[SIPForum-discussion] Call transfer

[Moshe Ostrovsky]

Hi Shiba.

1)
180 is an optional response while the session is established (view the
section 14 of rfc 3261).

"A UAS MAY choose not to generate 180 (Ringing) responses for a re-
INVITE because UACs do not typically render this information to the
user."

2)
According to rfc 3262, UAS must send an 1xx-RPR (non-100) if the
"require" header is present (pay an attention that tis clause refers the
initial request, so I'm not sure that the same behavior will be applied
for re-Invite).

" The UAS MUST send any non-100 provisional response reliably if the
   initial request contained a Require header field with the option tag
   100rel."

3)
I saw the 180/183 response for Reinvite sending from GW in
sip-call-flows draft.

Hope it helps.

Moshe Ostrovsky
SIP-IMS team leader

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-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org] On Behalf Of shiba shankar rout
Sent: Saturday, January 17, 2009 10:21 AM
To: nabam serbang
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Call transfer

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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