Test Plan One

Name:                             Know-Endpoint to Known-Endpoint

Domains                End-User Devices

Devices Required:          Any Two End User Devices

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                The IP address of the Target is know to the Originator

Description: The device placing the call resolves the dialed telephone number to the IP Address of the device it is calling.  The calling device must properly signal the target device.  The target device answers the call and a full-duplex media path is established between the devices.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Name:                             Know-Endpoint to Known-Endpoint

Domains                End-User Devices

Devices Required:          Any Two End User Devices

Protocols:             H.323

Codecs:                  G711, G729, G723

Assumptions:                The IP address of the Target is know to the Originator

Description: The device placing the call resolves the dialed telephone number to the IP Address of the device it is calling.  The calling device must properly signal the target device.  The target device answers the call and a full-duplex media path is established between the devices.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Two

Name:                             Endpoint Registration

Domains:               End-User Devices
                              SIP

Devices Required:          One SIP End-User Device
                              One REGISTRAR Server

Protocols:             SIP

Codecs:                  N/A

Assumptions:                None

Description: The device is provisioned with the IP address of the REGISTRAR server. When the connection to the network is established, the user agent client sends REGISTER request to the REGISTRAR server. The REGISTRAR server should accept an incoming request and put the device into the location database.

Test Plan Three

Name:                             Endpoint Registration

Domains:               End-User Devices
                              H.323

Devices Required:          One H.323 End-User Device
                              One Gatekeeper

Protocols:             H.323/H.225RAS

Codecs:                  N/A

Assumptions:                Dynamic Gatekeeper Discovery is disabled or not active

Description: The device is provisioned with the IP address of the H.323 Gatekeeper. When the connection to the network is established, the endpoint sends registration (RRQ) request to the H.323 Gatekeeper. Endpoint registers it’s alias with the Gatekeeper for further routing to this endpoint.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. After 2 minutes, the H.323 gatekeeper unregisters endpoint (URQ).

Test Termination: Upon the receipt of gatekeeper’s unregister message, the endpoint will re-registers with the gatekeeper.

Test Plan Four

Name:                             Gatekeeper Routed Call with separate H,245 Channel Establishment

Domains:                        H.323 End-User Devices
H.323 Gatekeeper

Devices Required:          Two H.323 End-User Devices
                              One Gatekeeper

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: Both originating and terminating endpoint are registered with the same gatekeeper.

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. The gatekeeper matches incoming request with pre-registered alias of terminating endpoint and completes the call to the terminating endpoint. The endpoints negotiate H.245 capabilities and establish full duplex voice communication using one of the coders.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Five

Name:                             Gatekeeper Routed Call using Fast Start Method

Domains:                        H.323 End-User Devices
H.33 Gatekeeper

Devices Required:          Two H.323 End-User Devices
                              One Gatekeeper

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: Both originating and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. The gatekeeper matches incoming request with pre-registered alias of terminating endpoint and completes the call. When the terminating endpoint answers the call, the full duplex communication is established between two endpoints

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Six

Name:                             Gatekeeper Routed Call using H.245 Tunneling Method

Domains:                        H.323 End-User Devices
H.323 Gatekeeper

Devices Required:            Two H.323 End-User Devices
            One Gatekeeper

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: Both originating and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. The gatekeeper matches incoming request with pre-registered alias of terminating endpoint and completes the call to the terminating endpoint. When the terminating endpoint answers the call, the full duplex communication is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Seven

Name:                             Gatekeeper Routed Call when Invalid E164 number is dialed

Domains:                        H.323 End-User Devices
H.323 Gatekeeper

Devices Required:          Two H.323 End-User Devices
                              One Gatekeeper

Protocols:             H.323/H.245

Codecs:                  N/A

Assumptions: Both originating and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. However the Gatekeeper cannot resolve E.164 and sends Reject (ARJ) back to the client.

Test Termination: The endpoint device generates re-order tone. The call is properly tear down and is ready to accept new calls.

