Patrick,
Thanks for your questions on the draft. The flow could be done either way, with
B sending the REFER to either C or A.
There are two objectives in this flow - that after the REFER is sent, neither A
or C's phone should ring for the resulting INVITE, and that the recipient of the
INVITE can determine that the resulting session between A and C should replace
the existing session between A and B. I thought that sending the REFER to C and
possibly including a Replaces header in the resulting INVITE from C to A does
this rather cleanly.
As for billing issues, I think this flow matches the existing PSTN if the
transfer occurs behind a PBX. For example, if A calls B, then B signals the PBX
to transfer A to C, the call would be hairpined at the PBX, and the B to C leg
would be billed to B.
However, this flow could work either way.
As for the RTP flow between B and C, it is initially set up so that B can tell C
"I'm going to transfer A to you - hold on." Since this is an attend transfer
flow, B should maintain sessions with both A and C in the event of a failure.
Once B is notified that the transfer has succeeded, a BYE is sent tearing down
this session. Note that B could put C on hold before sending the REFER.
If you have an alternative flow for this example, lets discuss.
Thanks,
Alan Johnston
WorldCom
sip:[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
> Sirs,
>
> I have questions regarding I-D draft-ietf-sip-service-examples-01.txt.
> In the the "attended transfer" why B asks C to call A whereas A has called B.
> Could i imagine that after B having in communication with C, B puts C on
> hold and asks A to call C?
> It could solve some Billing Pbs. A would stay the originator of the call and
> could be charged .
>
> In the same examples: What is the RTP flow kept between B and C for ?
>
> Thank you for your answers,
>
> pm
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