B-46
Cisco SIP IP Phone Administrator Guide
Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
Call Flow Scenarios for Failed Calls
This section describes call flows for the following scenarios, which illustrate unsuccessful calls:
•
Gateway-to-Cisco SIP IP Phone—Called User Is Busy, page B-46
•
Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer, page B-48
•
Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error, page B-50
•
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Is Busy, page B-51
•
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer, page B-52
•
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Authentication Error, page B-53
Gateway-to-Cisco SIP IP Phone—Called User Is Busy
Figure B-14
illustrates an unsuccessful call in which User A initiates a call to User B, but User B is on
the phone and is unable or unwilling to take another call.
12.
INVITE—Cisco SIP IP phone B
to Cisco SIP IP phone A
Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with the
same call ID as the previous INVITE and new SDP session parameters
(IP address), which are used to reestablish the call.
Call_ID=1
SDP: c=IN IP4 181.23.250.2
To reestablish the call between phone A and phone B, the IP address of phone B
is inserted into the c= SDP field.
13.
200 OK—Cisco SIP IP phone A
to Cisco SIP IP phone B
Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
14.
ACK—Cisco SIP IP phone B to
Cisco SIP IP phone A
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK
confirms that Cisco SIP IP phone B has received the 200 OK response from
Cisco SIP IP phone A.
SIP IP phone B acts as a bridge mixing the RTP channel between User A and User B with the channel between User B and
User C; establishing a conference bridge between User A and User C.
Step
Action
Description
