B-41
Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8.0 (SIP)
OL-7890-01
Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
Call from a Cisco SIP IP Phone to a Gateway Acting As a Backup Proxy in a SIP Network
Figure B-15
illustrates a successful call from a Cisco SIP IP phone to a gateway acting as a backup
proxy.
8.
INVITE—Phone B to Phone C
Phone B sends a SIP INVITE request to Phone C. The request is an invitation to
Phone B to participate in a call session. In the INVITE request:
•
The phone number of Phone B is inserted in the Request-URI field in the form
of a SIP URL. The SIP URL identifies the address of B and takes a form
similar to an e-mail address (
user
@
host,
where
user
is the telephone number
and
host
is either a domain name or a numeric network address). For example,
the Request-URI field in the INVITE request to C appears as “INVITE
sip:[email protected]; user=phone.” The “user=phone” parameter
distinquishes that the Request-URI address is a telephone number rather than
a username.
•
Phone B 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 B is ready to receive is specified.
9.
180 Ringing—Phone C to
Phone B
Phone C sends a SIP 180 Ringing response to Phone B.
10.
200 OK—Phone C to Phone B
Phone C sends a SIP 200 OK response to Phone B. The response notifies Phone B
that the connection has been made.
If Phone C supports the media capability advertised in the INVITE message sent
by Phone B, it advertises the intersection of its own and Phone B’s media
capability in the 200 OK response. If Phone C does not support the media
capability advertised by Phone B, it sends back a 400 Bad Request response with
a 304 Warning header field.
11.
ACK—Phone B to Phone C
Phone B sends a SIP ACK to Phone C. The ACK confirms that Phone B has
received the 200 OK response from Phone C.
The ACK might contain a message body with the final session description to be
used by Phone C. If the message body of the ACK is empty, Phone C uses the
session description in the INVITE request.
A two-way RTP channel is established between SIP IP Phone B and SIP IP Phone C.
12.
INVITE—Phone B to Phone A
User B takes User A off hold. Phone B sends a SIP INVITE request to Phone A
with the same call ID as the previous INVITE and a new SDP attribute parameter
(sendrecv), which is used to reestablish the call.
13.
200 OK—Phone A to Phone B
Phone A sends a SIP 200 OK response to Phone B.
A is taken off hold. The RTP channel 1 between A and B is reestablished.
Phone B acts as a bridge mixing the RTP channel between A and B with the channel between B and C, establishing a
conference bridge between A and C.
Step
Action
Description
