Session Initiation Protocol Gateway Call Flows and Compliance Information

Step	Action	Description
1	Setup—PBX A to SIP Gateway 1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B.
2	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an INVITE request to SIP gateway 2. 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 email 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 indicates that the Request-URI address is a telephone number rather than a user name. 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 1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	Setup—SIP Gateway 2 to PBX B	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBX B.
5	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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.
6	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
7	Alerting—PBX B to SIP Gateway 2	PBX B locates User B and sends an Alert message to SIP gateway 2. User B's phone begins ringing.
8	180 Ringing—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that SIP gateway 2 has located, and is trying to alert, User B.
9	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A via PBX A. The Alert message indicates 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.
At this point, a one-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
10	Connect—PBX B to SIP Gateway 2	User B answers phone. PBX B sends a Connect message to SIP gateway 2. The Connect message notifies SIP gateway 2 that the connection has been made.
11	200 OK—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made. If User B supports the media capability advertised in the INVITE message sent by SIP gateway 1, it advertises the intersection of its own and User A's media capability in the 200 OK response. If User B does not support the media capability advertised by User A, it sends back a 400 Bad Request response with a 304 Warning header field.
12	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
13	Connect ACK—PBX A to SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
14	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a an ACK to SIP gateway 2. The ACK confirms that SIP gateway 1 has received the 200 OK response from SIP gateway 2.
15	Connect ACK—SIP Gateway 2 to PBX B	SIP gateway 2 acknowledges PBX B's Connect message. The call session is now active over a two-way voice path via Real Time Transport Protocol (RTP).
At this point, a two-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
16	Disconnect—PBX B to SIP Gateway 2	Once User B hangs up, PBX B sends a Disconnect message to SIP gateway 2. The Disconnect message starts the call session termination process.
17	BYE—SIP Gateway 2 to SIP Gateway 1	SIP Gateway 2 sends a BYE request to SIP gateway 1. The BYE request indicates that User B wants to release the call. Because it is User B that wants to terminate the call, the Request-URI field is now replaced with PBX A's SIP URL and the From field contains User B's SIP URL.
18	Release—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Release message to PBX B.
19	Disconnect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
20	Release—PBX A to SIP Gateway 1	PBX A sends a Disconnect message to SIP gateway 1.
21	200 OK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a 200 OK response to SIP gateway 2. The 200 OK response notifies SIP gateway 2 that SIP gateway 1 has received the BYE request.
22	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2.
23	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Redirect Server	SIP gateway 1 sends an INVITE request to the SIP redirect server. 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 email 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 user name. 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	300 Multiple Choice—SIP Redirect Server to SIP Gateway1	The SIP redirect server sends a 300 Multiple Choice response to SIP gateway 1. The 300 Multiple Choice response indicates that the SIP redirect server accepted 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 SIP gateway 1 in the 300 Multiple Choice response.
4	ACK—SIP Gateway 1 to SIP Redirect Server	SIP gateway 1 acknowledges the 300 Multiple Choice response with an ACK.
5	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a new INVITE request to SIP gateway 2. The new INVITE request 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	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBXB.
7	Call Proceeding—SIP Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
8	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP gateway1. 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.
9	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
10	Alerting—PBX B to SIP Gateway 2	PBX B locates User B and sends an Alert message to SIP gateway 2. User B's phone begins to ring.
11	180 Ringing—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that SIP gateway 2 has located, and is trying to alert, User B.
12	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to PBX A. User A hears ringback tone.
At this point, a one-way voice path is established between SIP gateway 1 and PBXA and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
13	Connect—PBX B to SIP Gateway 2	User B answers phone. PBX B sends a Connect message to SIP gateway 2. The Connect message notifies SIP gateway 2 that the connection has been made.
14	200 OK—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made. If User B supports the media capability advertised in the INVITE message sent by SIP gateway 1, it advertises the intersection of its own and User A's media capability in the 200 OK response. If User B does not support the media capability advertised by User A, it sends back a 400 Bad Request response with a 304 Warning header field.
