Cisco SIP IP Phone Administrator's Manual page 113

Appendix B SIP Call Flows
Figure B-1
User A
Step Action
1. Setup—PBX A to Gateway 1
2. INVITE—Gateway 1 to
Cisco SIP IP phone
3. Call Proceeding—Gateway 1 to
PBX A
Gateway-to-Cisco SIP IP Phone—Successful Setup and Disconnect
PBX A GW1
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
5. 180 Ringing
6. Alerting
7. 200 OK
8. Connect
9. Connect ACK
10. ACK
2-way voice path 2-way RTP channel
11. BYE
12. Disconnect
13. Release
14. 200 OK
15. Release Complete
Description
Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes
the standard transactions that take place as User A attempts to call User B.
Gateway 1 maps the SIP URL phone number to a dial peer. The dial peer includes
the IP address and the port number of the SIP-enabled entity to contact. Gateway 1
sends a SIP INVITE request to the address it receives as the dial peer, which, in
this scenario, is the Cisco SIP IP phone.
In the INVITE request:
The IP address of the Cisco SIP IP phone is inserted in the Request-URI field.
•
• PBX A is identified as the call session initiator in the From field.
A unique numeric identifier is assigned to the call and is inserted in the
•
Call-ID field.
The transaction number within a single call leg is identified in the CSeq field.
•
• The media capability User A is ready to receive is specified.
• The port on which the Gateway is prepared to receive the RTP data is
specified.
Gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call
Setup request.
Call Flow Scenarios for Successful Calls
SIP IP Phone
IP Network
Cisco SIP IP Phone Administrator Guide
User B
IP
B-3