Table Of Contents
SIP Call Flows
Call Flow Scenarios for Successful Calls
Gateway to Cisco SIP IP Phone
Call Setup and Disconnect
Call Setup and Hold
Call to a Gateway Acting as an Emergency Proxy from a Cisco SIP IP Phone
Cisco SIP IP Phone to Cisco SIP IP Phone
Simple Call Hold
Call Hold with Consultation
Call Waiting
Call Transfer Without Consultation
Call Transfer Without Consultation Using Failover
Call Transfer with Consultation
Call Transfer with Consultation Using Failover
Network Call Forwarding (Unconditional)
Network Call Forwarding (Busy)
Network Call Forwarding (No Answer)
Three-Way Calling
Call from a Cisco SIP IP Phone to a Gateway Acting as a Backup Proxy
Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone By Way of a Backup Proxy
Call from Cisco SIP IP Phone to Cisco SIP IP Phone Using an Emergency Proxy
Call Flow Scenarios for Failed Calls
Gateway to Cisco SIP IP Phone
Called User Is Busy
Called User Does Not Answer
Client, Server, or Global Error
Cisco SIP IP Phone to Cisco SIP IP Phone
Called User Is Busy
Called User Does Not Answer
Authentication Error
SIP Call Flows
This appendix includes the following sections:
•
Call Flow Scenarios for Successful Calls
•Call Flow Scenarios for Failed Calls
SIP uses the following request methods:
•INVITE—Indicates that a user or service is being invited to participate in a call session.
•ACK—Confirms that the client has received a final response to an INVITE request.
•BYE—Terminates a call and can be sent by either the caller or the callee.
•CANCEL—Cancels any pending searches but does not terminate a call that has already been accepted.
•OPTIONS—Queries the capabilities of servers.
•REGISTER—Registers the address listed in the To header field with a SIP server.
•REFER—Indicates that the user (recipient) should contact a third party for use in transferring parties.
•NOTIFY—Notifies the user of the status of a transfer using REFER. Also used for remote reset.
The following types of responses are used by SIP and generated by the Cisco SIP gateway:
•SIP 1xx—Informational Responses
•SIP 2xx—Successful Responses
•SIP 3xx—Redirection Responses
•SIP 4xx—Client Failure Responses
•SIP 5xx—Server Failure Responses
•SIP 6xx—Global Failure Responses
Call Flow Scenarios for Successful Calls
This section describes successful call flows scenarios, which are as follows:
•Gateway to Cisco SIP IP Phone
•Cisco SIP IP Phone to Cisco SIP IP Phone
Gateway to Cisco SIP IP Phone
The following scenarios describe and illustrate successful calls in a gateway to a Cisco SIP IP phone:
•Call Setup and Disconnect
•Call Setup and Hold
•Call to a Gateway Acting as an Emergency Proxy from a Cisco SIP IP Phone
Call Setup and Disconnect
Figure B-1 illustrates a successful phone call setup and disconnect. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
The call flow is as follows:
1. User A calls User B.
2. User B answers the call.
3. User B hangs up.
Figure B-1 Successful Setup and Disconnect
Step
|
Action
|
Description
|
1.
|
Setup—PBX A to Gateway 1
|
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.
|
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 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.
|
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 Cisco SIP IP phone sends a SIP 100 Trying response to Gateway 1. The 100 Trying response indicates that the INVITE request has been received by the Cisco SIP IP phone.
|
5.
|
180 Ringing—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP 180 Ringing response to Gateway 1. The 180 Ringing response indicates that the user is being alerted.
|
6.
|
Alerting—Gateway 1 to PBX A
|
Gateway 1 sends an Alert message to User A. The Alert message indicates that Gateway 1 has received a 180 Ringing response from the Cisco SIP IP phone. User A hears the ringback tone that indicates that User B is being alerted.
|
7.
|
200 OK—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1. The 200 OK response notifies Gateway 1 that the connection has been made.
|
8.
|
Connect—Gateway 1 to PBX A
|
Gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
|
9.
|
Connect ACK—PBX A to Gateway 1
|
PBX A acknowledges Gateway 1's Connect message.
|
10.
|
ACK—Gateway 1 to Cisco SIP IP phone
|
Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The ACK confirms that Gateway 1 has received the 200 OK response. The call session is now active.
|
11.
|
BYE—Cisco SIP IP phone to Gateway 1
|
User B terminates the call session at his Cisco SIP IP phone and the phone sends a SIP BYE request to Gateway 1. The BYE request indicates that User B wants to release the call.
|
12.