15	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
16	Connect ACK—PBX A to SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
17	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 200 OK response has been received. The call is now in progress over a two-way voice path via RTP.
18	Connect ACK—SIP Gateway 2 to PBX B	SIP gateway 2 acknowledges PBX B's Connect message.
At this point, a two-way voice path is established between SIP gateway 1 and PBXA and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
19	Disconnect—PBX B to SIP Gateway2	Once User B hangs up, PBX B sends a Disconnect message to SIP gateway 2. The Disconnect message starts the call session termination process.
20	BYE—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a BYE request to SIP gateway 1. The BYE request indicates that User B wants to release the call. Because it is User B that wants to terminate the call, the Request-URI field is now replaced with PBX A's SIP URL and the From field contains User B's SIP URL.
21	Disconnect—SIP Gateway 1 to PBXA	SIP gateway 1 sends a Disconnect message to PBX A.
22	Release—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Release message to PBX B.
23	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
24	200 OK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a 200 OK response to SIP gateway 2. The 200 OK response notifies SIP gateway 2 that SIP gateway 1 has received the BYE request.
25	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2.
26	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to Proxy Server	SIP gateway 1 sends an INVITE request to the SIP proxy server. 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 email 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 user name. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	INVITE—SIP Proxy Server to SIP Gateway 2	The SIP proxy server checks whether it's own address is contained in the Via 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. In the INVITE request, the SIP proxy server also adds the Record-Route header to ensure that it is in the path of subsequent SIP requests for the same call leg. In the Record-Route field, the SIP proxy server adds its Request-URI.
5	100 Trying—SIP Proxy Server to SIP Gateway 1	The SIP proxy server sends a 100 Trying response to SIP gateway 1.
6	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with UserB via PBXB.
7	100 Trying—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 100 Trying response to the SIP proxy server. The SIP proxy server might or might not forward the 100 Trying response to SIP gateway 1.
8	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
9	Alerting—PBX B to SIP Gateway 2	PBX B locates User B and sends an Alert message to SIP gateway 2. User B's phone begin to ring.
10	180 Ringing—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 180 Ringing response to the SIP proxy server.
11	180 Ringing—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the 180 Ringing response to SIP gateway 1.
12	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A via PBX A. The Alert message indicates that SIP gateway 1 has received a 180 Ringing response. User A hears the ringback tone that indicates that User B is being alerted.
At this point, a one-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
13	Connect—PBX B to SIP Gateway 2	User B answers the phone. PBX B sends a Connect message to SIP gateway 2. The connect message notifies SIP gateway 2 that the connection has been made.
14	200 OK—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 200 OK response to the SIP proxy server. The 200 OK response notifies the SIP proxy server that the connection has been made. In the 200 OK response, the SIP gateway 2 adds the Record-Route header (received in the INVITE request) to ensure that it is in the path of subsequent SIP requests for the same call leg. If User B supports the media capability advertised in the INVITE message sent by the SIP proxy server, it advertises the intersection of its own and User A's media capability in the 200 OK response. If User B does not support the media capability advertised by User A, it sends back a 400 Bad Request response with a 304 Warning header field. The SIP proxy server must forward 200 OK responses.
15	200 OK—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the 200 OK response that it received from SIP gateway 2 to SIP gateway 1.
16	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
17	Connect ACK—PBX A to SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
18	ACK—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an ACK to the SIP proxy server. The ACK confirms that SIP gateway 1 has received the 200 OK response from the SIP proxy server.
19	ACK—SIP Proxy Server to SIP Gateway2	Depending on the values in the To, From, CSeq, and Call-ID field, the SIP proxy server might process the ACK locally or proxy it. If the fields in the ACK match those in previous requests processed by the SIP proxy server, the server proxies the ACK. If there is no match, the ACK is proxied as if it were an INVITE request. The SIP proxy server forwards SIP gateway 1's ACK response to SIP gateway 2.
