SIP Call Flows: Call Flow Scenarios For Successful Calls, Page B-2 Call Flow Scenarios For Failed Calls, Page B-47
SIP Call Flows: Call Flow Scenarios For Successful Calls, Page B-2 Call Flow Scenarios For Failed Calls, Page B-47
Note If you have enabled the rfc_2543_hold parameter, the phone will use the RFC 2543 method for putting
a call on hold, and will set the media address to 0.0.0.0. The examples in this chapter assume that this
parameter is not enabled, and show the phone using the RFC 3264 method.
For example, if the rfc_2543_hold parameter is enabled, the INVITE request in step 5 in the “Simple
Call Hold” section on page B-9 would be sent as INVITE (c=IN IP4 0.0.0.0 a=inactive).
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
SIP IP Phone
User A PBX A GW1 IP Network User B
IP
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
11. BYE
12. Disconnect
13. Release
14. 200 OK
15. Release Complete
41724
Step Action Description
1. Setup—PBX A to Gateway 1 Call setup is initiated between PBX A and Gateway 1. Setup includes
the standard transactions that take place as User A attempts to call
User B.
2. INVITE—Gateway 1 to Cisco SIP IP phone 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:
• 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 that User A is ready to receive is specified.
• The port on which the gateway is prepared to receive the RTP
data is specified.
3. Call Proceeding—Gateway 1 to PBX A Gateway 1 sends a Call Proceeding message to PBX A to
acknowledge the Call Setup request.
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Call Flow Scenarios for Successful Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
SIP IP Phone
User A PBX A GW1 IP Network User B
IP
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
5. 180 Ringing
6. Alerting
7. 200 OK
8. Connect
9. ACK
10. Connect ACK
12. 200 OK
13. ACK
15. 200 OK
16. ACK
137201
2-way voice path 2-way VP
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value
places the call on hold.
12. 200 OK—Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP 200 OK response to the phone. The response
notifies the phone that the INVITE was successfully processed.
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Call Flow Scenarios for Successful Calls
Figure B-3 Successful Call from Cisco SIP IP Phone to SIP Gateway (Emergency Proxy)
SIP IP Phone
User A Proxy GW PBX User B
IP
11. BYE
12. Disconnect
13. Release
14. 200 OK
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
SIP IP SIP IP
Phone User A IP Network Phone User B
IP IP
1. INVITE B
2. 180 RINGING
3. 200 OK
4. ACK
5. INVITE (a=sendonly)
6. 200 OK
7. ACK
8. INVITE (a=sendrecv)
9. 200 OK
10. ACK
137202
Step Action Description
1. INVITE—Phone A to Phone B Phone A sends a SIP INVITE request to Phone B. The request is an invitation
to User B to participate in a call session. In the INVITE request:
• The phone number of User B is inserted in the Request-URI field in the
form of a SIP URL. The SIP URL identifies the address of User 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
User B appears as “INVITE sip:555-0199@companyb.com; user=phone.”
The “user=phone” parameter distinquishes that the Request-URI address
is a telephone number rather than a username.
• Phone 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.
2. 180 Ringing—Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A.
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the
call on hold.
6. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
7. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
The RTP channel between Phone A and Phone B is torn down.
8. 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.
9. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
10. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
A two-way RTP channel is reestablished between Phone A and Phone B.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
SIP IP SIP IP
SIP IP Phone User B Phone
Phone User A IP Network User C
IP IP IP
1. INVITE B
2. 180 Ringing
3. 200 OK
4. ACK
5. INVITE (a=sendonly)
6. 200 OK
7. ACK
8. INVITE C
A is put on hold. The RTP channel between A and B is torn down.
9. 180 Ringing
10. 200 OK
11. ACK
12. BYE
13. 200 OK
16. ACK
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places
the call on hold.
6. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
7. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
The RTP channel between Phone A and Phone B is torn down.
8. INVITE—Phone B to Phone C Phone B sends a SIP INVITE request to Phone C. The request is an
invitation to User C to participate in a call session.
9. 180 Ringing—Phone C to Phone B Phone C sends a SIP 180 Ringing response to Phone B.
