Subject: RE: [Sip] Call Transfer Scenario

From: "Bai, Gali" <[email-masked]>
Date: Wed, 20 Oct 2004 08:53:50 -0500

Anurag,

In the attended call transfer scenario, A-transferor needs to call
C-transfer target first and if C agrees to accept call from B, A will
send REFER with Replaces in its Refer-to header to B-transferee; And the
Invite from B to C also contains Replaces header to C.

So the differences are:
1. In attended call transfer, A calls C first and sends REFER with
Replaces in the Refer-to header; Unattended call transfer, A does not
need to call C, just sends REFER to B;
2. B Invite C, in attended call transfer, the Invite contains Replaces
header to replace the dialog between A and C; in unattended call
transfer, B Invite to C does not have Replaces header
3. If C receives Invite with Replaces header, it will be answered as an
attended call transfer, C needs to replace the existing dialog based on
Replaces header. 

Gali Bai

-----Original Message-----
From: [email-masked] [mailto:[email-masked]] On Behalf Of
Anurag Kabra
Sent: Tuesday, October 19, 2004 12:18 PM
To: [email-masked]
Subject: [Sip] Call Transfer Scenario

Hi

     I need to clear few of my doubts regarding call transfer, 

In a scenario of A, B and C - Unattended Transfer (Blind Transfer)

A = transferor
B = transferee
C = transfer target

1. There is a call set up between A and B
2.  A need to transfer to C - A puts the RTP session on Hold with
Re-INVITE 
and sends REFER with refer-to C, 
3. B on getting refer needs to put audio streaming on hold and so sends

Re-INVITE!! -- (DOUBT - is this step right)
4. B sends NOTIFY with subscription-state = active and expires to A
5. B now INVITES C 
6. On getting response its sends a NOTIFY to A with subscription-state =

terminated
(Need to know if the REFER expires because of expires in NOTIFY what can
be 
done ie if we get a time out on A side)
7. Now A sends a BYE to B
(What to do if B wants to terminate the dialog with CANCEL or BYE)

How can we differentiate Attended Transfer and Blind Transfer in the 
application??

Can anyone tell me free sip UA implementing Call Transfer?

Please do let me know any of your views. Please correct me if i am wrong

anywhere. 

Best Regards
  Anurag.

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_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip