[SR-Users] How does SIP transfer work?

[Gaston Draque]

http://www.vocal.com/sip-1/call-transferring/



On Thu, Nov 22, 2012 at 10:58 AM, Dmytro Bogovych <dmytro.bogovych at gmail.com
> wrote:

> REFER
> http://tools.ietf.org/html/rfc3515
>
>
> On Thu, Nov 22, 2012 at 3:55 PM, Grant Bagdasarian <GB at cm.nl> wrote:
> > Hello,
> >
> >
> >
> > I’ve been searching the internet to find an explanation on how SIP
> transfer
> > works using Re-INVITE and/or UPDATE, but I can’t seem to find a good
> source.
> >
> >
> >
> > From what I understand(and this is the way we do it), the following
> happens:
> >
> >
> >
> > Bob=Caller
> >
> > Alice=Called
> >
> > John=Transfer party
> >
> >
> >
> > 1)    Bob calls Alice. The usual INVITE,Trying,200 OK, ACK.
> >
> > 2)    Alice transfers the call to John using Re-INVITE.
> >
> > a.    Alice calls John. The usual INVITE,Trying,200 OK, ACK.
> >
> > b.    Alice Re-INVITEs Bob using INVITE with adjusted SDP.
> >
> > 3)    Bob is connected to John through Alice in some magical way. I’m
> > guessing because the SDP has been changed and for some reason the RTP
> stream
> > flows between Bob and John through Alice?
> >
> >
> >
> > Is this correct? If not, perhaps someone could explain it to me from
> > scratch.
> >
> >
> >
> > Maybe useful to know that we are using Cisco equipment for call handling
> > (VXML and TCL scripts).
> >
> >
> >
> > Thanks,
> >
> >
> >
> > Grant
> >
> >
> > _______________________________________________
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users at lists.sip-router.org
> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
> >
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>



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