|
Disconnect—Gateway 1 to PBX A
|
Gateway 1 sends a Disconnect message to PBX A.
|
13.
|
Release—PBX A to Gateway 1
|
PBX A sends a Release message to Gateway 1.
|
14.
|
200 OK—Gateway 1 to Cisco SIP IP phone
|
Gateway 1 sends a SIP 200 OK response to the Cisco SIP IP phone. The 200 OK response notifies the phone that Gateway 1 has received the BYE request.
|
15.
|
Release Complete—Gateway 1 to PBX A
|
Gateway 1 sends a Release Complete message to PBX A and the call session is terminated.
|
Call Setup and Hold
Figure B-2 illustrates a successful phone call setup and call hold. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
The call flow 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 takes User A off hold.
Figure B-2 Successful Call Setup and Hold
Step
|
Action
|
Description
|
1.
|
Setup—PBX A to Gateway 1
|
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.
|
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 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.
|
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 Cisco SIP IP phone sends a SIP 100 Trying response to Gateway 1. The 100 Trying response indicates that the INVITE request has been received by the Cisco SIP IP phone.
|
5.
|
180 Ringing—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP 180 Ringing response to Gateway 1. The 180 Ringing response indicates that the user is being alerted.
|
6.
|
Alerting—Gateway 1 to PBX A
|
Gateway 1 sends an Alert message to User A. The Alert message indicates that Gateway 1 has received a 180 Ringing response from the Cisco SIP IP phone. User A hears the ringback tone that indicates that User B is being alerted.
|
7.
|
200 OK—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1. The 200 OK response notifies Gateway 1 that the connection has been made.
|
8.
|
Connect—Gateway 1 to PBX A
|
Gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
|
9.
|
ACK—Gateway 1 to Cisco SIP IP phone
|
Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The ACK confirms that User A has received the 200 OK response. The call session is now active.
|
10.
|
Connect ACK—PBX A to Gateway 1
|
PBX A acknowledges Gateway 1's Connect message.
|
11.
|
INVITE—Cisco SIP IP phone to Gateway 1
|
User B puts User A on hold. The Cisco SIP IP phone sends a SIP INVITE request to Gateway 1.
|
12.
|
200 OK—Gateway 1 to Cisco SIP IP phone
|
Gateway 1 sends a SIP 200 OK response to the Cisco SIP IP phone. The 200 OK response notifies the Cisco SIP IP phone that the INVITE was successfully processed.
|
13.
|
ACK—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP ACK to Gateway 1. The ACK confirms that the Cisco SIP IP phone has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
|
14.
|
INVITE—Cisco SIP IP phone to Gateway 1
|
User B takes User A off hold. The Cisco SIP IP phone sends a SIP INVITE request to Gateway 1.
|
15.
|
200 OK—Gateway 1 to Cisco SIP IP phone
|
Gateway 1 sends a SIP 200 OK response to the Cisco SIP IP phone. The 200 OK response notifies the Cisco SIP IP phone that the INVITE was successfully processed.
|
16.
|
ACK—Cisco SIP IP phone to Gateway 1
|
The Cisco SIP IP phone sends a SIP ACK to Gateway 1. The ACK confirms that the Cisco SIP IP phone has received the 200 OK response. The call session is now active.
|
Call to a Gateway Acting as an Emergency Proxy from a Cisco SIP IP Phone
Figure B-3 illustrates a successful call from a Cisco SIP IP phone to a gateway acting as an emergency proxy.
Figure B-3 Successful Call from Cisco SIP IP Phone to Gateway (Emergency Proxy)
Step
|
Action
|
Description
|
1.
|
INVITE—Cisco SIP IP phone to gateway (emergency proxy)
|
Cisco SIP IP phone tries to connect to the gateway (emergency proxy) by sending out the INVITE message. The dial template for the emergency route is matched.
|
2.
|
Setup—Gateway to PBX
|
Call setup is initiated between the gateway and PBX. The call setup includes the standard transactions that take place as User A attempts to call User B.
|
3.
|
Call Proceeding—PBX to gateway
|
PBX sends a Call Proceeding message to gateway to acknowledge the Call Setup request.
|
4.
|
100 Trying—Gateway to
Cisco SIP IP phone (User A)
|
Gateway sends a SIP 100 Trying response to User A. The 100 Trying response indicates that the INVITE request has been received by the gateway.
|
5.
|
Alerting—PBX to gateway
|
PBX sends an Alert message to the gateway. The Alert message indicates that the PBX has received a 100 Trying Ringing response from the gateway.
|
6.
