Available Scenarios

1. Call Barring 

Assesses the ability to restrict certain types of calls like international, outgoing, incoming, premium based on user preferences or network policies. 

SIP Call with Operator Incoming Barring and Announcement

Description: This test scenario validates a SIP call with Operator Incoming Barring enabled, ensuring the Calling Party (UEa) receives appropriate announcements during the call setup, ends, and clears by either UEa or UEb. The scenario includes all relevant SIP messages (183, ACK).
Evaluation Steps: (6)

SIP Call with Operator International Barring and Announcement

Description: This test scenario validates a SIP call with Operator International Barring enabled, ensuring that the Calling Party User Equipment (UEa) receives appropriate announcements played during the international call setup. The scenario includes all relevant SIP messages (183, ACK).
Evaluation Steps: (11)

SIP Call with Operator Outgoing Barring and Announcement

Description: This test scenario validates a SIP call with Operator Outgoing Barring enabled, ensuring that appropriate announcements are played during the call setup by Calling Party User Equipment (UEa). The scenario includes all relevant SIP messages (183, ACK).
Evaluation Steps: (12)

SIP Call with Operator Premium Barring and Announcement

Description: This test scenario validates a SIP call with Operator Premium Barring enabled, ensuring the Calling User Equipment (UEa) receives appropriate announcements during the call setup, ends, and clears by either UEa or UEb. The scenario includes all intermediary SIP messages (183, ACK).
Evaluation Steps: (7)

SIP Call with Operator Premium Short Code Barring and Announcement

Description: This test scenario validates a SIP call with Operator Premium Short Code Barring enabled, ensuring that appropriate announcements are played when shortcodes are called by Calling Party User Equipment (UEa). The scenario includes all relevant SIP messages (183, ACK).
Evaluation Steps: (8)

SIP Call without Operator International Barring

Description: This test scenario validates a SIP call without Operator International Barring, ensuring successful call establishment between users (UEa and UEb), ends, and clears. The scenario includes all relevant SIP messages (100, 180, ACK).
Evaluation Steps: (7)

SIP Call without Operator Premium Barring

Description: This test scenario validates a SIP call without Operator Premium Barring, ensuring successful call establishment with Calling User Equipment (UEa) receiving appropriate announcements during the call setup, ends, and clears by either UEa or UEb with bidirectional speech. The scenario includes all relevant SIP messages (180, 183, ACK).
Evaluation Steps: (8)

2. Call Breakout 

Evaluates call routing from IMS to external networks like PSTN, 2G, 3G, PBX, Short Code or other IMS domains. 

SIP Call 4G to Fixed IMS with Call Duration Failure

Description: This test scenario validates a SIP call from a 4G user (UEa) to a Fixed IMS user (UEb), ensuring that the call duration is monitored and any duration failures are accurately recorded. The scenario includes all relevant SIP messages (180, 183, ACK) and accounting record updates.
Evaluation Steps: (12)

SIP Call VoWiFi to Any User with Call Max Duration Detected

Description: This test scenario validates a SIP call from a Voice over Wi-Fi (VoWiFi) user (UEa) to any user (UEb), ensuring that the call's maximum duration is detected and properly managed. The scenario includes all relevant SIP messages (180, 183, ACK) and accounting record updates.
Evaluation Steps: (14)

VoLTE SIP Call Breakout with Ringing to PSTN

Description: This test scenario validates a SIP call from a 4G user (UEa) to the PSTN, ensuring the call rings and is answered before being cleared. The scenario includes all relevant SIP messages (180, ACK) and captures the call's establishment and clearing after a specified duration.
Evaluation Steps: (8)

3. Call Charging and Billing 

Verifies the correct generation of billing and charging records for various IMS services. 

SIP Call with Session Based Offline Charging Accounting Control

Description: This test scenario validates the call flow between UEa and UEb, focusing on call setup, bidirectional speech validation, and accounting record management. UEa initiates a call, and UEb responds with an INVITE 180 Ringing message. Once the call is successfully answered, the accounting record starts, and UEa acknowledges the call establishment. Bidirectional speech is validated, and UEb or the tTAS sends a SIP UPDATE message, which updates the accounting record. The scenario concludes with the call ending and the accounting record closing.
Evaluation Steps: (11)

SIP Call with Session Based Offline Charging Accounting Control (No Interim Record)

Description: This test scenario validates the call flow between UEa and UEb, focusing on proper call setup, bidirectional speech, and accounting record management. UEa initiates a call, and UEb responds with an INVITE 180 Ringing message. The call is successfully answered, and the accounting record is initiated. After UEa acknowledges the call establishment, bidirectional speech is validated. The scenario concludes with the call ending, and the accounting record is closed.
Evaluation Steps: (9)

4. Call Conference 

Verifies the functionality of multi-party calls, including adding/removing participants and managing conference features. 

