C H A P T E R 3

Call Flow Scenarios for Failed Calls

This chapter describes call flows for the following scenarios, which illustrate failed calls:
• SIP Gateway-to-SIP Gateway Calls
– The Called User is Busy, page 3-2
– The Called User Does Not Answer, page 3-3
– A Client, Server, or Global Error Has Occurred, page 3-5
• SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server
– The Called User is Busy, page 3-7
– The Called User Does Not Answer, page 3-9
– A Client, Server, or Global Error Has Occurred, page 3-12
• SIP Gateway-to-SIP Gateway Calls via a Proxy Server
– The Called User is Busy, page 3-14
– A Client or Server Error Has Occurred, page 3-16
– A Global Error Has Occurred, page 3-18
• SIP Gateway-to-SIP IP Phone
– The Called User is Busy, page 3-21
– The Called User Does Not Answer, page 3-22
– A Client, Server, or Global Error Has Occurred, page 3-24

SIP Gateway-to-SIP Gateway Calls

This section describes the call flows for failed SIP gateway-to-SIP gateway calls. In the following call
flows, the network configuration is the same as the network configuration outlined in the “SIP
Gateway-to-SIP Gateway Calls” section on page 3-1. However, instead of successfully establishing a
call session, one of the following situations occurs:
• The Called User is Busy, page 3-2
• The Called User Does Not Answer, page 3-3
• A Client, Server, or Global Error Has Occurred, page 3-5

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SIP Gateway-to-SIP Gateway Calls

The Called User is Busy

Figure 1 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the
phone and is unable or unwilling to accept another call.

Figure 1 SIP Gateway-to-SIP Gateway Call—Called User is Busy

User A PBX A GW1 IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. Call Proceeding 4. Setup
5. 100 Trying
6. Call Proceeding

8. 486 Busy Here 7. Disconnect (Busy)

9. Disconnect (Busy)
10. Release
11. Release
12. ACK
13. Release Complete
14. Release Complete

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Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to SIP gateway 2. The INVITE request
Gateway 2 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP Gateway SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
1 to PBX A Call Setup request.
4 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a
PBX B Call Setup with User B via PBX B.

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SIP Gateway-to-SIP Gateway Calls

Step Action Description

5 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
SIP Gateway 1 gateway 1. The 100 Trying message indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.
6 Call Proceeding—PBX B to SIP PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the
Gateway 2 Call Setup request.
7 Disconnect (Busy)—PBX B to PBX B sends a Disconnect message to SIP gateway 2. In the Disconnect
SIP Gateway 2 message, the cause code indicates that User B is busy. The Disconnect message
starts the call session termination process.
8 486 Busy Here—SIP Gateway 2 SIP gateway 2 maps the Release message cause code (Busy) to the 486 Busy
to SIP Gateway 1 Here response and sends the response to SIP gateway 1. The 486 Busy Here
response is a client error response that indicates that User B’s phone was
successfully contacted but User B was not willing or was unable to take another
call.
9 Disconnect (Busy)—SIP SIP gateway 1 sends a Release message to PBX A. User A hears a busy tone.
Gateway 1 to PBX A
10 Release—SIP Gateway 2 to SIP gateway 2 sends a Release message to PBX B.
PBX B
11 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
12 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 486
Gateway 2 Busy Here response has been received.
13 Release Complete—PBX B to PBX B sends a Release Complete message to SIP gateway 2.
SIP Gateway 2
14 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

The Called User Does Not Answer

Figure 2 illustrates an unsuccessful call in which User A initiates a call to User B but User B does not
answer.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls

Figure 2 SIP Gateway-to-SIP Gateway Call—Called User Does Not Answer

User A PBX A GW1 IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. Call Proceeding 4. Setup
5. 100 Trying
6. Call Proceeding

7. Alerting
8. 180 Ringing
9. Alerting
10. Cancel
11. Disconnect
12. Disconnect
13. Release
14. Release
15. 200 OK
16. Release Complete

28952
17. Release Complete

Step Action Description

1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to SIP gateway 2. The INVITE request
Gateway 2 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.
4 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a
PBX B Call Setup with User B via PBX B.
5 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
SIP Gateway 1 gateway 1. The 100 Trying response indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.