|
180 Ringing—Gateway to
Cisco SIP IP phone (User A)
|
The gateway sends a SIP 180 Ringing response to User A. The 180 Ringing response indicates that the gateway is being alerted.
|
7.
|
Connect—PBX to gateway
|
PBX sends a Connect message to gateway. The Connect message notifies the gateway that the connection has been made.
|
8.
|
200 OK—Gateway to Cisco SIP IP phone (User A)
|
Gateway sends a SIP 200 OK response to the User A. The 200 OK response notifies User A that the connection has been made.
|
9.
|
ACK—Cisco SIP IP phone
(User A) to gateway
|
User A sends a SIP ACK to the gateway. The ACK confirms that User A has received the 200 OK response. The call session is now active.
|
10.
|
Connect ACK—Gateway to PBX
|
Gateway acknowledges PBX's Connect message.
|
11.
|
BYE—Cisco SIP IP phone
(User A) to gateway
|
User A terminates the call session and sends a SIP BYE request to gateway. The BYE request indicates that User A wants to release the call.
|
12.
|
Disconnect—Gateway to PBX
|
Gateway sends a Disconnect message to PBX.
|
13.
|
Release—PBX to gateway
|
PBX sends a Release message to the gateway.
|
14.
|
200 OK—Gateway to Cisco SIP IP phone (User A)
|
Gateway sends a SIP 200 OK response to User A. The 200 OK response notifies User A that the gateway has received the BYE request.
|
15.
|
Release Complete—Gateway to PBX
|
Gateway sends a Release Complete message to the PBX and the call session is terminated.
|
Cisco SIP IP Phone to Cisco SIP IP Phone
The following sections describe and illustrate successful calls from Cisco SIP IP phone to Cisco SIP IP phone:
•Simple Call Hold
•Call Hold with Consultation
•Call Waiting
•Call Transfer Without Consultation
•Call Transfer Without Consultation Using Failover
•Call Transfer with Consultation
•Call Transfer with Consultation Using Failover
•Network Call Forwarding (Unconditional)
•Network Call Forwarding (Busy)
•Network Call Forwarding (No Answer)
•Three-Way Calling
•Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone By Way of a Backup Proxy
•Cisco SIP IP Phone to Cisco SIP IP Phone
Simple Call Hold
Figure B-4 illustrates a successful call between Cisco SIP IP phones in which one of the participants places the other on hold and then returns to the call. In this call flow scenario, the two end users are User A and User B. User A and User B are both using Cisco SIP IP phones, which are connected via 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 places User A on hold.
4. User B takes User A off hold.
5. The call continues.
Figure B-4 Simple Call Hold
Step
|
Action
|
Description
|
1.
|
INVITE—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE 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:555-0002@companyb.com; user=phone." The "user=phone" parameter distinquishes that the Request-URI address is a telephone number rather than a username.
•Cisco SIP IP 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—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A.
|
3.
|
200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A 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.
|
4.
|
ACK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone B.
The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone B. If the message body of the ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request.
|
A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B.
|
5.
|
INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with new Session Description Protocol (SDP) session parameters (IP address), which are used to place the call on hold.
The c= SDP field of the SIP INVITE contains an 0.0.0.0. This places the call in hold.
|
6.
|
200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
|
7.
|
ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
|
The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down.
|
8.
|
INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters (IP address), which are used to reestablish the call.
SDP: c=IN IP4 181.23.250.2
To reestablish the call between phone A and phone B, the IP address of phone B is inserted into the c= SDP field.
|
9.
|
200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
|
10.
|
ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
|
A two-way RTP channel is reestablished between IP phone A and IP phone B.
|
Call Hold with Consultation
Figure B-5 illustrates a successful call between Cisco SIP IP phones in which one of the participants places the other on hold, calls a third party (consultation), 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 SIP IP phones, which are connected via 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 places User A on hold.
4. User B calls User C.
5. User B disconnects from User C.
6. User B takes User A off hold.
7. The original call continues.
Figure B-5 Call Hold with Consultation
Step
|
Action
|
Description
|
1.
|
INVITE—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE 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:555-0002@companyb.com; user=phone." The "user=phone" parameter distinquishes that the Request-URI address is a telephone number rather than a username.
•Cisco SIP IP 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—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A.
|
3.
|
200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A 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.
|
4.
|
ACK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone B.
The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone B. If the message body of the ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request.
|
A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B.
|
5.
|
INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with new SDP session parameters (IP address), which are used to place the call on hold.
