[Serusers] RE: Transfer and Conferencing - Please help!
Greg Fausak
greg at addabrand.com
Fri Nov 5 00:23:23 CET 2004
Bob,
I can offer some ideas that might help.
I certainly don't intend to be condescending...
Many of my questions are answered with call traces.
For example, I worked on a bug with REINVITES today.
http://www.addaline.com/traces/andy_index.html
This is created by using a :
1) switch with port monitoring
2) ethereal (or tcpdump) to grab data
3) sipscenario to format the data into the call trace
A transfer can be done in a few different ways, especially when
you get an IP-PBX involved. There is a popular one called
Asterisk that can do transfers between extensions. If you built
it, get phones to register with it, and connected the outside with
a SIP provider you could do some call traces and see how Asterisk
makes it happen.
A conference is a different animal. I don't think there is any
SIP call per se to build a conference. Some UAs have the
function built in, and they actually create more than one phone call
and mix the sound internally. For example, the Cisco 7960 IP
phone does that.
I guess the basic problem is that SIP is a protocol, transfer is
a feature that is implemented with the SIP protocol. There are
quite a few ways to skin that cat :-)
-g
On Nov 4, 2004, at 4:46 PM, Bob Carlson wrote:
> I sent this earlier and got no responses. Perhaps this is not the
> right
> forum to ask this question. Can any one suggest a better place to go
> for
> this information?
>
> Thanks, Bob
>
> -----Original Message-----
> From: Bob Carlson
> Sent: Wednesday, November 03, 2004 3:22 PM
> To: 'SerUsers'
> Subject: Transfer and Conferencing
>
> Let me apologize in advance for my question, which is a little
> rudimentary.
> We are just starting a project that will use SER and I am being forced
> to
> document right now how transfer and conferencing will be handled. I
> have
> spent a lot of time looking for definitive information on the subject
> with
> no luck. Well, maybe too much luck. There seem to be many proposals
> and
> models and so on, but it is not clear to me what is actually being
> done in
> practice. I have downloaded all the RFCs and proposal papers on the
> subject. I am still reviewing them, but I think the folks on this
> forum can
> help me a lot.
>
> I need to know the SIP message sequences for performing a call
> transfer and
> a blind call transfer and for constructing a conference. I have found
> information in proposals, but I need to know what actual, available SIP
> phones can do. We have some phones that we will test, but I do not
> know
> what they do when you press their transfer and conference buttons.
> Pardon
> me again for my impatience in asking before I have tried this out.
>
> The Transfer models are straightforward, but conferencing is more
> complicated. We must construct a simple conferencing model where the
> conferencing is performed by a central server, a SIP IPX. Only
> conferences
> of 3 participants need to be supported. We want it to look exactly
> like
> 3-way calling on your home phone. During a call, put the call on hold
> with
> a conference button, call another phone, hit conference button, the two
> calls are joined in a 3-way conference.
>
> The document draft-ietf-sipping-service-examples-07.txt seems to be
> very
> helpful on the subject, but all examples are in the form of 3 or more
> UAs
> and do not address any examples from the point of view of a PBX. I
> can see
> how to extend the examples to a PBX case, except for one aspect. If
> the
> IP-PBX is to perform the action as a proxy, what does the phone send
> the
> IP-PBX to indicate the steps in the process. Put more plainly, what
> happens
> when the user hits the Transfer or Conference button on the phone?
> What
> message is sent to the IP-PBX?
>
> Can anyone tell me where else I should be looking? Is the service
> examples
> draft the best base document to work from?
>
> Thanks in advance, Bob Carlson
>
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>
>
Greg Fausak
www.AddaBrand.com
(US) 469-546-1265
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