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SIP Gateway-to-SIP Gateway Calls

Step Action Description

6 Call Proceeding—PBX B to SIP PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the
Gateway 2 Call Setup request.
7 Alerting—PBX B to SIP PBX B sends an Alert message to SIP gateway 2. The message indicates that the
Gateway 2 PBX is attempting to alert the user of the call (that is to say, the phone is ringing).
8 180 Ringing—SIP Gateway 2 to SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing
SIP Gateway 1 response indicates that SIP gateway 2 has located, and is trying to alert, User B.
9 Alerting—SIP Gateway 1 to SIP gateway 1 sends an Alert message to PBX A. The Alert message indicates
PBX A that SIP gateway 1 has received a 180 Ringing response from SIP gateway 2.
User A hears the ringback tone that indicates that User B is being alerted.
10 Cancel (Ring Timeout)—SIP Because SIP gateway 2 did not return an appropriate response within the time
Gateway 1 to SIP Gateway 2 specified by the Expires header in the INVITE request, SIP gateway 1 sends a
SIP CANCEL request to SIP gateway 2. A CANCEL request cancels a pending
request with the same Call-ID, To, From, and CSeq header field values.
11 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
12 Disconnect—SIP Gateway 2 to SIP gateway 2 sends a Disconnect message to PBX B.
PBX B
13 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
14 Release—PBX B to SIP PBX B sends a Release message to SIP gateway 2.
Gateway 2
15 200 OK—SIP Gateway 2 to SIP SIP gateway 2 sends a 200 OK response to SIP gateway 2. The 200 OK response
Gateway 1 confirms that the Cancel request has been received.
16 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A.
Gateway 1 to PBX A
17 Release Complete—SIP SIP gateway 2 sends a Release Complete message to PBX B and the call session
Gateway 2 to PBX B attempt is terminated.

A Client, Server, or Global Error Has Occurred

Figure 3 illustrates an unsuccessful call in which User A initiates a call to User B but there are no more
channels available on the gateway. Therefore, SIP gateway 2 refuses the connection.

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SIP Gateway-to-SIP Gateway Calls

Figure 3 SIP Gateway-to-SIP Gateway Call—Client, Server, or Global Error

User A PBX A GW1 IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. Call Proceeding
4. 100 Trying

5. 4xx/5xx/6xx Failure-503
Service Unavailable
6. Disconnect
7. Release
8. ACK

28953
9. Release Complete

Step Action Description

1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to SIP gateway 2. The INVITE request
Gateway 2 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.
4 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
SIP Gateway 1 gateway 1. The 100 Trying message indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.

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SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Step Action Description

5 Class 4xx/5xx/6xx Failure—SIP SIP gateway 2 determines that it does not have any more channels available,
Gateway 2 to SIP Gateway 1 refuses the connection, and sends a SIP 503 Service Unavailable response to SIP
gateway 1.
The 503 Service Unavailable response is a class 4xx, 5xx, or class 6xx failure
response. Depending on which class the failure response is, the call actions
differ.
If SIP gateway 2 sends a class 4xx failure response (a definite failure response
that is a client error), the request will not be retried without modification.
If SIP gateway 2 sends a class 5xx failure response (an indefinite failure that is
a server error), the request is not terminated but rather other possible locations
are tried.
If SIP gateway 2 sends a class 6xx failure response (a global error), the search
for User B is terminated because the 6xx response indicates that a server has
definite information about User B, but not for the particular instance indicated
in the Request-URI field. Therefore, all further searches for this user will fail.
The call failure on SIP gateway 2 might occur before a proceeding indication
from PBX B. In that case a SIP failure response is sent before the 100 Trying
response.
6 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
7 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
8 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 503
Gateway 2 Service Unavailable response has been received.
9 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

This section describes the call flows for gateway-to-gateway calls via a redirect server that have failed.
In the following call flows, the network configuration is the same as the network configuration outlined
in the “SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server” section on page 3-7. However,
instead of successfully establishing a call session, one of the following situations occurs:
• The Called User is Busy, page 3-7
• The Called User Does Not Answer, page 3-9
• A Client, Server, or Global Error Has Occurred, page 3-12

The Called User is Busy

Figure 4 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the
phone and is unable or unwilling to accept another call.