Test Plan Eight

Name:                             Gatekeeper Routed Call with terminating device is busy

Domains:                        H.323 End-User Devices
H.323 Gatekeeper

Devices Required:          Two H.323 End-User Devices
                              One Gatekeeper

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: Both originating and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. However the terminating device either is busy or does not respond to H.225 call setup message.

Test Termination: The endpoint device generates busy tone. The call is properly tear down and is ready to accept new calls.

Test Plan Nine

Name:                             SIP Redirect Call

Domains:                        SIP End-User Devices
SIP

Devices Required:            Two SIP End-User Devices
            One Redirect Server

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The originating endpoint will submit INVITE request to SIP redirect server. The redirect server will send back temporary move response to the originating endpoint (Response code 302). The originating device will then acknowledge the change and re-issue another INVITE message to the resolved entity. The terminating endpoint ACKs the INVITE messages and a full duplex voice path is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Ten

Name:                             SIP Proxied Call

Domains:                        SIP End-User Devices
SIP

Devices Required:            Two SIP End-User Devices
            One Redirect Server

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions: 1. The originating and terminating device are in the same domain.
2. The SIP Proxy server can be configured to dynamically change route selection via LDAP requests to Location Server, SQL Database or other methods.

Description: The originating device will submit an INVITE request to the outbound SIP proxy server. The server will perform routing lookup and issue an INVITE to resolved location. The termination endpoint accepts the connection and full duplex voice path is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Eleven

Name:                             SIP Proxied Call Across 2 Proxy Servers

Domains:                        End User Devices
SIP

Devices Required:            Two SIP End-User Devices
            Two Proxy Servers
Location Server

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions: The SIP Proxy server can be configured to dynamically change route selection via LDAP requests to Location Server, SQL Database or other methods.

Description: The originating device will submit INVITE request to outbound SIP proxy server. The server determines that the request is not for it’s authoritative domain and forward the request to another SIP Proxy Server. The second SIP proxy server contacts the location server to determine the final location of the endpoint. The termination endpoint accepts the connection and full duplex voice path is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Twelve

Name:                             SIP Proxied Call with unresolvable destination

Domains:                        End User Devices
SIP

Devices Required:            Two SIP End-User Devices
            One Proxy Server

Protocols:             SIP

Codecs:                  N/A

Assumptions:                None

Description: The originating device will submit INVITE request to outbound SIP proxy server. The server will perform routing lookup and issue 6XX type response back to the originating endpoint rejecting the request.

Test Termination: The endpoint device generates re-order tone. The call is properly tear down and is ready to accept new calls.

 

Test Plan Thirteen

Name:                             SIP Proxied Call with busy destination

Domains:                        End User Devices
SIP

Devices Required:            Two SIP End-User Devices
            One Proxy Server
One Location Server

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions: The location server has a CPL script associated with registered endpoint. The CPL call-busy action will be triggered to redirect the server to voice mail.

Description: The originating device will submit INVITE request to outbound SIP proxy server. The server will consult location server to perform resolution for terminating endpoint. However, it is determined that the endpoint is busy.

Test Termination: The endpoint is redirected to voice mail application and the caller can leave voice mail. The call is properly tear down and is ready to accept new calls.

 

Test Plan Fourteen

Name:                             IP to Phone (no FastStart , pre-assigned  PSTN Gateway)

Domains:                        H.323 End-User Devices
H.323

Devices Required:            One H.323 End-User Devices
One H.323 Gateway

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions:                The endpoint is pre-configured with the address of an H.323 PSTN Gateway

Description: After the telephone number is dialed, the endpoint sends the call to the gateway. The gateway accepts the incoming H.225 call setup message and routes the call for PSTN termination. The call is completed and the originating endpoint hears call progress tones from PSTN.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Fifteen

Name:                             Phone to IP (FastStart, Gatekeeper Routed)

Domains:                        H.323 End-User Devices
H.323

Devices Required:            One H.323 End-User Devices
One H.323 Gatekeeper
One H.323 Gateway

