[SIPForum-discussion] Call transfer
shiba shankar rout
rout.shibashankar at gmail.com
Sat Jan 17 08:21:25 UTC 2009
Hello All,
I am getting stucked at this scenario. Can anybody help me on this ?
My question : Suppose i am sending re-Invite without SDP with
supported header 100rel to UAS. Then shall the UAS send 180/183
Ringing with SDP offer for enabling the PRACK?
Please clarify my doubt. My doubt is after a established session ,
shall i get 180 ringing for Re-invite. ?
Any suggestion would be vital for me ?
Invite---->
180------>
200----->
Ack----->
====
Re-Invite---->
180----------->
Prack----->
Does this call flow right ??
Regards,
S S Rout
On 1/16/09, nabam serbang <nabamserbang at yahoo.com> wrote:
>
> Seems you are sending some sort of line appearance selection feature in INV
> to C at step 5 or there is already one call with C.
>
> Whatever be the case, it doesn't really matter on which line call comes so
> long call work ( I suggest you to see if any line selection feature is being
> sent on step 5). And Your call flow MUST execute steps 6, 7 and 8 as well
> for complete transfer.
>
> Thanks
> ~Nabam
>
>
>
>
> ________________________________
> From: Naga Yerramsetti <naga95 at gmail.com>
> To: nabam serbang <nabamserbang at yahoo.com>
> Sent: Friday, January 16, 2009 12:44:05 AM
> Subject: Re: [SIPForum-discussion] Call transfer
>
> I am having problem at step 5. C is getting the invite on second line
> instead of getting connected on the first line.
>
>
> On Thu, Jan 15, 2009 at 1:48 AM, nabam serbang <nabamserbang at yahoo.com>
> wrote:
>
>
>
>
> A B2BUA B C
>
>
> dlg1 (A-B call)
> 1) <--------------------->
>
> Hold dlg A-B (When C preses tranfer, it generates soft hold to A)
>
> 2) <----------------------
>
> B --> C ( B establishes call with C) dlg2
> 3) <------------
> -------------------------->
>
> REFER (to C from B as B has initiated transfer to C)
> Refer-to:c@<c-address>?Replaces=dlg2
> 4)<---------------------
>
> INVITE sip:c@<c-address>
> Replaces:dlg2
> 5) ---------->-------------------------->
> 200 OK
> <---------------------------------------
>
> NOTIFY (for transfer)
> 6) --------->--------------->
>
> BYE (to B) dlg1
> 7) --------->----------->
>
> BYE (to C) dlg2
> 8) ------------------------->
>
>
>
>
> Steps 4 and 5 in above call flow should solve your problem. dlgx is one
> complete call setup steps i.e. from INVITE to ACK.
>
>
> Thanks and regards
> Nabam, Avaya
> Home page: http://sites.google.com/site/nabamserbang/
>
>
>
> ________________________________
> From: Naga Yerramsetti <naga95 at gmail.com>
> To: discussion at sipforum.org
> Sent: Wednesday, January 14, 2009 9:42:34 PM
> Subject: [SIPForum-discussion] Call transfer
>
>
>
> Hi,
> I am a newbie in SIP. We are using web logic sip server as B2BUA.
>
> I am having problem implementing call transfer. Call is established between
> A and B
> B is trying to transfer A to C. Call is established between B and C.
> Call leg to A and Call leg to B exists in one application session and a call
> leg to B and call leg to C exists in another sip application session.
> Now what the B2bUA (app running in weblogic sip server) should do to
> establish call between A and C
>
> Thank you.
> Naga
>
>
>
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