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SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Figure 4 SIP Gateway-to-SIP Gateway Call via a SIP Redirect Server—Called User is Busy

User A PBX A GW1 RS IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. 302 Moved
Temporarily

4. ACK
6. Call
Proceeding 5. INVITE
7. Setup
8. 100 Trying
9. Call
Proceeding

12. 10. Disconnect

Disconnect (Busy)
11. 486 Busy Here
(Busy)

13. Release
14. Release
16. Release 15. ACK
Complete 17. Release
Complete

28939
Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes
Gateway 1 the standard transactions that take place as User A attempts to call User B.
2 INVITE—SIP Gateway 1 to SIP gateway 1 sends an INVITE request to the SIP redirect server. The INVITE
SIP Redirect Server 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is specified.
3 302 Moved Temporarily— SIP redirect server sends a 302 Moved Temporarily message to SIP gateway 1. The
SIP Redirect Server to SIP message indicates that User B is not available and includes instructions to locate
Gateway 1 User B.
4 ACK—SIP Gateway 1 to SIP SIP gateway 1 acknowledges the 302 Moved Temporarily response with an ACK.
Redirect Server

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Step Action Description

5 INVITE—SIP Gateway 1 to SIP gateway 1 sends a new INVITE request to User B. The new INVITE request
SIP Gateway 2 includes the first contact listed in the 300 Multiple Choice response as the new
address for User B, a higher transaction number in the CSeq field, and the same
Call-ID as the first INVITE request.
6 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call
Gateway 1 to PBX A Setup request.
7 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call
PBX B Setup with User B via PBX B.
8 100 Trying—SIP Gateway 2 SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
to SIP Gateway 1 gateway 1. The 100 Trying response indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.
9 Call Proceeding—PBX B to PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call
SIP Gateway 2 Setup request.
10 Disconnect (Busy)—PBX B PBX B sends a Disconnect message to SIP gateway 2. In the Disconnect message,
to SIP Gateway 2 the cause code indicates that User B is busy. The Disconnect message starts the call
session termination process.
11 486 Busy Here—SIP SIP gateway 2 maps the Release message cause code (Busy) to the 486 Busy
Gateway 2 to SIP Gateway 1 response and sends the response to SIP gateway 1. The 486 Busy Here response is a
client error response that indicates that User B’s phone was successfully contacted
but User B was not willing or was unable to take another call.
12 Disconnect (Busy) —SIP SIP gateway 1 sends a Disconnect message to PBX A. User A hears a busy tone.
Gateway 1 to PBX A
13 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
14 Release—SIP Gateway 2 to SIP gateway 1 sends a Release message to PBX B.
PBX B
15 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 486 Busy
Gateway 2 Here response has been received.
16 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.
17 Release Complete—PBX B PBX B sends a Release Complete message to SIP gateway 2.
to SIP Gateway 2

The Called User Does Not Answer

Figure 5 illustrates an unsuccessful call in which User A initiates a call to User B but User B does not
answer.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Figure 5 SIP Gateway-to-SIP Gateway Call via a SIP Redirect Server—Called User is Does Not
Answer

User A PBX A GW1 RS IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. 302 Moved
Temporarily

4. ACK
6. Call
Proceeding 5. INVITE
7. Setup
8. 100 Trying
9. Call
Proceeding
10. Alerting
11. 180 Ringing
12. Alerting
14. 13. CANCEL
Disconnect
15. Release 16. Disconnect
17. 200 OK
18. Release 19. Release
Complete
20. Release

28940
Complete

Step Action Description

1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to the SIP redirect server. The INVITE
Redirect Server 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.

