Cisco 7960G Administration Manual page 151

Unified ip phone for release 8.0 (sip)
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Appendix B
SIP Call Flows
Figure B-20 Client, Server, or Global Error
User A
Step
Action
Setup—PBX A to Gateway 1
1.
INVITE—Gateway 1 to Cisco SIP IP phone
2.
3.
Call Proceeding—Gateway 1 to PBX A
100 Trying—Cisco SIP IP phone to Gateway 1
4.
OL-7890-01
GW1
PBX A
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
5. 4xx/5xx/6xx Failure
6. Disconnect
7. Release
8. ACK
9. Release Complete
Description
Call setup is initiated between PBX A and Gateway 1. Setup
includes the standard transactions that take place as A attempts to
call 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 IP phone. In the INVITE request:
Gateway 1 sends a Call Proceeding message to PBX A to
acknowledge the call setup request.
The phone sends a SIP 100 Trying response to Gateway 1. The
response indicates that the INVITE request has been received.
Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8.0 (SIP)
IP Network
The IP address of the 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 A is ready to receive is specified.
The port on which the gateway is prepared to receive the RTP
data is specified.
Call Flow Scenarios for Failed Calls
SIP IP Phone
User B
IP
B-51

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