20	Connect ACK—SIP Gateway 2 to PBX B	SIP gateway 2 acknowledges PBX B's Connect message. The call session is now active. The 2-way voice path is established directly between SIP gateway 1 and SIP gateway 2; not via the SIP proxy server.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
21	Disconnect—PBX B to SIP Gateway2	After the call is completed, PBX B sends a Disconnect message to SIP gateway2. The Disconnect message starts the call session termination process.
22	BYE—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a BYE request to the SIP proxy server. The BYE request indicates that User B wants to release the call. Because it is User B that wants to terminate the call, the Request-URI field is now replaced with PBX A's SIP URL and the From field contains UserB's SIP URL.
23	BYE—SIP Proxy Server to SIP Gateway1	The SIP proxy server forwards the BYE request to SIP gateway 1.
24	Disconnect—SIP Gateway 1 to PBXA	SIP gateway 1 sends a Disconnect message to PBX A.
25	Release—SIP Gateway 2 to PBX B	After the call is completed, SIP gateway 2 sends a Release message to PBX B.
26	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
27	200 OK—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends a 200 OK response to the SIP proxy server. The 200 OK response notifies SIP gateway 2 that SIP gateway 1 has received the BYE request.
28	200 OK—SIP Proxy Server to SIP Gateway 2	The SIP proxy server forwards the 200 OK response to SIP gateway 2.
29	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2.
30	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an INVITE request to the SIP proxy server. 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 email 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 user name. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	INVITE—SIP Proxy Server to SIP Gateway 2	The SIP proxy server checks whether it's own address is contained in the Via 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.
5	100 Trying—SIP Proxy Server to SIP Gateway 1	The SIP proxy server sends a 100 Trying response to SIP gateway 1.
6	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with UserB via PBXB.
7	100 Trying—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 100 Trying response to the SIP proxy server. The SIP proxy server might or might not forward the 100 Trying response to SIP gateway 1.
8	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
9	Alerting—PBX B to SIP Gateway 2	PBX B locates User B and sends an Alert message to SIP gateway 2. User B's phone begin to ring.
10	180 Ringing—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 180 Ringing response to the SIP proxy server.
11	180 Ringing—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the 180 Ringing response to SIP gateway 1.
12	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A via PBX A. The Alert message indicates that SIP gateway 1 has received a 180 Ringing response. User A hears the ringback tone that indicates that UserB is being alerted.
At this point, a one-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
13	Connect—PBX B to SIP Gateway 2	User B answers the phone. PBX B sends a Connect message to SIP gateway 2. The connect message notifies SIP gateway 2 that the connection has been made.
14	200 OK—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 200 OK response to the SIP proxy server. The 200 OK response notifies the SIP proxy server that the connection has been made. If User B supports the media capability advertised in the INVITE message sent by the SIP proxy server, it advertises the intersection of its own and User A's media capability in the 200 OK response. If User B does not support the media capability advertised by User A, it sends back a 400 Bad Request response with a 304 Warning header field. The SIP proxy server must forward 200 OK responses.
15	200 OK—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the 200 OK response that it received from SIP gateway 2 to SIP gateway 1.
16	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
17	Connect ACK—PBX A to SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
18	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that SIP gateway 1 has received the 200 OK response from the SIP proxy server.
19	Connect ACK—SIP Gateway 2 to PBX B	SIP gateway 2 acknowledges PBX B's Connect message. The call session is now active. The 2-way voice path is established directly between SIP gateway 1 and SIP gateway 2; not via the SIP proxy server.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.
20	Disconnect—PBX B to SIP Gateway2	After the call is completed, PBX B sends a Disconnect message to SIP gateway2. The Disconnect message starts the call session termination process.
21	BYE—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a BYE request to SIP gateway 1. The BYE request indicates that User B wants to release the call. Because it is User B that wants to terminate the call, the Request-URI field is now replaced with PBX A's SIP URL and the From field contains User B's SIP URL.