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SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Step Action Description

3 302 Moved Temporarily— SIP SIP redirect server sends a 302 Moved Temporarily message to SIP gateway 1.
Redirect Server to SIP The message indicates that User B is not available and includes instructions to
Gateway 1 locate User B.
4 ACK—SIP Gateway 1 to SIP SIP gateway 1 acknowledges the 302 Moved Temporarily response with an
Redirect Server ACK.
5 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends a new INVITE request to User B. The new INVITE request
Gateway 2 includes a new address for User B, a higher transaction number in the CSeq field,
but the same Call-ID as the first INVITE request.
6 Call Proceeding—SIP Gateway SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
1 to PBX A Call Setup request.
7 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a
PBX B Call Setup with User B via PBX B.
8 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
SIP Gateway 1 gateway 1. The 100 Trying message indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.
9 Call Proceeding—PBX B to SIP PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the
Gateway 2 Call Setup request.
10 Alerting—PBX B to SIP PBX B sends an Alert message to SIP gateway 2. User B’s phone begins to ring.
Gateway 2
11 180 Ringing—SIP Gateway 2 to SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing
SIP Gateway 1 response indicates that SIP gateway 2 has located, and is trying to alert, User B.
12 Alerting—SIP Gateway 1 to SIP gateway 1 sends an Alert message to PBX A.
PBX A
13 CANCEL (Ring Timeout)—SIP Because SIP gateway 2 did not return an appropriate response within the time
Gateway 1 to SIP Gateway 2 specified by the Expires header in the INVITE request, SIP gateway 1 sends a
SIP CANCEL request to SIP gateway 2. A CANCEL request cancels a pending
request with the same Call-ID, To, From, and CSeq header field values.
14 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
15 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
16 Disconnect—SIP Gateway 2 to SIP gateway 2 sends a Disconnect message to PBX B.
PBX B
17 200 OK—SIP Gateway 1 to SIP SIP gateway 1 sends a 200 OK response to SIP gateway 2. The 200 OK response
Gateway 2 confirms that the CANCEL request has been received.
18 Release Complete—PBX A to PBX A sends a Release Complete message to SIP gateway 1 and the call session
SIP Gateway 1 attempt is terminated.
19 Release—PBX B to SIP PBX B sends a Release message to SIP gateway 2.
Gateway 2

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SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Step Action Description

20 Release Complete—SIP SIP gateway 2 sends a Release Complete message to PBX B.
Gateway 2 to PBX B

A Client, Server, or Global Error Has Occurred

Figure 6 illustrates an unsuccessful call in which User A initiates a call to User B but SIP gateway 2
determines that User B does not exist at the domain specified in the INVITE request sent by SIP
gateway 1. SIP gateway 2 refuses the connection.

Figure 6 SIP Gateway-to-SIP Gateway Call via a SIP Redirect Server—Client, Server, or Global

User A PBX A GW1 RS IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. 300 Multiple
Choice

4. ACK
6. Call
Proceeding 5. INVITE

7. 100 Trying

8. 4xx/5xx/6xx Failure-404 Not Found

9. Disconnect

11. Release
12. Release 11. ACK
Complete

28941

Step Action Description

1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.

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SIP Gateway-to-SIP Gateway Calls via a SIP Redirect Server

Step Action Description

2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to the SIP redirect server. The INVITE
Redirect Server 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.
• PBX A is identified as the 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 is specified.
• The port on which SIP gateway 1 is prepared to receive the RTP data is
specified.
3 300 Multiple Choice—SIP The SIP redirect server sends a 300 Multiple Choice response to SIP gateway 1.
Redirect Server to SIP The 300 Multiple Choice response indicates that the SIP redirect server accepted
Gateway 1 the INVITE request, contacted a location server with all or part of User B’s SIP
URL, and the location server provided a list of alternative locations where User
B might be located. The SIP redirect server returns these possible addresses to
User A in the 300 Multiple Choice response.
4 ACK—SIP Gateway 1 to SIP SIP gateway 1 acknowledges the 300 Multiple Choice response with an ACK.
Redirect Server
5 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends a new INVITE request to User B. The new INVITE request
Gateway 2 includes a new address for User B, a higher transaction number in the CSeq field,
but the same Call-ID as the first INVITE request.
6 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to SIP Gateway 2 Call Setup request.
7 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP
SIP Gateway 1 gateway 1. The 100 Trying message indicates that the INVITE request has been
received by SIP gateway 2 but that User B has not yet been located and that some
unspecified action, such as a database consultation, is taking place.

