[SR-Users] How does SIP transfer work?

[Dmytro Bogovych]

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
>
>
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