22	Disconnect—SIP Gateway 1 to PBXA	SIP gateway 1 sends a Disconnect message to PBX A.
23	Release—SIP Gateway 2 to PBX B	After the call is completed, SIP gateway 2 sends a Release message to PBX B.
24	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
25	200 OK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a 200 OK response to SIP gateway 2. The 200 OK response notifies SIP gateway 2 that SIP gateway 1 has received the BYE request.
26	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2.
27	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	180 Ringing—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that the user is being alerted.
6	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A. The Alert message indicates 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	200 OK—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made.
8	Connect—SIP Gateway 1 to PBX A	SIP 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 SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
10	ACK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that SIP gateway1 has received the 200 OK response. The call session is now active.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A. A two-way RTP channel is established between SIP gateway 1 and SIP IP phone B.
11	BYE—SIP IP Phone to SIP Gateway1	User B terminates the call session at his SIP IP phone and the phone sends a BYE request to SIP gateway 1. The BYE request indicates that User B wants to release the call.
12	Disconnect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
13	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
14	200 OK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends a 200 OK response to the SIP IP phone. The 200 OK response notifies the phone that SIP gateway 1 has received the BYE request.
15	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway 1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	180 Ringing—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that the user is being alerted.
6	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A. The Alert message indicates 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	200 OK—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made.
8	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
9	ACK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends an ACK to the 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 SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A. A two-way RTP channel is established between SIP gateway 1 and SIP IP phone B.
11	INVITE—SIP IP Phone to SIP Gateway 1	User B puts User A on hold. The SIP IP phone sends an INVITE request to SIP gateway 1.
12	200 OK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends a 200 OK response to the SIP IP phone. The 200 OK response notifies the SIP IP phone that the INVITE was successfully processed.
13	ACK—SIP IP Phone to SIP Gateway1	The SIP IP phone sends an ACK to SIP gateway 1. The ACK confirms that the SIP IP phone has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent.
The two-way RTP channel is torn down. The call is on hold.
14	INVITE—SIP IP Phone to SIP Gateway 1	User B takes User A off hold. The SIP IP phone sends an INVITE request to SIP gateway 1.
15	200 OK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends a 200 OK response to the SIP IP phone. The 200 OK response notifies the SIP IP phone that the INVITE was successfully processed.
16	ACK—SIP IP Phone to SIP Gateway1	The SIP IP phone sends an ACK to SIP gateway 1. The ACK confirms that the SIP IP phone has received the 200 OK response. The call session is now active.
At this point, a two-way voice path exists between SIP gateway 1 and PBX A and the two-way RTP channel is reestablished between SIP gateway 1 and SIP IP phone B.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	180 Ringing—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that the user is being alerted.
6	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to User A. The Alert message indicates 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	200 OK—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made.
8	Connect—SIP Gateway 1 to PBX A	SIP 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 SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
10	ACK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that SIP gateway1 has received the 200 OK response. The call session is now active.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A. A two-way RTP channel is established between SIP gateway 1 and SIP IP phone B.
11	BYE—SIP IP Phone to SIP Gateway1	User B transfers User A's call to User C and then hangs up. The SIP IP phone sends a BYE request to SIP gateway 1. The BYE request includes the Also header. In this scenario, the Also header indicates that User C needs to be brought into the call while User B hangs up. This header distinguishes the call transfer BYE request from a normal disconnect BYE request. The Request-By header could be included in the BYE request, however, Cisco's implementation does not require the header to complete the transfer. If the Requested-By header is included, the INVITE sent to the transferred-to party will include the Requested-By header. If the Requested-By header is not included, the INVITE sent to the transferred-to party will not include the Requested-By header.
12	200 OK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends a 200 OK message to the SIP IP phone. The 200 OK response notifies the SIP IP phone that the BYE request has been received. The call session between User A and User B is now terminated.
13	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an INVITE request to SIP gateway 2. In the INVITE request, a unique Call-ID is generated and the Requested-By field indicates that User B requested the call.