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SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Step Action Description

8 Class 4xx/5xx/6xx Failure—SIP SIP gateway 2 determines that User B does not exist at the domain specified in
Gateway 2 to SIP Gateway 1 the INVITE request sent by SIP gateway 1. SIP gateway 2 refuses the connection
and sends a 404 Not Found response to SIP gateway 1.
The 404 Not Found response is a class 4xx failure response. Depending on which
class the failure response is, the call actions differ.
If SIP gateway 2 sends a class 4xx failure response (a definite failure response
that is a client error), the request will not be retried without modification.
If SIP gateway 2 sends a class 5xx failure response (an indefinite failure that is
a server error), the request is not terminated but rather other possible locations
are tried.
If SIP gateway 2 sends a class 6xx failure response (a global error), the search
for User B is terminated because the 6xx response indicates that a server has
definite information about User B, but not for the particular instance indicated
in the Request-URI field. Therefore, all further searches for this user will fail.
9 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
10 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
11 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 404
Gateway 2 Not Found failure response has been received.
12 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

SIP Gateway-to-SIP Gateway Calls via a Proxy Server

This section describes the call flows for gateway-to-gateway calls via a proxy server that have failed. In
the following call flows, the network configuration is the same as the network configuration outlined in
the “SIP Gateway-to-SIP Gateway Calls via a Proxy Server” section on page 3-14. However, instead of
successfully establishing a call session, one of the following situations occurs:
• The Called User is Busy, page 3-14
• A Client or Server Error Has Occurred, page 3-16
• A Global Error Has Occurred, page 3-18

The Called User is Busy

Figure 7 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the
phone and is unwilling or unable to accept another call.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Figure 7 SIP Gateway-to-SIP Gateway Call via a SIP Proxy Server—Called User is Busy
Proxy
User A PBX A GW1 Server IP Network GW2 PBX B User B

1. Setup
2. INVITE
4. Call 3. INVITE
Proceeding
5. Setup
6. 100 Trying
7. 100 Trying 8. Release
Complete
(Busy)
9. 486 Busy Here
11. 10. 486 Busy Here
Disconnect
(Busy)

12. Release
13. ACK
15. Release 14. ACK
Complete

28943
Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to the SIP proxy server. The INVITE
Proxy Server 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 INVITE—SIP Proxy Server to The SIP proxy server checks whether it’s own address is contained in the Via
SIP Gateway 2 field (to prevent loops), directly copies the To, From, Call-ID, and Contact fields
from the request it received from SIP gateway 1, changes the Request-URI to
indicate the server to which it intends to send the INVITE request, and then
sends a new INVITE request to SIP gateway 2.
4 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Step Action Description

