Test Scenarios¶
Sipfront offers a variety of test scenarios to cover different testing needs, from simple signaling checks to complex multi-party call setups involving browsers, mobile apps, mobile phones and SIP/PSTN endpoints.
The table below provides an overview of all available test scenarios, a short description of what they do, and which agent capabilities are involved for each agent slot used in the scenario.
Important
Depending on the test scenario, different parameters and conditions might be available. See Test Parameters and Test Conditions for more details.
| Name | Description | Necessary Clients |
|---|---|---|
| Desktop Browser - Calling Party Only | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. | Browser |
| Desktop Browser - Incoming Call with SIP/PSTN Caller | Launch a browser as called party and perform certain user interactions based on the CodeceptJS test script you provide. Then, initiate a call from a generic calling party to the browser, accept the call and measure SIP and RTP statistics. | Browser, SIP Client |
| Desktop Browser - Incoming Call with SIP/PSTN Caller and Mobile Callee | Launch a browser as called party and perform certain user interactions based on the CodeceptJS test script you provide. One calling party and one called party are measuring SIP and RTP statistics. | Mobile Phone, Browser, SIP Client |
| Desktop Browser - Incoming Call with SIP/PSTN Caller and SIP/PSTN Callee | Launch a browser as called party and perform certain user interactions based on the CodeceptJS test script you provide. One calling party and one called party are measuring SIP and RTP statistics. | Browser, SIP Client (x2) |
| Desktop Browser - Incoming Call with SIP/PSTN Caller and two Mobile Callees | Launch a browser as called party and perform certain user interactions based on the CodeceptJS test script you provide. One calling party and one called party are measuring SIP and RTP statistics. | Mobile Phone (x2), Browser, SIP Client |
| Desktop Browser - Outgoing Call with 2 SIP/PSTN Callees | Launch a browser as calling party on chosen OS in Browserstack and perform certain user interactions based on the CodeceptJS test script you provide. Two called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | SIP Client (x2), Browser |
| Desktop Browser - Outgoing Call with Browser Callee | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One additional browser is registered and waiting to answer the call and measure SIP and RTP statistics. | Browser (x2) |
| Desktop Browser - Outgoing Call with Browser & SIP/PSTN Called Party | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One browser and one SIP called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | Browser (x2), SIP Client |
| Desktop Browser - Outgoing Call with Mobile App Callee | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One additional mobile app is registered and waiting to answer the call and measure SIP and RTP statistics. | Mobile App, Browser |
| Desktop Browser - Outgoing Call with Mobile App & SIP/PSTN Callees | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One mobile app and one SIP called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | Mobile App, SIP Client, Browser |
| Desktop Browser - Outgoing Call with SIP/PSTN Callee | Launch a browser as calling party and perform certain user interactions based on the CodeceptJS test script you provide. The called party answers the call and measures SIP and RTP statistics. | SIP Client, Browser |
| DNS Request | Perform single DNS request | DNS Client |
| Media analysis for Audio Call | Register a called party, then launch an audio call from a calling party and measure round-trip times, jitter and packet loss of the RTP audio packets. | SIP Client (x2) |
| Media analysis for Audio Call - Called Party Only | Register a called party, then wait for an incoming audio call and stream an announcement to the calling party while measuring round-trip times, jitter and packet loss of the RTP audio packets. | SIP Client |
| Media analysis for Audio Call - Called Party w/ AI Voice Bot | Register a called party voice AI bot, then wait for an incoming audio call and let the bot talk to the calling party while measuring round-trip times, jitter and packet loss of the RTP audio packets. | SIP Client |
| Media analysis for Audio Call - Called Party w/ TTS | Register a called party, then wait for an incoming audio call and stream an announcement to the calling party with optional text-to-speech input. | SIP Client |
| Media analysis for Audio Call - Calling Party Only | Launch an audio call from a calling party and measure round-trip times, jitter and packet loss of the RTP audio packets. | SIP Client |
| Media analysis for Audio Call - Calling Party w/ AI Voice Bot | Launch call from a calling party with an AI voice bot. | SIP Client |
| Media analysis for Audio Call - Calling Party w/ TTS | Launch call from a calling party with optional text-to-speech input. | SIP Client |
| Media analysis for Audio Call - One calling party and three called parties | Register three called parties, then launch an audio call from a calling party to one of the called parties and measure round-trip times, jitter and packet loss of the RTP audio packets. Using call actions, the call can be transferred to the second called party, or a second call to the second called party can be made. | SIP Client (x4) |
| Media analysis for Audio Call - One calling party and two called parties | Register two called parties, then launch an audio call from a calling party to one of the called parties and measure round-trip times, jitter and packet loss of the RTP audio packets. Using call actions, the call can be transferred to the second called party, or a second call to the second called party can be made. | SIP Client (x3) |