14	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User C via PBX B.
15	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP gateway 1. The 100 Trying response indicates that SIP gateway 2 has received the INVITE request but that User C has not yet been located.
16	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2. User C's phone begins to ring.
17	Alerting—PBX B to SIP Gateway 2	PBX B sends an Alert message to SIP gateway 2. The message indicates that the PBX is attempting to alert the user of the call (that is to say, the phone is ringing).
18	180 Ringing—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that SIP gateway 2 has located, and is trying to alert, User C.
19	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to PBX A. The Alert message indicates 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 C is being alerted.
20	Connect—PBX B to SIP Gateway 2	User C answers the phone. PBX B sends a Connect message to SIP gateway 2. The Connect message notifies SIP gateway 2 that the connection has been made.
21	200 OK—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 200 OK response to SIP gateway 1. The 200 OK response notifies SIP gateway 1 that the connection has been made. If User C supports the media capability advertised in the INVITE message sent by User A, it advertises the intersection of its own and User A's media capability in the 200 OK response. If User C does not support the media capability advertised by User A, it sends back a 400 Bad Request response with a 304 Warning header field.
22	Connect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Connect message to PBX A. The Connect message notifies PBX A that the connection has been made.
23	Connect ACK—PBX A to SIP Gateway 1	PBX A acknowledges SIP gateway 1's Connect message.
24	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that SIP gateway 1 has received the 200 OK message from SIP gateway 2.
25	Connect ACK—SIP Gateway 2 to PBX B	SIP gateway 2 acknowledges PBX B's Connect message.
At this point, a two-way voice path is established between SIP gateway 1 and PBX A and between SIP gateway 2 and PBX B. A two-way RTP channel is established between SIP gateway 1 and SIP gateway 2.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an INVITE request to SIP gateway 2. 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. 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 1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBXB.
5	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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 Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
7	Disconnect (Busy)—PBX B to SIP Gateway 2	PBX B sends a Disconnect message to SIP gateway 2. In the Disconnect 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 to SIP Gateway 1	SIP gateway 2 maps the Release message cause code (Busy) to the 486 Busy 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 Gateway 1 to PBX A	SIP gateway 1 sends a Release message to PBX A. User A hears a busy tone.
10	Release—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Release message to PBX B.
11	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
12	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 486 Busy Here response has been received.
13	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2.
14	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an INVITE request to SIP gateway 2. 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBXB.
5	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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.
6	Call Proceeding—PBX B to SIP Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
7	Alerting—PBX B to SIP Gateway 2	PBX B sends an Alert message to SIP gateway 2. The message indicates that the 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 1	SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that SIP gateway 2 has located, and is trying to alert, User B.
9	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to PBX A. The Alert message indicates 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 Gateway 1 to SIP Gateway 2	Because SIP gateway 2 did not return an appropriate response within the time 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 PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
12	Disconnect—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Disconnect message to PBX B.
13	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
14	Release—PBX B to SIP Gateway 2	PBX B sends a Release message to SIP gateway 2.
15	200 OK—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 200 OK response to SIP gateway 2. The 200 OK response confirms that the Cancel request has been received.
16	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A.
17	Release Complete—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Release Complete message to PBX B and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an INVITE request to SIP gateway 2. 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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.
5	Class 4xx/5xx/6xx Failure—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 determines that it does not have any more channels available, 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 PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
7	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
8	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 503 Service Unavailable response has been received.
9	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Redirect Server	SIP gateway 1 sends an INVITE request to the SIP redirect server. 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. 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 to SIP Gateway1	SIP redirect server sends a 302 Moved Temporarily message to SIP gateway 1. The message indicates that User B is not available and includes instructions to locate UserB.
4	ACK—SIP Gateway 1 to SIP Redirect Server	SIP gateway 1 acknowledges the 302 Moved Temporarily response with an ACK.
5	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a new INVITE request to User B. The new INVITE request 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 1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
7	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBXB.