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Call Flow Scenarios for Successful Calls
Call Waiting
Figure B-6 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call,
and one of the participants receives a call from a third party and then returns to the original call. In this
call flow scenario, the end users are User A, User B, and User C. They are all using
Cisco IP Phone 7960G/7940G, which are connected using an IP network.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User C calls User B.
4. User B accepts the call from User C.
5. User B switches back to User A.
6. User B hangs up, ending the call with User A.
7. User B is notified of the remaining call with User C.
8. User B answers the notification and continues the call with User C.
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2. 180 Ringing
3. 200 OK
4. ACK
6. 180 Ringing
7. INVITE (a=sendonly)
8. 200 OK
9. ACK
10. 200 OK
A is put on hold. The RTP channel between A and B is torn down.
11. ACK
13. 200 OK
14. ACK
15. INVITE (a=sendrecv) C is on hold. The RTP channel
between B and C is torn down.
16. 200 OK
17. ACK
18. BYE
19. 200 OK
20. INVITE (a=sendrecv)
B has disconnected from A, but the call with C (on hold) remains.
21. 200 OK
22. ACK
C is taken off hold. The RTP
channel between B and C is 137210
reestablished.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places
the call on hold.
8. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
9. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
The RTP channel between Phone A and Phone B is torn down.
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places
the call on hold.
13. 200 OK—Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B.
14. 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 RTP channel between Phone B and Phone C is torn down.
15. 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.
16. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
17. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
A two-way RTP channel is reestablished between Phone A and Phone B.
18. BYE—Phone B to Phone A The call continues and then User B hangs up. Phone B sends a SIP BYE
request to Phone A. The request indicates that User B wants to release the
call.
19. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B. The response notifies
Phone B that the BYE request has been received. The call session between
User A and User B terminates.
The RTP channel between Phone A and Phone B is torn down.
20. INVITE—Phone B to Phone C User B takes User C off hold. Phone B sends a SIP INVITE request to
Phone C with the same call ID as the previous INVITE (sent to Phone C) and
a new SDP attribute parameter (sendrecv), which is used to reestablish the
call.
21. 200 OK—Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B.
22. 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 A.
A two-way RTP channel is reestablished between Phone B and Phone C.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
IP IP Network IP IP
1. INVITE
2. 100 TRYING
3. 180 RINGING
4. 200 OK
5. ACK
7. 200 OK
8. ACK
User B dials user C.
9. REFER (Refer-To: C,Referred-By: B)
12. BYE
13. 200 OK
17. 200 OK
18. ACK
20. 200 OK
137204
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the
call on hold.
7. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
8. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
User B dials User C.
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
IP IP Network IP IP
1. INVITE
2. 100 TRYING
3. 180 RINGING
4. 200 OK
5. ACK
7. 200 OK
8. ACK
User B dials user C.
9. REFER (Refer-To: C,Referred-By: B)
11. BYE(Also: C)
12. 200 OK
16. 200 OK
17. ACK
137205
2-way voice path
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the call
on hold.
7. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
8. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
User B dials User C.
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
IP IP Network IP IP
1. INVITE
2. 100 TRYING
3. 180 RINGING
4. 200 OK
5. ACK
7. 200 OK
8. ACK
9. INVITE C
12. 200 OK
13. ACK
User B presses transfer. 2- way voice path
15. 200 OK
16. ACK
17. REFER (Refer-To: C,
Replaces: B,Referred-By: B)
21. 200 OK
22. ACK
23. BYE
24. 200 OK
26. 200 OK
27. BYE
28. 200 OK
137206
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the call
on hold.
7. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
8. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
User B dials User C.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the call
on hold.
15. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
16. 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.
17. REFER—Phone B to Phone A Phone B sends a REFER message to Phone A. The message contains the
following information:
• Refer-To: C
• Replaces: B
• Referred-By: B
The message indicates that the user (recipient) should contact a third party for use
in transferring parties.
18. 202 ACCEPTED—Phone A to Phone A sends a SIP 202 ACCEPTED message to Phone B. The confirms that the
Phone B REFER message has been received.
19. NOTIFY (Event:Refer; Phone A sends a NOTIFY message to Phone B. This message notifies B that the
Subscription-State: Active) REFER processing has started.