The c= SDP field of the SIP INVITE contains 0.0.0.0. This places the call in hold.
|
6.
|
200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
|
7.
|
ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
|
The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down.
|
8.
|
INVITE—Cisco SIP IP phone B to Cisco SIP IP phone C
|
Cisco SIP IP phone B sends a SIP INVITE request to Cisco SIP IP phone C. The INVITE request is an invitation to User C to participate in a call session.
|
9.
|
180 Ringing—Cisco SIP IP phone C to Cisco SIP IP phone B
|
Cisco SIP IP phone C sends a SIP 180 Ringing response to Cisco SIP IP phone B.
|
10.
|
200 OK—Cisco SIP IP phone C to Cisco SIP IP phone B
|
Cisco SIP IP phone C sends a SIP 200 OK response to 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 C
|
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP 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 Cisco SIP IP phone C
|
The call continues and then User B hangs up. Cisco SIP IP 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 to Cisco SIP IP phone B
|
Cisco SIP IP phone C sends a SIP 200 OK message to 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.
|
14.
|
INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters (IP address), which are used to reestablish the call.
SDP: c=IN IP4 181.23.250.2
To reestablish the call between phone A and phone B, the IP address of phone B is inserted into the c= SDP field.
|
15.
|
200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
|
16.
|
ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
|
Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
|
A two-way RTP channel is reestablished between Cisco SIP IP phone A and Cisco SIP IP phone B.
|
Call Waiting
Figure B-6 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call, 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 SIP IP phones, which are connected via 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.
Figure B-6 Call Waiting
Step
|
Action
|
Description
|
1.
|
INVITE—Cisco SIP IP phone A to Cisco SIP IP phone B
|
Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE 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:555-0002@companyb.com; user=phone." The "user=phone" parameter distinquishes that the Request-URI address is a telephone number rather than a username.
•Cisco SIP IP 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.
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2.
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180 Ringing—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A.
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3.
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200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A 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.
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4.
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ACK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone B.
The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone B. If the message body of the ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request.
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A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B.
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5.
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INVITE—Cisco SIP IP phone C to Cisco SIP IP phone B
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Cisco SIP IP phone C sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
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6.
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180 Ringing—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone C.
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7.
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INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with new SDP session parameters (IP address), which are used to place the call on hold.
The c= SDP field of the SIP INVITE contains 0.0.0.0. This places the call in hold.
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8.
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200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
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9.
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ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
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The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down.
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10.
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200 OK—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone C. The 200 OK response notifies Cisco SIP IP phone C that the connection has been made.
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11.
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ACK—Cisco SIP IP phone C to Cisco SIP IP phone B
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Cisco SIP IP phone C sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone C has received the 200 OK response from Cisco SIP IP phone B.
The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone B. If the message body of the ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request.
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A two-way RTP channel is established between Cisco SIP IP phone B and Cisco SIP IP phone C.
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12.
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INVITE—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone C with new SDP session parameters (IP address), which are used to place the call on hold.
To establish the call between phone B and phone C, the IP address of phone B is inserted into the c= SDP field.
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13.
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200 OK—Cisco SIP IP phone C to Cisco SIP IP phone B
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Cisco SIP IP phone C sends a SIP 200 OK response to Cisco SIP IP phone B.
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14.
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ACK—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone C. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone C.
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The RTP channel between Cisco SIP IP phone B and Cisco SIP IP phone C is torn down.
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15.
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INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with the same call ID as the previous INVITE (sent to Cisco SIP IP phone A) and new SDP session parameters (IP address), which are used to reestablish the call.
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16.
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200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
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17.
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ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
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A two-way RTP channel is reestablished between Cisco SIP IP phone A and Cisco SIP IP phone B.
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18.
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BYE—Cisco SIP IP phone B to Cisco SIP IP phone A
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The call continues and then User B hangs up. Cisco SIP IP phone B sends a SIP BYE request to Cisco SIP IP phone A. The BYE request indicates that User B wants to release the call.
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19.
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200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 200 OK message to 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.
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The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down.
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20.
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INVITE—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone C with the same call ID as the previous INVITE (sent to Cisco SIP IP phone C) and new SDP session parameters (IP address), which are used to reestablish the call.
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21.
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200 OK—Cisco SIP IP phone C to Cisco SIP IP phone B
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Cisco SIP IP phone C sends a SIP 200 OK response to Cisco SIP IP phone B.
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22.
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ACK—Cisco SIP IP phone B to Cisco SIP IP phone C
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Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone C. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
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A two-way RTP channel is reestablished between Cisco SIP IP phone B and Cisco SIP IP phone C.