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: 1. The endpoint is already registered with the gatekeeper.
2. Both originating gateway and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. The incoming request will be originating from PSTN by the user dialing the number that will be terminated on H.323 Gateway. The gateway asks the gatekeeper to match incoming request with pre-registered alias of terminating endpoint and completes the call to the terminating endpoint. Full duplex communication is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Sixteen

Name:                             IP to Phone (FastStart ,Gatekeeper Routed)

Domains:                        H.323 End-User Devices
H.323

Devices Required:            One H.323 End-User Devices
One H.323 Gatekeeper
One H.323 Gateway

Protocols:             H.323/H.245

Codecs:                  G711, G729, G723

Assumptions: 1. The gateway is already registered with the gatekeeper.
2. Both originating gateway and terminating endpoint are registered with the same gatekeeper

Description: The endpoint asks (ARQ) the gatekeeper to be admitted into the network. The origination endpoint dials a regular PSTN telephone number. Upon the receipt of H225 message (SETUP), the gatekeeper will match incoming request with pre-registered alias of terminating gateway and routes the call to the terminating gateway. The gateway places the call into PSTN and full duplex communication between the originating endpoint and PSTN destination is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Seventeen

Name:                             IP to Phone (Peer-to-Peer)

Domains:                        End-User Devices
SIP

Devices Required:            One SIP End-User Devices
One SIP Gateway

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                The endpoint is pre-configured with and address of SIP/PSTN Gateway.

Description: After the telephone number is dialed, the endpoint originates INVITE method to place the call to PSTN gateway. The gateway acknowledges INVITE message and routes the call for PSTN termination. The call is completed and the originating endpoint hears call progress tones from PSTN.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Eighteen

Name:                             Phone to IP (Proxy Routed)

Domains:                        End-User Devices
SIP

Devices Required:            One SIP End-User Devices
One SIP Registrar
One Location Server
One SIP Proxy Server
One SIP Gateway

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                The endpoint is already registered with SIP registrar server.

Description: The incoming request is originating from PSTN by dialing the number that will be terminated on SIP PSTN Gateway. The gateway asks the SIP Proxy to route incoming request. The SIP proxy server queries the location server to determine the proper location for the terminating endpoint. When the endpoint location is resolved, the SIP gateway sends an INVITE message to the terminating endpoint. Full duplex voice communication between the originating PSTN caller and IP endpoint is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

 

Test Plan Nineteen

Name:                             IP to Phone Proxy Routed Call

Domains:                        End-User Devices
SIP

Devices Required:            One SIP End-User Devices
One SIP Proxy Server
One SIP Gateway

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The originating endpoint dials a regular PSTN number. Upon the receipt of an INVITE message, the SIP Proxy will route the request to the SIP PSTN gateway. The gateway places the call into the PSTN and acknowledges the INVITE message. Full duplex communication between the originating endpoint and the PSTN destination is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty

Name:                             IP to Phone

Domains:                        End-User Devices
Softswitch

Devices Required:            One SIP End-User Devices
One SIP Softswitch
One MGCP Media Gateway

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The originating endpoint dials a regular PSTN number. Upon the receipt of the INVITE message, the softswitch will route the call to the responsible media gateway. The media gateway places the call into PSTN and acknowledges.  A full duplex communication between the originating endpoint and the PSTN destination is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty One

Name:                             Phone to IP

Domains:                        End-User Devices
Softswitch

Devices Required:            One SIP End-User Devices
One SIP Softswitch
One Location Server
One MGCP Media Gateway (trunking)

Protocols:             SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives the call from PSTN. The media gateway reports the event to the softswitch using an MGCP command. The softswitch asks the location server to resolve the destination for the termination. The location server resolves the terminating endpoint to be an IP device and sends an INVITE message to the terminating endpoint. The full duplex communication is established between the PSTN and IP endpoints.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Two

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch

Devices Required:            Two PSTN Phones
Two SIP Softswitch
Two MGCP Media Gateway

