RE: AW: [Sip] Extension of conference procedures

["Even, Roni" <[e-mail]>]

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
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> > > > > > >>> 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
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> >


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