Appendix B SIP Call Flows

 

 

Call Flow Scenarios for Successful Calls

 

 

 

 

Step

Action

Description

 

 

 

9

180 Ringing—Cisco SIP IP

Cisco SIP IP phone C sends a SIP 180 Ringing response to

 

phone C to Cisco SIP IP phone B

Cisco SIP IP phone B.

 

 

 

10

200 OK—Cisco SIP IP phone C

Cisco SIP IP phone C sends a SIP 200 OK response to

 

to Cisco SIP IP phone B

Cisco SIP IP phone B. The 200 OK response notifies Cisco

 

 

SIP IP phone B that the connection has been made.

 

 

If Cisco SIP IP phone B supports the media capability

 

 

advertised in the INVITE message sent by Cisco SIP IP

 

 

phone A, it advertises the intersection of its own and Cisco

 

 

SIP IP phone A’s media capability in the 200 OK response.

 

 

If Cisco SIP IP phone B does not support the media

 

 

capability advertised by Cisco SIP IP phone A, it sends

 

 

back a 400 Bad Request response with a 304 Warning

 

 

header field.

 

 

 

11

ACK—Cisco SIP IP phone B to

Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP

 

Cisco SIP IP phone C

phone C. The ACK confirms that Cisco SIP IP phone B has

 

 

received the 200 OK response from Cisco SIP IP phone C.

 

 

The ACK might contain a message body with the final

 

 

session description to be used by Cisco SIP IP phone C. If

 

 

the message body of the ACK is empty, Cisco SIP IP phone

 

 

C uses the session description in the INVITE request.

 

 

 

 

A two-way RTP channel is established between Cisco SIP IP phone B and Cisco SIP IP phone C.

12

BYE—Cisco SIP IP phone B to

The call continues and then User B hangs up. Cisco SIP IP

 

Cisco SIP IP phone C

phone B sends a SIP BYE request to Cisco SIP IP phone C.

 

 

The BYE request indicates that User B wants to release the

 

 

call.

 

 

 

13

200 OK—Cisco SIP IP phone C

Cisco SIP IP phone C sends a SIP 200 OK message to

 

to Cisco SIP IP phone B

Cisco SIP IP phone B. The 200 OK response notifies Cisco

 

 

SIP IP phone B that the BYE request has been received.

 

 

The call session between User A and User B is now

 

 

terminated.

 

 

 

The RTP channel between Cisco SIP IP phone B and Cisco SIP IP phone C is torn down.

 

 

Cisco SIP IP Phone 7960 Administrator Guide

 

 

 

 

 

 

78-10497-02

 

 

B-39

 

 

 

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Cisco Systems IP phone 7960 manual Appendix B SIP Call Flows