5 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from the SIP proxy server and
PBX B initiates a Call Setup with User B via PBX B.
6 100 Trying—SIP Proxy Server The SIP proxy server sends a 100 Trying response to SIP gateway 1.
to SIP Gateway 1
7 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the SIP proxy server.
SIP Proxy Server
8 Release Complete PBX B sends a Release Complete message to SIP gateway 2. In the Release
(Busy)—PBX B to SIP Complete message, the cause code indicates that User B is busy. The Release
Gateway 2 Complete message starts the call session termination process.
9 486 Busy Here—SIP Gateway 2 SIP gateway 2 maps the Release message cause code (Busy) to the 486 Busy
to SIP Proxy Server response and sends the response to the SIP proxy server. The 486 Busy Here
response is a client error response that indicates that User B’s phone was
successfully contacted but User B was not willing or was unable to take another
call.
10 486 Busy Here—SIP Proxy The SIP proxy server forwards the 486 Busy response to SIP gateway 1.
Server to SIP Gateway 1
11 Disconnect (Busy)—SIP SIP gateway 1 sends a Disconnect message to PBX A.
Gateway 1 to PBX A
12 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
13 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an SIP ACK to the SIP proxy server.
Proxy Server
14 ACK—SIP Proxy Server to SIP The SIP proxy server forwards the SIP ACK to SIP gateway 2. The ACK
Gateway 2 confirms that the 486 Busy Here response has been received.
15 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

A Client or Server Error Has Occurred

Figure 8 illustrates an unsuccessful call in which User A initiates a call to User B but there are no more
channels available on SIP gateway 2. Therefore, SIP gateway 2 refuses the connection.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Figure 8 SIP Gateway-to-SIP Gateway Call via a SIP Proxy Server—Client or Server Error
Proxy
User A PBX A GW1 Server IP Network GW2 PBX B User B

1. Setup
2. INVITE
4. Call 3. INVITE
Proceeding
5. 100 Trying
6. 100 Trying
8. 4xx/5xx/
Failure-503 7. 4xx/5xx/ Failure-503
Service Service Unavailable
Unavailable
9. Disconnect

10. Release
11. ACK
13. Release 12. ACK
Complete

28945
Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP Gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to the SIP proxy server. The INVITE
Proxy Server 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.
• PBX A is identified as the 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 INVITE—SIP Proxy Server to The SIP proxy server checks whether it’s own address is contained in the Via
SIP Gateway 2 field (to prevent loops), directly copies the To, From, Call-ID, and Contact fields
from the request it received from SIP gateway 1, changes the Request-URI to
indicate the server to which it intends to send the INVITE request, and then
sends a new INVITE request to SIP gateway 2.
4 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Step Action Description

5 100 Trying—SIP Proxy Server The SIP proxy server sends a 100 Trying response to SIP gateway 1.
to SIP Gateway 1
6 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the SIP proxy server.
SIP Proxy Server
7 Class 4xx/5xx/6xx Failure—SIP SIP gateway 2 determines that it does not have any more channels available,
Gateway 2 to SIP Proxy Server refuses the connection, and sends a SIP 503 Service Unavailable response to the
SIP proxy server.
8 Class 4xx/5xx/6xx Failure—SIP The SIP proxy server forwards the SIP 503 Service Unavailable response to SIP
Proxy Server to SIP Gateway 1 gateway 1.
9 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
10 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
11 ACK—SIP Gateway 1 to SIP SIP Gateway 1 sends an ACK to the SIP proxy server.
Proxy Server
12 ACK—SIP Proxy Server to SIP The SIP proxy server forwards the SIP ACK to SIP Gateway 2. The ACK
Gateway 2 confirms that the 503 Service Unavailable response has been received.
13 Release Complete—SIP SIP Gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

A Global Error Has Occurred

Figure 9 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class
6xx response.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP Gateway Calls via a Proxy Server

Figure 9 SIP Gateway-to-SIP Gateway Call via a SIP Proxy Server—Global Error Response

Proxy
User A PBX A GW1 Server IP Network GW2 PBX B User B

1. Setup
2. INVITE
3. Call
Proceeding 4. INVITE
5. Setup
6. 100 Trying
7. 100 Trying
8. Release
Complete
9. 6xx Failure
11. 10. 6xx Failure
Disconnect

12. Release
13. ACK
15. Release 14. ACK
Complete

28946
Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 sends an INVITE request to the SIP proxy server. The INVITE
Proxy Server 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.
• 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 SIP gateway 1 is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.
4 INVITE—SIP Proxy Server to The SIP proxy server checks whether it’s own address is contained in the Via
SIP Gateway 2 field (to prevent loops), directly copies the To, From, Call-ID, and Contact fields
from the request it received from SIP gateway 1, changes the Request-URI to
indicate the server to which it intends to send the INVITE request, and then
sends a new INVITE request to SIP gateway 2.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