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Call Flow Scenarios for Successful Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
IP IP Network IP IP
1. INVITE
2. 100 TRYING
3. 180 RINGING
4. 200 OK
5. ACK
7. 200 OK
8. ACK
9. INVITE C
12. 200 OK
13. ACK
User B presses transfer.
2-way voice path
15. 200 OK
16. ACK
17. REFER (Refer-To: C,
Replaces: B,Referred-By: B)
19. BYE(Also: C)
20. 200 OK
21. BYE
22. 200 OK
26. 200 OK
27. ACK
137207
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the
call on hold.
15. 200 OK—Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B.
16. 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.
17. REFER—Phone B to Phone A Phone B sends a REFER message to Phone A. The message contains the
following information:
• Refer-To: C
• Replaces: B
• Referred-By: B
The message indicates that the user (recipient) should contact a third party for
use in transferring parties.
18. 501 Not Implemented—Cisco SIP Phone A sends a 501 Not Implemented message to Phone B. The message
IP Phone A to Cisco SIP IP indicates that the REFER message is not supported and that Phone B should
Phone B failover to Bye/Also.
19. BYE—Phone B to Phone A Phone B sends a BYE message to Phone A. The message includes the following
information:
• Also: C
The message indicates that the 501 Not Implemented message was received in
response to a REFER message.
20. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B. The response notifies
Phone B that the BYE request has been received.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
IP Network
IP IP IP
SIP IP Proxy Redirect SIP IP SIP IP
Phone Server Server Phone Phone
User A User B User C
1. INVITE B
2. INVITE B
4. ACK
5. INVITE C
6. 180 Ringing
7. 200 OK
8. 200 OK
9. ACK
10. ACK
41471
Step Action Description
1. INVITE—Phone A to SIP proxy Phone A sends a SIP INVITE request to the SIP proxy server. The request is an
server invitation to User B to participate in a call session. In the INVITE request:
• The phone number of User B is inserted in the Request-URI field in the form
of a SIP URL. The SIP URL identifies the address of User 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 User B appears as “INVITE
sip:555-0199@companyb.com; user=phone.” The “user=phone” parameter
distinquishes that the Request-URI address is a telephone number rather than
a username.
• Phone 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.
2. INVITE—SIP proxy server to The SIP proxy server sends the SIP INVITE request to the SIP redirect server.
SIP redirect server
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Call Flow Scenarios for Successful Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
IP Network
IP IP IP
SIP IP Proxy Redirect SIP IP SIP IP
Phone Server Server Phone Phone
User A User B User C
1. INVITE B
2. INVITE B
4. ACK
5. INVITE B
7. ACK
8. INVITE C
9. 180 Ringing
10. 200 OK
11. 200 OK
12. ACK
13. ACK
2-way RTP channel
41472
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Call Flow Scenarios for Successful Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
IP Network
IP IP IP
SIP IP Proxy Redirect SIP IP SIP IP
Phone Server Server Phone Phone
User A User B User C
1. INVITE B
2. INVITE B
4. ACK
5. INVITE B
6. 180 Ringing
7. 180 Ringing
8. CANCEL
9. 200 OK
10. INVITE C
12. 200 OK
13. 200 OK
14. ACK
15. ACK
2-way RTP channel
41473
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Call Flow Scenarios for Successful Calls
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Call Flow Scenarios for Successful Calls
Three-Way Calling
Figure B-14 illustrates successful three-way calling between Cisco SIP IP phones in which User B mixes
two RTP channels and therefore establishes a conference bridge between User A and User C. In this call
flow scenario, the end users are User A, User B, and User C. They are all using Cisco Unified IP Phone
7960G and 7940G models, which are connected using an IP network.
The call flow scenario is as follows:
1. User A calls User B.
2. User B answers the call.
3. User B puts User A on hold.
4. User B calls User C.
5. User C answers the call.
6. User B takes User A off hold.
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Call Flow Scenarios for Successful Calls
IP Network
IP IP IP
SIP IP Proxy Redirect SIP IP SIP IP
Phone Server Server Phone Phone
User A User B User C
1. INVITE B (Call-ID=1)
2. 180 Ringing
3. 200 OK
4. ACK
6. 200 OK
7. ACK
User A is on hold. The RTP channel 1 between User A and B is torn down. 8. INVITE C (Call-ID=2)
9. 180 Ringing
10. 200 OK
11. ACK
2-way RTP channel 2
between User B and
C is established
12. INVITE A (Call-ID=1, a=sendrecv)
13. 200 OK
User A is taken off hold. The RTP channel 1 between User A and B
is re-established.