| Media analysis for Audio Call, One calling party, One reject called party, One called party | Register two called parties, then launch an audio call from a calling party and reject it with a given code and reason on one called party, expect the call to be routed to the other called party. (e.g. for 302 tests) | SIP Client (x2), SIP Load Client |
| Media analysis for Audio Call, Reject called party | Register a called party, then launch an audio call from a calling party and reject it with a given code and reason. | SIP Load Client, SIP Client |
| Media analysis for Audio Call, Reject called party - Called Party Only | Register a called party, expect an upstream audio call from a calling party and reject it with a given code and reason. | SIP Load Client |
| Media analysis for Audio Call - Two calling parties and one called party | Register one called party, then launch an audio call from two calling parties to the called parties and measure round-trip times, jitter and packet loss of the RTP audio packets. | SIP Client (x3) |
| Media analysis for Audio Call w/ Packet Capture | Register a called party, then launch an audio call from a calling party and measure round-trip times, jitter and packet loss of the RTP audio packets along with an Auxiliary Agent for capturing traffic. | Packet Capture Client, SIP Client (x2) |
| Media analysis for Audio Call w/ TTS | Register a called party, then launch an audio call from a calling party and measure round-trip times, jitter and packet loss of the RTP audio packets; with optional text-to-speech input in the caller side | SIP Client (x2) |
| Media analysis for Fax Call | Register a called party, then launch a fax call from a calling party and measure round-trip times, jitter and packet loss of the RTP packets. | SIP Fax Client (x2) |
| Media analysis for Fax Call - Called Party only | Register a called party and wait for a fax call from a calling party and measure round-trip times, jitter and packet loss of the RTP packets. | SIP Fax Client |
| Media analysis for Fax Call - Calling Party only | Launch a fax call from a calling party and measure round-trip times, jitter and packet loss of the RTP packets. | SIP Fax Client |
| Mobile App - Calling Party Only | Launch a mobile app and perform certain user interactions based on the CodeceptJS test script you provide. | Mobile App |
| Mobile App - Incoming Call with 2 SIP/PSTN Callers | Launch a mobile app as called party and perform certain user interactions based on the CodeceptJS test script you provide. Two calling parties measuring SIP and RTP statistics. | Mobile App, SIP Client (x2) |
| Mobile App - Incoming Call with SIP/PSTN Caller | Launch a mobile app as called party and perform certain user interactions based on the CodeceptJS test script to log in and wait for an inbound call. Then, initiate a call from a generic calling party and measures SIP and RTP statistics. | Mobile App, SIP Client |
| Mobile App - Outgoing Call with 2 SIP/PSTN Callees | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script you provide. Two called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | SIP Client (x2), Mobile App |
| Mobile App - Outgoing Call with Browser Callee | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One browser app is registered and waiting to answer the call and measure RTCP statistics. | Browser, Mobile App |
| Mobile App - Outgoing Call with Browser & SIP/PSTN Callees | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One browser and one SIP called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | Browser, SIP Client, Mobile App |
| Mobile App - Outgoing Call with Mobile App Callee | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One additional mobile app is registered and waiting to answer the call and measure SIP and RTP statistics. | Mobile App (x2) |
| Mobile App - Outgoing Call with Mobile App & SIP/PSTN Callees | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script you provide. One mobile app and one SIP called parties are registered and waiting to answer the call and measure SIP and RTP statistics. | Mobile App (x2), SIP Client |
| Mobile App - Outgoing Call with SIP/PSTN Callee | Launch a mobile app as calling party and perform certain user interactions based on the CodeceptJS test script to trigger an outbound call. The called party answers the call and measures SIP and RTP statistics. | SIP Client, Mobile App |
| Mobile call - Called Party Only | Wait for and answer incoming mobile calls via a connected mobile phone and record and analyze the audio. | Mobile Phone |
| Mobile call - Call from Mobile to Mobile | Launch a mobile call via a connected mobile phone to a connected mobile phone | Mobile Phone (x2) |
| Mobile call - Call from Mobile to SIP | Launch a mobile call via a connected mobile phone to a SIP account, record the audio and analyze SIP and media statistics. | SIP Client, Mobile Phone |
| Mobile call - Call from SIP to Mobile | Launch a call from a SIP account to a connected mobile phone, record the audio and analyze SIP and media statistics. | Mobile Phone, SIP Client |
| Mobile call - Calling Party Only | Launch a mobile call via a connected mobile phone and record and analyze the audio. | Mobile Phone |
