REFER http://tools.ietf.org/html/rfc3515
On Thu, Nov 22, 2012 at 3:55 PM, Grant Bagdasarian <g...@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@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@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