3-Party SIP Conference Call - All Participants Leave Conference

Description: This test scenario validates multiple calls from User A, where User A initiates calls to Users B and C, places them on hold, establishes a conference call, and subsequently tears down the call after both Users B and C leave.
Evaluation Steps: (24)

3-Party SIP Conference Call - Called Party (UEb) Initiated

Description: This test scenario validates the process of setting up a terminating conference call involving three users (A, B, and C) across multiple call flows. Initially, User A calls User B, and after a successful call establishment, User B places User A on hold, triggering the MRFC to play an on-hold announcement. User B then initiates a second call with User C, follows the same procedure, and puts User C on hold. Finally, User B initiates a conference call that successfully connects all three participants (A, B, and C). The scenario concludes with User B ending the conference call.
Evaluation Steps: (18)

3-Party SIP Conference Call - Conference Initiator (UEa) Holds/Resumes

Description: This test scenario validates a sequence of call flows originating from User A, encompassing two individual calls and a conference call. User A initiates a call with User B, who responds with an INVITE 180 Ringing. After successfully establishing the call, User A places User B on hold while the MRFC plays an on-hold announcement. User A then resumes the call and initiates a second call to User C, following the same procedure. After resuming the call with User C, User A sets up a conference call that successfully connects all three participants (A, B, and C). User A subsequently places the conference call on hold, prompting the MRFC to play another on-hold announcement, before resuming the conference call.
Evaluation Steps: (22)

3-Party SIP Conference Call - Conference Initiator (UEa) Leaves Conference

Description: This test scenario validates a SIP conference call involving User A and two other users (User B and User C). It includes the call initiation and acceptance, placing users on hold with announcements, and the final termination of the conference call. All intermediary SIP messages and call establishments are accounted for throughout the process.
Evaluation Steps: (18)

3-Party SIP Conference Call - Participant (UEb) Leaves Conference

Description: This test scenario validates a multi-call flow originating from the same User A, involving two separate calls and a conference call. User A first initiates a call with User B, successfully establishes it, and places User B on hold while the MRFC plays an on-hold announcement. User A resumes the call and then initiates a second call with User C, following the same procedure. After resuming the call with User C, User A initiates a conference call that successfully connects all three participants (A, B, and C). The scenario concludes with User B leaving the conference call, and the TAS notifying the MRFC of User B's departure.
Evaluation Steps: (21)

3-Party SIP Conference Call Initiated by Calling Party (UEa) with Different Codecs

Description: This test scenario validates a conference call where User A initiates calls to Users B and C, both of whom respond with INVITE 180 Ringing, followed by successful call establishments and on-hold announcements. Then, with different codecs, User A initiates a conference call, which is acknowledged by all participants before User A ends the call.
Evaluation Steps: (19)

SIP Conference Call Between 3 Parties Initiated by Calling Party with Participant on Hold

Description: This test scenario validates flows from User A, where User A initiates calls to Users B and C. Both respond with INVITE 180 Ringing, and after successful call establishments, User A places both calls on hold with announcements. User A then initiates a conference call, which is acknowledged by all participants. Subsequently, User B places the conference on hold and finally resumes it.
Evaluation Steps: (20)

5. Call Media 

Evaluates the quality and handling of voice and video streams, including codec negotiation, media path establishment and dual tone multi frequency. 

SIP Call 4G to Fixed IMS with Bidirectional DTMF

Description: This test scenario validates a SIP call between a User Equipment (UE) on an LTE network (UEa) and a User Equipment on a Fixed IMS network (UEb) with bidirectional DTMF (Dual-Tone Multi-Frequency) tone exchange. It includes the initiation of a call by UEa with a DTMF tone offer, the responses from UEb, and the successful acknowledgment of call establishment. The scenario verifies that DTMF tone support is confirmed and that INFO messages containing DTMF tones are exchanged successfully between both UEs during the call.
Evaluation Steps: (11)

SIP Call AMR-WB to G.722 Transcoding

Description: This test scenario validates SIP calls between two users, User UEa, UEb, and vice versa, ensuring successful call establishment and transcoding between Adaptive Multi-Rate Wideband (AMR-WB) and G.722 codecs. The scenario includes relevant SIP messages (100, 180, ACK) and the validation of bidirectional speech and codec transcoding for both calls.
Evaluation Steps: (17)