137208
User B mixes the RTP channels 1 and 2 and establishes a conference brigde between Users A and C
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
The a= SDP field of the SIP INVITE contains sendonly. This value places the call
on hold.
6. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B.
7. ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has
received the 200 OK response from Phone A.
The RTP channel between Phone A and Phone B is torn down. A is put on hold.
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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.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure B-15 Call from a Cisco SIP IP Phone to a Gateway Acting As a Backup Proxy
SIP IP Phone
User A Primary GW PBX User B
IP
1. INVITE
2. INVITE (retry)
3. INVITE (retry)
4. INVITE (retry)
5. INVITE (retry)
6. INVITE (retry)
7. INVITE (retry)
8. INVITE (retry)
9. Setup
10. Call Proceeding
11. 100 Trying
12. Alerting
13. 180 Ringing
14. Connect
15. 200 OK
16. ACK
17. Connect ACK
2-way voice path
18. BYE
19. Disconnect
20. Release
62069
21. 200 OK
22. Release Complete
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Call Flow Scenarios for Successful Calls
Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Backup Proxy
Figure B-16 illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone that uses a SIP
backup proxy.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure B-16 A Successful Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Backup
Proxy
IP IP
1. INVITE
2. INVITE (retry)
3. INVITE (retry)
4. INVITE (retry)
5. INVITE (retry)
6. INVITE (retry)
7. INVITE (retry)
8. INVITE (retry)
9. 100 Trying
10. INVITE
11. 100 Trying
12. 180 Ringing
13. 180 Ringing
14. 200 OK
15. 200 OK
16. ACK
17. ACK
2-way voice path
18. BYE
19. BYE
20. 200 OK 62068
21. 200 OK
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Call Flow Scenarios for Successful Calls
Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Emergency Proxy
Figure B-17 illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone via an
emergency proxy. B is the extension of the dial template with the “Route” attribute as “emergency” in
the dialplan.xml file.
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Appendix B SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure B-17 Successful Call from a Cisco IP Phone to a Cisco IP Phone Using a SIP Emergency Proxy
IP IP
11. BYE
12. BYE
13. 200 OK
14. 200 OK
62071
Step Action Description
1. INVITE—Cisco SIP IP phone The phone tries to connect to the emergency proxy by sending out the INVITE
(A) to emergency proxy message. The dial template for the emergency route is matched.
2. 100 Trying—Emergency proxy The emergency proxy sends a SIP 100 Trying response to A. The response
to Cisco SIP IP phone (A) indicates that the INVITE request has been received.
3. INVITE—Emergency proxy to The backup proxy tries to connect to B by sending out the INVITE message.
Cisco SIP IP phone (B)
4. 100 Trying—Cisco SIP IP phone User B sends a SIP 100 Trying response to the emergency proxy. The response
(B) to emergency proxy indicates that the INVITE request has been received.
5. 180 Ringing—Cisco SIP IP User B sends a SIP 180 Ringing response to the emergency proxy. The response
phone (B) to emergency proxy indicates that User B is being alerted.
6. 180 Ringing—Emergency proxy The emergency proxy sends a SIP 180 Ringing response to User A. The response
to Cisco SIP IP phone (A) indicates that the emergency proxy is being alerted.
7. 200 OK—Cisco SIP IP phone User B sends a SIP 200 OK response to the emergency proxy. The response
(B) to emergency proxy notifies the emergency proxy that the connection has been made.
8. 200 OK—Emergency proxy to The emergency proxy sends a SIP 200 OK response to User A. The response
Cisco SIP IP phone (A) notifies User A that the connection has been made.
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Call Flow Scenarios for Failed Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
SIP IP Phone
User A PBX A GW1 IP Network User B
IP
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
41725
Step Action Description
1. Setup—PBX A to Gateway 1 Call setup is initiated between PBX A and Gateway 1. Setup includes the
standard transactions that take place as A attempts to call B.