| Mobile call - One calling party and two called parties, one mobile | Register two called parties, one mobile, then launch an audio call from a calling party to one of the called parties and measure round-trip times, jitter and packet loss of the RTP audio packets. Using call actions, the call can be transferred to the second called party, or a second call to the second called party can be made. | SIP Client (x2), Mobile Phone |
| Mobile call - One mobile calling party and two called parties, one SIP, one mobile | Register two called parties, one SIP, one mobile, then launch an audio call from a mobile calling party. | SIP Client, Mobile Phone (x2) |
| Mobile call - One mobile calling party and two mobile called parties | Register two called parties, both mobile, then launch an audio call from a mobile phone. | Mobile Phone (x3) |
| Mobile call - One SIP calling party and two mobile called parties | Register two called parties, both mobile, then launch an audio call from a calling party to one of the called parties and measure round-trip times, jitter and packet loss of the RTP audio packets. Using call actions, the call can be transferred to the second called party, or a second call to the second called party can be made. | Mobile Phone (x2), SIP Client |
| Packet Capture | Auxiliary agent for capturing traffic | Packet Capture Client |
| Signaling and Media benchmark for Basic Calls | Register a list of called parties, then launch calls with optional media from a list of calling parties to these called parties in a loop. | SIP Load Client (x2) |
| Signaling and Media benchmark for Basic Calls - Called Party Only | Register a list of called parties, then wait for any incoming calls and answer them. | SIP Load Client |
| Signaling and Media benchmark for Basic Calls - Calling Party Only | Launch calls with optional media from a list of calling parties to a list of called parties in a loop. The called parties are NOT served by this scenario. This is useful to test voice applications like voicemail or IVR systems, which automatically answer the call. | SIP Load Client |
| Signaling and Media benchmark for Basic Calls - Calling Party Only - Customised | Launch calls with optional media from a list of calling parties to a list of called parties in a loop. The called parties are NOT served by this scenario. This is useful to test voice applications like voicemail or IVR systems, which automatically answer the call. | SIP Load Client |
| Signaling and Media benchmark for Basic Calls - Calling Party Only w/ Registration | First register the list of calling parties, then launch calls with optional media from a list of calling parties to a list of called parties in a loop. The called parties are NOT served by this scenario. This is useful to test voice applications like voicemail or IVR systems, which automatically answer the call. | SIP Load Client |
| Signaling and Media benchmark for Basic Calls - Calling Party Only w/ TTS | Launch calls with TTS media from a list of calling parties to a list of called parties in a loop. The called parties are NOT served by this scenario. This is useful to test voice applications like voicemail or IVR systems, which automatically answer the call. | SIP Load Client |
| Signaling and Media benchmark for Basic Calls w/ Calling Party Registration | Register a list of called parties, then register the list of calling parties and launch calls with optional media from these calling parties to the called parties in a loop. | SIP Load Client (x2) |
| Signaling and Media benchmark for Basic Calls w/o Registration | Launch calls with optional media from a list of calling parties to a list of called parties in a loop, without registering the called parties first. | SIP Load Client (x2) |
| Signaling benchmark for presence-event SUBSCRIBE + PUBLISH messages (Beta) | Callers subscribe to presence states of different callees and then these callees publish 4 different states (Online, Away, Busy, Offline). | SIP Load Client (x2) |
| Signaling benchmark for presence-event SUBSCRIBE + PUBLISH messages w/ Registrations on the called party | Register callees, callers subscribe to presence states of different callees and then these callees publish 4 different states (Online, Away, Busy, Offline). | SIP Load Client (x2) |
| Signaling benchmark for presence-event SUBSCRIBE w/ Registrations on the called party and web request | Register and subscribe callees and send a web request to modify the test behavior | SIP Load Client, Web Request |
| Signaling benchmark for PUBLISH messages (Beta) | Publish different presence states of a user. | SIP Load Client |
| Signaling benchmark for Registrations | Register a list of users in a loop. | SIP Load Client |
| Signaling benchmark for Rejected Calls - Called Party Only | Register a list of called parties, then wait for any incoming calls and reject them with a given code and reason. | SIP Load Client |
| Signaling benchmark for Subscriptions (Beta) | Subscribe to a list of users in a loop. | SIP Load Client |
| Speed Test client only | Perform a speed test of the available bandwidth | Bandwidth |
| Speed Test Client/Server | Perform a speed test of the available bandwidth by emulating both the client and the server side. | Bandwidth (x2) |
| Subscribe Loadtest with Calls (Beta) | Subscribe to a list of presence states and wait for a number of NOTIFY messages. In parallel register a list of called parties and additionally launch calls with optional media from a list of calling parties to these called parties in a loop. | SIP Load Client (x3) |
| TLS Scan | Perform TLS scan | TLS Client |
| Traceroute Test | Perform a traceroute to a destination to determine the hops and their latency. | Traceroute |
| Web Request | Perform single web request | Web Request |