarding 3GPP 
> > > > > > conferencing spec.
> > > > > > > The disadvantage of the usage of the
Repaces header is,
> > > > > > that it puts requirements on the UE of the
invited user, it 
> > > > > > would have to support RFC 3891. And if it
supports, it 
> > > > > > really would have to accept the 2nd INVITE,
which is not 
> > > > > > mandatory according to RFC 3891 I think.
> > > > > > > Anyway what is needed in addition is a
solution how also
> > > > > > very simple terminals (e. g. only supporting
RFC 3261) can 
> > > > > > be invited to an ad-hoc conference, whithout
having to say 
> > > > > > to the invited user to please hang up
because there will be 
> > > > > > a call from the focus shortly.
> > > > > > >
> > > > > > > Re-using an already established dialog at
least at the
> > > > > > first glance seems to be a simply and
invited UE independent 
> > > > > > solution. And as this re-INVITE it wanted by
at least one of 
> > > > > > the involved user, I would more compare it
to some kind of 
> > > > > > triggered 3rd party call control than to
spoofing.
> > > > > > >
> > > > > > > Of course as you said it would have to be
made clear that
> > > > > > the focus is able to collect all the
information needed for 
> > > > > > sending re-INVITEs (proposed "?" mechanism
usage, dialog 
> > > > > > event package, etc.).
> > > > > > >
> > > > > > >
> > > > > > > Regards, Martin
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >
> > > > > > >> -----Ursprüngliche Nachricht-----
> > > > > > >> Von: Jeroen van Bemmel
[mailto:[e-mail]]
> > > > > > >> Gesendet: Freitag, 24. August 2007 17:06
> > > > > > >> An: Alexeitsev, Denis; [e-mail]
> > > > > > >> Betreff: Re: [Sip] Extension of
conference procedures
> > > > > > >>
> > > > > > >>
> > > > > > >> Denis,
> > > > > > >>
> > > > > > >> If I understand your scenario, "focus" is
a third party 
> > > > > > >> separate from A and B, right? (e.g. a
conference server)
> > > > > > >>
> > > > > > >> In that case the focus is not a party in
the A-B dialog, 
> > > > > > >> and would need more than Call-ID, From
and To to be able 
> > > > > > >> to construct a reINVITE that B would
accept as coming 
> > > > > > >> from A (e.g. CSeq). In any case, this
looks like a very 
> > > > > > >> inelegant, hacked solution (as the
conference server is 
> > > > > > >> basically spoofing)
> > > > > > >>
> > > > > > >> RFC4579 section 5.10 provides some
insipration, as well 
> > > > > > >> as
> > > > > > >>
http://www.ietf.org/internet-drafts/draft-ietf-sipping-
> service
> > > > > > >> -examples-13.txt
> > > > > > >> scenario 2.5
> > > > > > >>
> > > > > > >> For example, A could include a 'Replaces'
header in the 
> > > > > > >> URI it includes in its conference URI
list. Then the 
> > > > > > >> conference server would
> > > > > > send a new
> > > > > > >> (out-of-dialog) INVITE to B containing
this Replaces 
> > > > > > >> header, and B would know that it is
associated with the 
> > > > > > >> dialog it has with A (and can replace it,
without ringing 
> > > > > > >> if the UA is constructed like that)
> > > > > > >>
> > > > > > >> The conference server should probably
also include a 
> > > > > > >> Referred-By containing A's AoR, either
automatically 
> > > > > > >> (i.e. copy from From header in
> > > > > > >> INVITE) or
> > > > > > >> included by A in the URI (former is
better)
> > > > > > >>
> > > > > > >> Regards,
> > > > > > >> Jeroen
> > > > > > >>
> > > > > > >> Alexeitsev, D wrote:
> > > > > > >>
> > > > > > >>> I'd like to discuss the extension of the
current 
> > > > > > >>> conference
> > > > > > >>>
> > > > > > >> procedures
> > > > > > >>
> > > > > > >>> with the following functionality.
> > > > > > >>>
> > > > > > >>> 3GPP conference specifications are
basing generally on 
> > > > > > >>> the Conferencing Framework (RFC 4353)
and for one 
> > > > > > >>> possibility
> > > > > > >>>
> > > > > > >> of inviting
> > > > > > >>
> > > > > > >>> users to the confrence on
> > > > draft-ietf-sip-uri-list-conferencing.