Protocols:             SIP/SIP-T

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the softswitch. The softswitch performs route resolution and determines that in order to complete the call, a connection to another softswitch needs to be established. The softswitch establishes a SIP-T connection between the softswitches. The terminating softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Three

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch

Devices Required:            Two PSTN Phones
One SIP Softswitch
One MGCP Media Gateway

Protocols:             SIP/SIP-T

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the softswitch. The softswitch performs route resolution and determines that in order to complete the call, the call needs to go through the originating media gateway. The softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Four

Name:                             IP to Phone

Domains:                        End-User Devices
Softswitch

Devices Required:            One H.323 End-User Devices
One H.323 Softswitch
One MGCP Media Gateway

Protocols:             H.323

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The origination endpoint dials a regular PSTN number. Upon the receipt of H225(SETUP) message, the softswitch will route the call to the responsible media gateway. The media gateway places the call into the PSTN and acknowledges. A full duplex communication between originating endpoint and the PSTN destination is established.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Five

Name:                             Phone to IP

Domains:                        End-User Devices
Softswitch

Devices Required:            One H.323 End-User Devices
One H.323 Softswitch
One MGCP Media Gateway (trunking)

Protocols:             H.323

Codecs:                  G711, G729, G723

Assumptions:                The endpoint is already register with the gatekeeper

Description: The media gateway receives the call from the PSTN. The media gateway reports the event to the softswitch using an MGCP command. The softswitch asks the H.323 gatekeeper to resolve the destination for the termination. The H.323 Gatekeeper resolves the terminating endpoint to be an IP device and sends an H.225 (SETUP) message to the terminating endpoint. The full duplex communication is established between the PSTN and IP endpoints.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Six

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch

Devices Required:            Two PSTN Phones
Two H.323 Softswitch
Two MGCP Media Gateway

Protocols:             H.323

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the softswitch. The softswitch performs route resolution and determines that in order to complete the call, a connection to another softswitch needs to be established. The softswitch establishes th connection between the softswitches. The terminating softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Seven

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch

Devices Required:            Two PSTN Phones
Two H.323 Softswitch
One MGCP Media Gateway

Protocols:             H.323

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the softswitch. The Softswitch performs route resolution and determines that in order to complete the call, the call needs to go through the originating media gateway. The softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

 

Test Plan Twenty Eight

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch (Multi-Stack)

Devices Required:            One PSTN Phone
One H.323 Softswitch
One SIP Softswitch
Two MGCP Media Gateway
            One H.323 Phone

Protocols:             H.323 and SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the SIP softswitch. The softswitch performs route resolution and determines that in order to complete the call, the connection to another softswitch needs to be established. However, the other softswitch only understands H.323. The originating softswitch establishes the connection between the two softswitches. The terminating softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The Media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

 

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan Twenty Nine

Name:                             Phone to Phone

Domains:                        PSTN Phone
Softswitch (Multi-Stack)

Devices Required:            One PSTN Phone
One H.323 Softswitch
One SIP Softswitch
Two MGCP Media Gateway
            One H.323 Phone

Protocols:             H.323 and SIP

Codecs:                  G711, G729, G723

Assumptions:                None

Description: The media gateway receives a call from the PSTN. The media gateway reports the event to the H.323 softswitch. The softswitch performs route resolution and determines that in order to complete the call, a connection to another softswitch needs to be established. However, the other softswitch only understands SIP. The originating softswitch establishes the connection between the two softswitches. The terminating softswitch sends an MGCP message to the media gateway. The media gateway dials into the PSTN and informs the softswitch that the connection is complete. The media gateway establishes full duplex communication between the originating PSTN line and the terminating PSTN line.

 

Test Duration:  The call should stay active indefinitely, but for the purpose of the test we will require the call to stay active for two minutes. 

Test Termination: The call is properly tear down and both devices are ready to accept new calls, when either side terminates the call.

Test Plan _______________

Name:                            

Domains:                       

Devices Required:           

Protocols:            

Codecs:                 

Assumptions:               

Description:

Test Duration:

Test Termination: