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Successful Call Flow Scenarios

Successful Call Flow Scenarios

This chapter describes call flows for the following scenarios, which illustrate successful calls:

SIP Gateway-to-SIP Gateway Call

Figure 1 illustrates a successful gateway-to-gateway call setup and disconnect. In this call flow scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to SIP gateway 1 via a T1/E1. User B is located at PBX B. PBX B is connected to SIP gateway 2 via a T1/E1. User B's phone number is 555-0002. SIP gateway 1 is connected to SIP gateway 2 over an IP network.

The call flow scenario is as follows:

    1. User A calls User B.

    2. User B answers the call.

    3. User B hangs up.


Figure 1: SIP Gateway-to-SIP Gateway Call

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.

SIP Gateway-to-SIP Gateway Call via SIP Redirect Server

Figure 2 illustrates a successful gateway-to-gateway call setup and disconnect via a SIP redirect server. In this scenario, the two end users are identified as User A and User B. User A is located at PBX A. PBX A is connected to SIP gateway 1 via a T1/E1. SIP gateway 1 is using a SIP redirect server. User B is located at PBX B. PBX B is connected to SIP gateway 2 via a T1/E1. User B's phone number is 555-0002. SIP gateway 1 is connected to SIP gateway 2 over an IP network.

The call flow scenario is as follows:

    1. User A calls User B via SIP gateway 1 using a SIP redirect server.

    2. User B answers the call.

    3. User B hangs up.


Figure 2: SIP Gateway-to-SIP Gateway Call via SIP Redirect Server

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 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 Gateway 1

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 PBX B

SIP gateway 2 receives the INVITE request from SIP gateway 1 and initiates a Call Setup with User B via PBX B.

7

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.

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 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 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.

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 PBX A

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.

SIP Gateway-to-SIP Gateway Call via SIP Proxy Server

Figure 3 and Figure 4 illustrate a successful gateway-to-gateway call setup and disconnect via a proxy server. In these scenarios, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to SIP Gateway 1 via a T1/E1. SIP Gateway 1 is using a proxy server. SIP Gateway 1 is connected to SIP Gateway 2 over an IP network. User B is located at PBX B. PBX B is connected to SIP Gateway 2 (a SIP gateway) via a T1/E1. User B's phone number is 555-0002.

In the scenario illustrated by Figure 3, the record route feature is enabled on the proxy server. In the scenario illustrated by Figure 4, record route is disabled on the proxy server.

When record route is enabled, the proxy server adds the Record-Route header to the SIP messages to ensure that it is in the path of subsequent SIP requests for the same call leg. The Record-Route field contains a globally reachable Request-URI that identifies the proxy server. When record route is enabled, each proxy server adds its Request-URI to the beginning of the list.

When record route is disabled, SIP messages flow directly through the SIP gateways once a call has been established.

The call flow is as follows:

    1. User A calls User B via SIP gateway 1 using a proxy server.

    2. User B answers the call.

    3. User B hangs up.


Figure 3: SIP Gateway-to-SIP Gateway Call via SIP Proxy Server with Record Route Enabled

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 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 Gateway 1 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 PBX B

SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with User B via PBX B.

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 Gateway 2

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 Gateway 2

After the call is completed, PBX B sends a Disconnect message to SIP gateway 2. 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 User B's SIP URL.

23

BYE—SIP Proxy Server to SIP Gateway 1

The SIP proxy server forwards the BYE request to SIP gateway 1.

24

Disconnect—SIP Gateway 1 to PBX A

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.


Figure 4:
SIP Gateway-to-SIP Gateway Call via a Proxy Server with Record Route Disabled

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 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 Gateway 1 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 PBX B

SIP gateway 2 receives the INVITE request from the SIP proxy server and initiates a Call Setup with User B via PBX B.

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.

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 Gateway 2

After the call is completed, PBX B sends a Disconnect message to SIP gateway 2. 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 PBX A

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.

SIP Gateway-to-SIP IP Phone Calls

This section describes the following SIP Gateway-to-SIP IP phone call flows:

SIP Gateway-to-SIP IP Phone—Call Setup and Disconnect

Figure 5 illustrates a successful gateway-to-SIP IP phone call setup and disconnect. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to SIP gateway 1 via a T1/E1. User B is located at a SIP IP phone. SIP gateway 1 is connected to the SIP IP phone over an IP network.

The call flow is as follows:

    1. User A calls User B.

    2. User B answers the call.

    3. User B hangs up.


Figure 5: SIP Gateway-to-SIP IP Phone—Call Setup and Disconnect

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 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

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 gateway 1 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 Gateway 1

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.

SIP Gateway-to-SIP IP Phone—Call Setup and Call Hold

Figure 6 illustrates a successful gateway-to-SIP IP phone call setup and call hold. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to SIP gateway 1 via a T1/E1. User B is located at a SIP IP phone. SIP gateway 1 is connected to the SIP IP phone over an IP network.

The call flow is as follows:

    1. User A calls User B.

    2. User B answers the call.

    3. User B puts User A on hold.

    4. User B takes User A off hold.


Figure 6: SIP Gateway-to-SIP IP Phone Call—Call Setup and Call Hold

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 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 Gateway 1

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 Gateway 1

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.

SIP Gateway-to-SIP IP Phone—Call Setup and Transfer

Figure 7 illustrates a successful gateway-to-SIP IP phone call setup and call transfer without consultation. In this scenario, there are three end users: User A, User B, and User C. User A is located at PBX A. PBX A is connected to SIP gateway 1 via a T1/E1. User B is located at a SIP IP phone and is directly connected to the IP network. User C is located at PBX B. PBX B is connected to SIP gateway 2 via a T1/E1. SIP gateway 1, SIP gateway 2, and the SIP IP phone are connected to one another over an IP network.

The call flow is as follows:

    1. User A calls User B.

    2. User B answers the call.

    3. User B transfers User A's call to User C and then hangs up.

    4. User C answers the call.


Figure 7: SIP Gateway-to-SIP IP Phone Call—Call Setup and Transfer

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 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

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 gateway 1 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 Gateway 1

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 PBX B

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.

s


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Posted: Wed Sep 11 13:59:43 PDT 2002
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