Step Action Description

5 Setup—SIP Gateway 2 to SIP gateway 2 receives the INVITE request from the SIP proxy server and
PBX B initiates a Call Setup with User B via PBX B.
6 100 Trying—SIP Gateway 2 to SIP gateway 2 sends a 100 Trying response to the SIP proxy server. The SIP
SIP Proxy Server proxy server might or might not forward the 100 Trying response to SIP
gateway 1.
7 100 Trying—SIP Proxy Server The SIP proxy server forwards the 100 Trying response to SIP gateway 1.
to SIP Gateway 1
8 Release Complete—PBX B to PBX B sends a Release Complete message to SIP gateway 2. The Release
SIP Gateway 2 Complete message starts the call session termination process.
9 6xx Failure—SIP Gateway 2 to SIP gateway 2 sends a class 6xx failure response (a global error) to the SIP proxy
SIP Proxy Server server. A class 6xx failure response indicates that a server has definite
information about User B, but not for the particular instance indicated in the
Request-URI field. All further searches for this user will fail, therefore the
search is terminated.
The SIP proxy server must forward all class 6xx failure responses to the client.
10 6xx Failure—SIP Proxy Server The SIP proxy server forwards the 6xx failure to SIP gateway 1.
to SIP Gateway 1
11 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
12 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
13 ACK—SIP Gateway 1 to SIP SIP gateway 1 sends an ACK to the SIP proxy server.
Proxy Server
14 ACK—SIP Proxy Server to SIP The SIP proxy server sends an ACK to SIP gateway 2. The ACK confirms that
Gateway 2 the 6xx failure response has been received.
15 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

SIP Gateway-to-SIP IP Phone

This section describes the call flows for failed SIP gateway-to-SIP IP phone calls. In the following call
flows, the network configuration is the same as the network configuration outlined in the “SIP
Gateway-to-SIP IP Phone” section on page 3-20. However, instead of successfully establishing a call
session, one of the following situations occurs:
• The Called User is Busy, page 3-21
• The Called User Does Not Answer, page 3-22
• A Client, Server, or Global Error Has Occurred, page 3-24

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

The Called User is Busy

Figure 10 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the
phone and is unable or unwilling to take another call.

Figure 10 SIP Gateway-to-SIP IP Phone—Called User is Busy

SIP IP Phone
User A PBX A GW1 IP Network User B

IP

1. Setup
2. INVITE
3. Call Proceeding

4. 100 Trying

5. 486 Busy Here

6. Disconnect (Busy)
7. Release
8. ACK
9. Release Complete

41725
Step Action Description
1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer
IP Phone includes the IP address and the port number of the SIP enabled entity to contact.
SIP gateway 1 sends an INVITE request to the address it receives in the dial peer
which, in this scenario, is the SIP IP phone.
In the INVITE request:
• The IP address of the 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 SIP gateway is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

Step Action Description

4 100 Trying—SIP IP Phone to The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying
SIP Gateway 1 response indicates that the INVITE request has been received by the SIP IP
phone.
5 486 Busy Here—SIP IP Phone The SIP IP phone sends a 486 Busy Here response to SIP gateway 1. The 486
to SIP Gateway 1 Busy Here response is a client error response that indicates that User B was
successfully contacted but User B was not willing or was unable to take the call.
6 Disconnect (Busy)—SIP SIP gateway 1 sends a Disconnect message to PBX A.
Gateway 1 to PBX A
7 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
8 ACK—SIP Gateway 1 to SIP IP SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that User
Phone A has received the 486 Busy Here response. The call session attempt is now
being terminated.
9 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

The Called User Does Not Answer

Figure 11 illustrates the call flow in which User A initiates a call to User B but User B does not answer.