> > > > > > >>>
> > > > > > >>> Using the conferencing framework, the
following 
> > > > > > >>> situation
> > > > > > can occur
> > > > > > >>> when a user is invited to an ad-hoc
conference:
> > > > > > >>> User A is in a dialog with user B, and
decides to start 
> > > > > > >>> a
> > > > > > >>>
> > > > > > >> conference,
> > > > > > >>
> > > > > > >>> for example using an INVITE request to
the focus which
> > > > > > >>>
> > > > > > >> includes a URI
> > > > > > >>
> > > > > > >>> list with the URIs of the users which
shall be added to 
> > > > > > >>> the conference, incl. B. So when the
INVITE request from 
> > > > > > >>> the
> focus
> > > > > > >>> arrives at B, he is still in the
original dialog with
> > > > A, and so it
> > > > > > >>> depends on B if he accepts the 2nd
INVITE and the
> > > > > > conference can be
> > > > > > >>> established.
> > > > > > >>>
> > > > > > >>> At the last 3GPP CT1 meeting the idea of
transporting 
> > > > > > >>> dialog identifiers together with the
URIs was introduced 
> > > > > > >>> to
> > > > solve this
> > > > > > >>> problem. Basing on the idea that the
procedures at
> > > > the conference
> > > > > > >>> server are extended in that way, that
the conference
> > > > > > server is aware
> > > > > > >>> of already established dialogs, the
focus then has the
> > > > > > >>>
> > > > > > >> possibility to
> > > > > > >>
> > > > > > >>> send re-INVITES in the indicated dialogs
and connect the
> > > > > > media from
> > > > > > >>> the invited users to the conference
bridge.
> > > > > > >>> In the URI list the dialogs can be
indicated using the
> > > > > > "?" mechanism
> > > > > > >>> according to subclause 19.1.1 of RFC
3261.
> > > > > > >>>
> > > > > > >>> Following example shows the proposed
mechanism:
> > > > > > >>>
> > > > > > >>> INVITE Conference
> > > > > > >>> To: Conference
> > > > > > >>> From: A
> > > > > > >>> Require: recipient-list-invite
> > > > > > >>>
> > > > > > >>> Content-Type:
application/resource-lists+xml
> > > > > > >>> Content-Disposition: recipient-list
> > > > > > >>>
> > > > > > >>> <?xml version="1.0" encoding="UTF-8"?>
<resource-lists 
> > > > > > >>> xmlns="urn:ietf:params:xml:ns:resource-
> lists"
> > > > > > >>>
xmlns:cp="urn:ietf:params:xml:ns:copyControl">
> > > > > > >>>   <list>
> > > > > > >>>    <entry
uri="B?Call-ID=1&From=A%3Btag%3Da&To=B%3Btag%3Db"
> > > > > > >>> cp:copyControl="to"/>
> > > > > > >>>    <entry
uri="C?Call-ID=2&From=A%3Btag%3Da&To=C%3btag%3Dc"
> > > > > > >>> cp:copyControl="to"/>
> > > > > > >>>   </list>
> > > > > > >>>  </resource-lists>
> > > > > > >>>
> > > > > > >>> Greetings,
> > > > > > >>> Denis Alexeitsev
> > > > > > >>>
> > > > > > >>>
> > > > > > >>>
_______________________________________________
> > > > > > >>> Sip mailing list  
> > > > > > >>>
https://www1.ietf.org/mailman/listinfo/sip
> > > > > > >>> This list is for NEW development of the
core SIP 
> > > > > > >>> Protocol Use
[e-mail] for 
> > > > > > >>> questions on
> > > > current sip
> > > > > > >>> Use [e-mail] for new
developments on the
> > > > > > application of sip
> > > > > > >>>
> > > > > > >>
> > > > > > >>
_______________________________________________
> > > > > > >> Sip mailing list  
> > > > > > >>
https://www1.ietf.org/mailman/listinfo/sip
> > > > > > >> This list is for NEW development of the
core SIP Protocol 
> > > > > > >> Use [e-mail] for
questions on
> > > > current sip
> > > > > > >> Use [e-mail] for new developments
on the
> > > > application of sip
> > > > > > >>
> > > > > > >>
> > > > > > >
> > > > > > >
> > > > > >
> > > > >
> > > > >
> > > > >
_______________________________________________
> > > > > Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip
> > > > > This list is for NEW development of the core
SIP Protocol Use 
> > > > > [e-mail] for questions
on current sip 
> > > > > Use [e-mail] for new developments on
the application 
> > > > > of
> sip
> > > >
> >
> >
> > _______________________________________________
> > Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip
> > This list is for NEW development of the core SIP
Protocol Use 
> > [e-mail] for questions on
current sip Use 
> > [e-mail] for new developments on the
application of sip
> >


_______________________________________________
Sip mailing list
https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP
Protocol Use [e-mail] for
questions on current sip Use [e-mail] for new
developments on the application of sip




_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use [e-mail] for questions on current sip
Use [e-mail] for new developments on the application of sip