SIP Call with AMR-WB to EVS Codec Negotiation

Description: This test scenario validates two SIP call flows between Calling Party (UEa) and Called Party (UEb). It ensures the INVITE message from UEa contains the expected codec, that the MRFC receives and supports the codec, returns the appropriate response code and description, and successfully receives an ACK. The same checks apply when UEb initiates a call to UEa.
Evaluation Steps: (17)

SIP Call with Barring All Outgoing Calls Announcement

Description: This test scenario validates a SIP call where a User Equipment (UEa) initiates a premium call with outgoing call barring enabled. The scenario outlines the responses received from the Network Element (NE), including the INVITE 183 Session Progress message containing a specific cause field and message. It verifies that the NE sends an INFO message with an audio file for the announcement and that the Media Resource Function Controller (MRFC) responds accordingly. Finally, the test confirms that the NE ends or clears the call, demonstrating the effect of the barring on the outgoing call attempt.
Evaluation Steps: (8)

SIP Call with EVS Codec Announcement

Description: This test scenario validates a SIP call flow where Call Party (UEa) initiates a call with Called Party (UEb), ensuring that the INVITE message from UEa contains the expected EVS codec. The scenario verifies that the MRFC receives the INVITE with the expected codec, returns the appropriate response code and description, supports the specified codec, and successfully receives an ACK.
Evaluation Steps: (9)

SIP Call with Hold and Resume Announcement

Description: This test scenario validates a SIP call where Calling User (UEa) initiates a call with Called User (UEb), including hold and resume functionality. It covers the successful call establishment, validation of bidirectional speech, acceptance of hold, and playback of on-hold announcements (628_de_call_hold.wav or 629_en_call_hold.wav), followed by resuming the call and ensuring speech validation after the resumption.
Evaluation Steps: (14)

SIP Call With Calling Party Originating Identification Restriction

Description: This test scenario validates an anonymous call scenario, where UEa initiates a call with UEb. UEa receives an INVITE 183 Session Progress with specified cause codes, and {ne} sends an INFO message, leading to a response from MRFC and {ne} before UEa acknowledges the call failure or rejection.
Evaluation Steps: (8)

6. Call Continuity 

Assesses the seamless handover of calls between different access networks like VoLTE to VoWiFi, VoLTE to 2G, VoNR to 3G. 

7. Call Setup 

Verifies SIP Call initiation, establishment, quality of service and termination of voice and video calls originating from 4G, 5G, Fixed IMS and Voice over Wi-Fi clients. 

SIP Call with Calling Party Canceling

Description: This test scenario validates a SIP call from a Calling User Equipment (UEa) to a Called User Equipment (UEb), where the calling party cancels the call attempt before it is answered. It includes Call Duration and SIP message (487).
Evaluation Steps: (7)

SIP Call with Invite Timer Expiry and Offline Charging

Description: This test scenario validates a SIP call where UEa initiates a call to UEb, focusing on the call flow and invite timer expiration after a specified duration. It includes all intermediary SIP messages (100 Trying, 180 Ringing), the call establishment process, accounting record management, an UPDATE request during the call, and eventual call termination triggered by the invite timer expiry.
Evaluation Steps: (15)

VoNR SIP Call to Digital Subscriber Line

Description: This test scenario validates a SIP call from a Calling New Radio (NR) User Equipment (UEa) to a Called Digital Subscriber Line (DSL) User Equipment (UEb), including all intermediary SIP messages (100, 180, 200, ACK).
Evaluation Steps: (9)

8. Call Supplementary Services 

Checks various call forwarding scenarios like unconditional, busy, no answer, not reachable, call waiting, call hold, and caller ID 

Communication Forwarding on No Reply (CFNR)

Description: This test scenario validates the forwarding of a call to another user when User A does not respond within a specified time.
Evaluation Steps: (13)

SIP Call with Call Forwarding Unconditional (CFU)

Description: This test scenario validates a call flow between UEa and UEb, where the Network Element (NE) intermediates the call setup by forwarding the INVITE messages to UEc on behalf of UEa. The scenario ensures proper SIP messaging, including specific response codes (181, 183, 180, 200) and corresponding descriptions for "Call Being Forwarded," "Session Progress," "Ringing," and "OK." It checks that the correct SIP header field values are present in the responses, bidirectional speech is validated, and one participant successfully terminates the call.
Evaluation Steps: (10)

SIP Call with Call Waiting and Answer

Description: This test scenario validates SIP calls between User A, User B, and User C, including intermediary SIP messages (183 Session Progress, 180 Ringing, 200 OK, ACK), with call waiting and resumption features.
Evaluation Steps: (16)

VoLTE SIP Call - Called Party Rejects

Description: This test scenario validates a SIP call from UEa to UEb, both on LTE networks, where the called party (UEb) rejects the call. It includes all intermediary SIP messages (183 Session Progress, 180 Ringing), followed by UEb's rejection and UEa's acknowledgment of the failure.
Evaluation Steps: (8)

9. Network Resilience 

Verifies network load balancing, failover, and availability across control and user plane network functions. 