Figure 11 SIP Gateway-to-SIP IP Phone—Called User Does Not Answer

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

Step Action Description

1 Setup—PBX A to SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call User
B.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

Step Action Description

2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer
IP Phone includes the IP address and the port number of the SIP enabled entity to contact.
SIP gateway 1 sends an INVITE request to the address it receives in the dial peer
which, in this scenario, is the SIP IP phone.
In the INVITE request:
• The IP address of the 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 SIP gateway is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.
4 100 Trying—SIP IP Phone to The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying
SIP Gateway 1 response indicates that the INVITE request has been received by the SIP IP
phone.
5 180 Ringing—SIP IP Phone to The SIP IP phone sends a 180 Ringing response to SIP gateway 1. The
SIP Gateway 1 180 Ringing response indicates that the user is being alerted.
6 Alerting—SIP Gateway 1 to SIP gateway 1 sends an Alert message to PBX A. The Alert message indicates
PBX A that SIP gateway 1 has received a 180 Ringing response from the SIP IP phone.
User A hears the ringback tone that indicates that User B is being alerted.
7 CANCEL (Ring Timeout)—SIP Because SIP gateway 1 did not return an appropriate response within the time
Gateway 1 to SIP IP Phone specified by the Expires header in the INVITE request, SIP gateway 1 sends a
SIP CANCEL request to SIP gateway 2. A CANCEL request cancels a pending
request with the same Call-ID, To, From, and CSeq header field values.
8 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Disconnect message to PBX A.
PBX A
9 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.
10 200 OK—SIP IP Phone to SIP The SIP IP phone sends a 200 OK response to SIP gateway 1. The 200 OK
Gateway 1 response confirms that User A has received the Cancel request. The call session
attempt is now being terminated.
11 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A is terminated.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

A Client, Server, or Global Error Has Occurred

Figure 12 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class
4xx, 5xx, or 6xx response.

Figure 12 SIP Gateway-to-SIP IP Phone—Client, Server, or Global Error

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 SIP Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup
Gateway 1 includes the standard transactions that take place as User A attempts to call
User B.
2 INVITE—SIP Gateway 1 to SIP SIP gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer
IP Phone includes the IP address and the port number of the SIP enabled entity to contact.
SIP gateway 1 sends an INVITE request to the address it receives in the dial peer
which, in this scenario, is the SIP IP phone.
In the INVITE request:
• The IP address of the 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 SIP gateway is prepared to receive the RTP data is
specified.
3 Call Proceeding—SIP SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the
Gateway 1 to PBX A Call Setup request.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

Step Action Description

4 100 Trying—SIP IP Phone to The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying
SIP Gateway 1 response indicates that the INVITE request has been received by the SIP IP
phone.
5 4xx/5xx/6xx Failure—SIP IP The SIP IP phone sends a class 4xx, 5xx, or class 6xx failure response to SIP
Phone to SIP Gateway 1 gateway 1. Depending on which class the failure response is, the call actions
differ.
If the SIP IP phone sends a class 4xx failure response (a definite failure response
that is a client error), the request will not be retried without modification.
If the SIP IP phone sends a class 5xx failure response (an indefinite failure that
is a server error), the request is not terminated but rather other possible locations
are tried.
If the SIP IP phone sends a class 6xx failure response (a global error), the search
for User B is terminated because the 6xx response indicates that a server has
definite information about User B, but not for the particular instance indicated
in the Request-URI field. Therefore, all further searches for this user will fail.
6 Disconnect—SIP Gateway 1 to SIP gateway 1 sends a Release message to PBX A.
PBX A
7 Release—PBX A to SIP PBX A sends a Release message to SIP gateway 1.
Gateway 1
8 ACK—SIP Gateway 1 to SIP IP SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that User A
Phone has received the 4xx/5xx/6xx failure response. The call session attempt is now
being terminated.
9 Release Complete—SIP SIP gateway 1 sends a Release Complete message to PBX A and the call session
Gateway 1 to PBX A attempt is terminated.

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 Chapter 3 Call Flow Scenarios for Failed Calls
SIP Gateway-to-SIP IP Phone

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