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Call Transfer Without Consultation
Figure B-7 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call and then one of the participants transfers the call to a third party without first contacting the third party. This is called a blind or unattended transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Cisco SIP IP phones, which are connected via 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 transfers the call to User C.
Figure B-7 Call Transfer without Consultation
Step
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Action
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Description
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1.
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INVITE—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE 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:555-0002@companyb.com; user=phone." The "user=phone" parameter distinquishes that the Request-URI address is a telephone number rather than a username.
•Cisco SIP IP 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.
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2.
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100 Trying—Cisco SIP IP phone B to Cisco SIP IP phone A
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The Cisco SIP IP phone B sends a SIP 100 Trying response to Cisco SIP IP phone A. The 100 Trying response indicates that the INVITE request has been received by Cisco SIP IP phone B.
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3.
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180 Ringing—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A.
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4.
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200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A 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.
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5.
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ACK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone B. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone B.
The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone B. If the message body of the ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request.
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A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B.
User B then selects the option to blind transfer the call to User C.
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6.
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INVITE—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a mid-call INVITE to Cisco SIP IP phone A with new SDP session parameters (IP address), which are used to place the call on hold.
The c= SDP field of the SIP INVITE contains an 0.0.0.0. This places the call in hold.
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7.
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200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B.
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8.
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ACK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A.
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User B dials User C.
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9.
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REFER—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a REFER message to Cisco SIP IP phone A. The REFER message contains the following information:
•Refer-To: C
•Referred-By: B
The REFER message indicates that Cisco SIP IP phone A should send an INVITE request to Cisco SIP IP phone C.
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10.
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202 ACCEPTED—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 202 ACCEPTED message to Cisco SIP IP phone B. The 202 ACCEPTED confirms that the REFER message has been received.
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11.
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BYE—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a BYE message to Cisco SIP IP phone A. This message indicates that Cisco SIP IP phone B will be disconnecting from the call.
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12.
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200 OK—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B. The 200 OK response notifies Cisco SIP IP phone B that the BYE message was received.
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13.
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INVITE—Cisco SIP IP phone A to Cisco SIP IP phone C
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Because of the REFER message from Cisco SIP IP phone B, Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone C. The INVITE request is an invitation to User C to participate in a call session. The INVITE request contains the following information:
•Referred-By: B
This message indicates that the INVITE was referred by Cisco SIP IP phone B.
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14.
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100 Trying—Cisco SIP IP phone C to Cisco SIP IP phone A
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The Cisco SIP IP phone C sends a SIP 100 Trying response to Cisco SIP IP phone A. The 100 Trying response indicates that the INVITE request has been received by Cisco SIP IP phone C.
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15.
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180 Ringing—Cisco SIP IP phone C to Cisco SIP IP phone A
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Cisco SIP IP phone C sends a SIP 180 Ringing response to Cisco SIP IP phone A.
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16.
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200 OK—Cisco SIP IP phone C to Cisco SIP IP phone A
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Cisco SIP IP phone C sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
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17.
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ACK—Cisco SIP IP phone A to Cisco SIP IP phone C
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Cisco SIP IP phone A sends a SIP ACK to Cisco SIP IP phone C. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone C.
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18.
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NOTIFY—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a NOTIFY message to Cisco SIP IP phone B. The NOTIFY message notifies Cisco SIP IP phone C of the REFER event.
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19.
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200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the NOTIFY message was received.
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A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone C.
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Call Transfer Without Consultation Using Failover
Figure B-7 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call and then one of the participants transfers the call to a third party without first contacting the third party. This is called a blind or unattended transfer. In this call flow scenario, the end users are User A, User B, and User C. They are all using Cisco SIP IP phones, which are connected via 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 transfers the call to User C.
Figure B-8 Call Transfer Without Consultation Using Failover
Step
|
Action
|
Description
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1.
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INVITE—Cisco SIP IP phone A to Cisco SIP IP phone B
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Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone B. The INVITE 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:555-0002@companyb.com; user=phone." The "user=phone" parameter distinquishes that the Request-URI address is a telephone number rather than a username.
•Cisco SIP IP 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.
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100 Trying—Cisco SIP IP phone B to Cisco SIP IP phone A
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The Cisco SIP IP phone B sends a SIP 100 Trying response to Cisco SIP IP phone A. The 100 Trying response indicates that the INVITE request has been received by Cisco SIP IP phone B.
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3.
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180 Ringing—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A.
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4.
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200 OK—Cisco SIP IP phone B to Cisco SIP IP phone A
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Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A 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.
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