8	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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 Gateway 2	PBX B sends a Call Proceeding message to SIP gateway 2 to acknowledge the Call Setup request.
10	Alerting—PBX B to SIP Gateway 2	PBX B sends an Alert message to SIP gateway 2. User B's phone begins to ring.
11	180 Ringing—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 180 Ringing response to SIP gateway 1. The 180 Ringing response indicates that SIP gateway 2 has located, and is trying to alert, User B.
12	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to PBX A.
13	CANCEL (Ring Timeout)—SIP Gateway 1 to SIP Gateway 2	Because SIP gateway 2 did not return an appropriate response within the time 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 PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
15	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
16	Disconnect—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Disconnect message to PBX B.
17	200 OK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a 200 OK response to SIP gateway 2. The 200 OK response confirms that the CANCEL request has been received.
18	Release Complete—PBX A to SIP Gateway 1	PBX A sends a Release Complete message to SIP gateway 1 and the call session attempt is terminated.
19	Release—PBX B to SIP Gateway 2	PBX B sends a Release message to SIP gateway 2.
20	Release Complete—SIP Gateway 2 to PBX B	SIP gateway 2 sends a Release Complete message to PBX B.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Redirect Server	SIP gateway 1 sends an INVITE request to the SIP redirect server. 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. 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 Redirect Server to SIP Gateway1	The SIP redirect server sends a 300 Multiple Choice response to SIP gateway 1. The 300 Multiple Choice response indicates that the SIP redirect server accepted 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 Redirect Server	SIP gateway 1 acknowledges the 300 Multiple Choice response with an ACK.
5	INVITE—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends a new INVITE request to User B. The new INVITE request 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 Gateway1 to SIP Gateway 2	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
7	100 Trying—SIP Gateway 2 to SIP Gateway 1	SIP gateway 2 sends a 100 Trying response to the INVITE request sent by SIP 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.
8	Class 4xx/5xx/6xx Failure—SIP Gateway 2 to SIP Gateway 1	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 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 PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
10	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
11	ACK—SIP Gateway 1 to SIP Gateway 2	SIP gateway 1 sends an ACK to SIP gateway 2. The ACK confirms that the 404 Not Found failure response has been received.
12	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an INVITE request to the SIP proxy server. 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. 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 SIP Gateway 2	The SIP proxy server checks whether it's own address is contained in the Via 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
5	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with UserB via PBXB.
6	100 Trying—SIP Proxy Server to SIP Gateway 1	The SIP proxy server sends a 100 Trying response to SIP gateway 1.
7	100 Trying—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 100 Trying response to the SIP proxy server.
8	Release Complete (Busy)—PBXBto SIP Gateway2	PBX B sends a Release Complete message to SIP gateway 2. In the Release Complete message, the cause code indicates that User B is busy. The Release Complete message starts the call session termination process.
9	486 Busy Here—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 maps the Release message cause code (Busy) to the 486 Busy 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 Server to SIP Gateway 1	The SIP proxy server forwards the 486 Busy response to SIP gateway 1.
11	Disconnect (Busy)—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
12	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
13	ACK—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an SIP ACK to the SIP proxy server.
14	ACK—SIP Proxy Server to SIP Gateway 2	The SIP proxy server forwards the SIP ACK to SIP gateway 2. The ACK confirms that the 486 Busy Here response has been received.
15	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an INVITE request to the SIP proxy server. 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. 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 SIP Gateway 2	The SIP proxy server checks whether it's own address is contained in the Via 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
5	100 Trying—SIP Proxy Server to SIP Gateway 1	The SIP proxy server sends a 100 Trying response to SIP gateway 1.
6	100 Trying—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 100 Trying response to the SIP proxy server.
7	Class 4xx/5xx/6xx Failure—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 determines that it does not have any more channels available, refuses the connection, and sends a SIP 503 Service Unavailable response to the SIP proxy server.