SIP Call with TAS Failover

Description: This test scenario validates the resilience of a SIP call from a Calling User Equipment (UEa) to a Called User Equipment (UEb) in the event of a Telephony Application Server (TAS) failover. The scenario tests call continuity, where UEa attempts to connect with UEb, but the call setup is interrupted due to TAS failover. This includes handling call cancellation and ensuring that SIP messages, including the "487 Request Terminated" message, are properly managed in response to the failover.
Evaluation Steps: (10)

10. User Registration 

Verifies IMS registration, re-registration, de-registration, third-party registration, de-registration processes, including authentication and authorization. 

User Registration with IMS-AKA Algorithm Parameter

Description: This test scenario validates the SIP registration process for a User Equipment (UE) on an LTE network, ensuring successful registration using the IMS-AKA algorithm. The scenario includes the UE's registration attempt, authentication with the IMS-AKA algorithm, successful registration, registration refresh, and de-registration.
Evaluation Steps: (14)

User Registration without IMS-AKA Algorithm Parameter

Description: This test scenario validates the authentication process, where the User Equipment (UE) initially attempts registration without the IMS-AKA algorithm, subsequently reattempting with the IMS-AKA to ensure successful registration.
Evaluation Steps: (14)

User Registration On 4G - LTE

Description: This test scenario validates a SIP registration flow from a User Equipment (UE) on an LTE network, confirming that the specified authentication algorithm is used and that the registration process completes successfully.
Evaluation Steps: (8)

Residential User Registration with MD5 Optimized Digest

Description: This test scenario verifies the SIP registration flow for a User Equipment (UE), focusing on proper handling of authentication across initial registration, re-registration, and de-registration. The scenario checks the presence of the Authorization header, validates responses to Diameter messages (UAR and SAR), and ensures that the SCSCF only challenges authentication on the initial registration.
Evaluation Steps: (22)

User Registration Rejected with 480/550 Error

Description: This test scenario validates the SIP registration flow for a residential User Equipment (UE), ensuring proper implementation of the Authorization header fields during registration attempts. It confirms that appropriate Diameter responses are received and that registration failures trigger the expected authentication challenges.
Evaluation Steps: (8)

Third-Party Registration with Two Subscribers in Different TAS

Description: This test scenario validates two SIP registration flows for distinct subscribers on an LTE network, ensuring successful registration of each UE and that the Third-Party Registration (TP-REGISTRATION) for both completes successfully across separate TAS instances.
Evaluation Steps: (10)

User Registration with 2 Telephony Application Servers (TAS) During Failover

Description: This test scenario validates the SIP registration process, ensuring successful registration of a UE and TP-REGISTRATION across consecutive TAS instances during failover. It confirms that the TAS handling each request differs, verifying seamless failover processing.
Evaluation Steps: (5)

Residential User Refresh Registration with No Authorization Challenge

Description: This test scenario validates the SIP registration flow for a residential User Equipment (UE), focusing on initial registration, refresh, and de-registration phases without an authorization challenge. The scenario confirms appropriate responses to SAR messages, MD5 algorithm compliance, and the presence of the Authorization header as expected.
Evaluation Steps: (19)

Residential User Refresh Registration Using Optimized Digest Authentication

Description: This test scenario verifies the SIP registration flow for a residential User Equipment (UE), emphasizing consistency in the sip.Via.sent-by.address field in the initial registration request and applying optimized digest authentication. Successful completion is confirmed by the correct Diameter responses at each stage.
Evaluation Steps: (11)

11. User Roaming 

Assesses user access to IMS services while roaming on different visited networks. 

VoLTE SIP Call with Roaming Enabled Redirects Call Due to Call Forwarding Unconditional (CFU)

Description: This test scenario validates a SIP call from a User Equipment (UE) on an LTE network with unconditional call forwarding (CFU). It covers the forwarding of the call from UEa to UEb, including intermediary responses (181, 183, 180, and 200 OK), and confirms that UEa acknowledges the call establishment before one participant ends the call.
Evaluation Steps: (13)