2. INVITE—Gateway 1 to Cisco SIP IP Gateway 1 maps the SIP URL phone number to a dial peer. The dial peer
phone 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:
• 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.
3. Call Proceeding—Gateway 1 to PBX A Gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Call Setup request.
4. 100 Trying—Cisco SIP IP phone to The phone sends a SIP 100 Trying response to Gateway 1. The response
Gateway 1 indicates that the INVITE request has been received.
5. 486 Busy Here—Cisco SIP IP phone to The phone sends a SIP 486 Busy Here response to Gateway 1. The
Gateway 1 response is a client error response that indicates that B was successfully
contacted but that B was not willing or was unable to take the call.
6. Disconnect (Busy)—Gateway 1 to Gateway 1 sends a Disconnect message to PBX A.
PBX A
7. Release—PBX A to Gateway 1 PBX A sends a Release message to Gateway 1.
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
SIP IP Phone
User A PBX A GW1 IP Network User B
IP
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
5. 180 Ringing
6. Alerting
7. CANCEL
8. Disconnect
9. Release
10. 200 OK
11. Release Complete
41726
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
SIP IP Phone
User A PBX A GW1 IP Network User B
IP
1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying
5. 4xx/5xx/6xx Failure
6. Disconnect
7. Release
8. ACK
9. Release Complete
41727
Step Action Description
1. Setup—PBX A to Gateway 1 Call setup is initiated between PBX A and Gateway 1. Setup
includes the standard transactions that take place as A attempts to
call B.
2. INVITE—Gateway 1 to Cisco SIP IP phone 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:
• 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.
3. Call Proceeding—Gateway 1 to PBX A Gateway 1 sends a Call Proceeding message to PBX A to
acknowledge the call setup request.
4. 100 Trying—Cisco SIP IP phone to Gateway 1 The phone sends a SIP 100 Trying response to Gateway 1. The
response indicates that the INVITE request has been received.
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
SIP IP SIP IP
Phone User A IP Network Phone User B
IP IP
1. INVITE B
3. ACK
41475
Step Action Description
1. INVITE—Phone A to Phone B Phone A sends a SIP INVITE request to Phone B. The request is an invitation to
User B to participate in a call session. In the INVITE request:
• The phone number of User 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 B appears as “INVITE
sip:555-0199@companyb.com; user=phone.” The “user=phone” parameter
distinquishes that the Request-URI address is a telephone number rather than
a username.
• Phone 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.
2. 486 Busy Here—Phone B to Phone B sends a 486 Busy Here message to the Phone A. The message indicates
Phone A that Phone B is in use and the user is not willing or able to take additional calls.
3. ACK—Phone A to Phone B Phone A sends a SIP ACK to the Phone B. The ACK confirms that Phone A has
received the 486 Busy Here response from Phone B.
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Appendix B SIP Call Flows
Call Flow Scenarios for Failed Calls
SIP IP SIP IP
Phone User A IP Network Phone User B
IP IP
1. INVITE B
2. 180 Ringing
3. CANCEL
4. 200 OK
137209
6. ACK
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Call Flow Scenarios for Failed Calls
Authentication Error
Figure B-23 illustrates an unsuccessful call in which User A initiates a call to User B but is prompted
for authentication credentials by the proxy server. User A’s SIP IP phone then reinitiates the call with a
SIP INVITE request that includes authentication credentials.
IP IP
1. INVITE B
3. ACK
4. Resend INVITE B
41477
Step Action Description
1. INVITE—Phone A to SIP proxy Phone A sends a SIP INVITE request to the SIP proxy server. The request is an
server invitation to B to participate in a call session. In the INVITE request:
• The phone number of B is inserted in the Request-URI field in the form of a
SIP URL. The SIP URL identifies the address of User 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 User B appears as “INVITE
sip:555-0199@companyb.com; user=phone.” The “user=phone” parameter
distinquishes that the Request-URI address is a telephone number rather than
a username.
• Phone 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.
2. 407 Authentication Error—SIP The SIP proxy server sends a SIP 407 Authentication Error response to Phone A.
proxy server to Phone A
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Call Flow Scenarios for Failed Calls
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