8	Class 4xx/5xx/6xx Failure—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the SIP 503 Service Unavailable response to SIP gateway 1.
9	Disconnect—SIP Gateway 1 to PBXA	SIP gateway 1 sends a Disconnect message to PBX A.
10	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
11	ACK—SIP Gateway 1 to SIP Proxy Server	SIP Gateway 1 sends an ACK to the SIP proxy server.
12	ACK—SIP Proxy Server to SIP Gateway 2	The SIP proxy server forwards the SIP ACK to SIP Gateway 2. The ACK confirms that the 503 Service Unavailable response has been received.
13	Release Complete—SIP Gateway 1 to PBX A	SIP Gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an INVITE request to the SIP proxy server. 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	INVITE—SIP Proxy Server to SIP Gateway 2	The SIP proxy server checks whether it's own address is contained in the Via 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.
5	Setup—SIP Gateway 2 to PBXB	SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with UserB via PBXB.
6	100 Trying—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a 100 Trying response to the SIP proxy server. The SIP proxy server might or might not forward the 100 Trying response to SIP gateway1.
7	100 Trying—SIP Proxy Server to SIP Gateway 1	The SIP proxy server forwards the 100 Trying response to SIP gateway 1.
8	Release Complete—PBX B to SIP Gateway 2	PBX B sends a Release Complete message to SIP gateway 2. The Release Complete message starts the call session termination process.
9	6xx Failure—SIP Gateway 2 to SIP Proxy Server	SIP gateway 2 sends a class 6xx failure response (a global error) to the SIP proxy 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 to SIP Gateway 1	The SIP proxy server forwards the 6xx failure to SIP gateway 1.
11	Disconnect—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
12	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
13	ACK—SIP Gateway 1 to SIP Proxy Server	SIP gateway 1 sends an ACK to the SIP proxy server.
14	ACK—SIP Proxy Server to SIP Gateway 2	The SIP proxy server sends an ACK to SIP gateway 2. The ACK confirms that the 6xx failure response has been received.
15	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	486 Busy Here—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 486 Busy Here response to SIP gateway 1. The 486 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 Gateway 1 to PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
7	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
8	ACK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that User A has received the 486 Busy Here response. The call session attempt is now being terminated.
9	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	180 Ringing—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 180 Ringing response to SIP gateway 1. The 180Ringing response indicates that the user is being alerted.
6	Alerting—SIP Gateway 1 to PBX A	SIP gateway 1 sends an Alert message to PBX A. The Alert message indicates 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 Gateway 1 to SIP IP Phone	Because SIP gateway 1 did not return an appropriate response within the time 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 PBX A	SIP gateway 1 sends a Disconnect message to PBX A.
9	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.
10	200 OK—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 200 OK response to SIP gateway 1. The 200 OK response confirms that User A has received the Cancel request. The call session attempt is now being terminated.
11	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session is terminated.

Step	Action	Description
1	Setup—PBX A to SIP Gateway1	Call Setup is initiated between PBX A and SIP gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call UserB.
2	INVITE—SIP Gateway 1 to SIP IP Phone	SIP 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. 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 Gateway1 to PBX A	SIP gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setup request.
4	100 Trying—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a 100 Trying response to SIP gateway 1. The 100 Trying response indicates that the INVITE request has been received by the SIP IP phone.
5	4xx/5xx/6xx Failure—SIP IP Phone to SIP Gateway 1	The SIP IP phone sends a class 4xx, 5xx, or class 6xx failure response to SIP gateway1. 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 PBX A	SIP gateway 1 sends a Release message to PBX A.
7	Release—PBX A to SIP Gateway 1	PBX A sends a Release message to SIP gateway 1.
8	ACK—SIP Gateway 1 to SIP IP Phone	SIP gateway 1 sends an ACK to the SIP IP phone. The ACK confirms that UserA has received the 4xx/5xx/6xx failure response. The call session attempt is now being terminated.
9	Release Complete—SIP Gateway 1 to PBX A	SIP gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated.