The SIP feature measures response times in SIP signaling and the voice quality of RTP media streams.
SIP monitoring works in a hub-and-spoke topology with a passive hub and a number of client Test Agents. In each monitoring cycle, the clients will register and then set up a call towards the hub. The call lasts for a specified amount of time, after which the client terminates the call and unregisters. The next cycle then begins.
The SIP clients verify the availability and performance of SIP servers by measuring the completion time for various SIP operations. During the test cycles, each client measures completion times for SIP operations (register, invite, hang-up, and unregister). During VoIP calls, both the hub and the client measure the performance and quality of the VoIP session (rate, loss, misorders, jitter) as well as voice quality MOS scores.
The Test Agents support execution of multiple concurrent SIP tests on different interfaces (one SIP test/account per interface).
The following audio codecs are supported: GSM, G.711 A-law, and G.711 µ-law.
Setting up a SIP test or monitoring
To run SIP measurements, you need to have at least two Netrounds Test Agents installed. If you haven't already done the installation, consult this page: Getting started with IP telephony (SIP and VoIP) measurements.
Make sure that you have prepared the Netrounds account with SIP accounts. Read more on this topic on the page Setting up SIP accounts.
Create a new SIP test or monitoring and fill in the mandatory parameters as explained under Parameters below.
It is possible to run the SIP test cycles without the Test Agents ever registering. The VoIP calls will then be set up directly towards the IP address of the hub Test Agent. It is also possible to perform only an initial registration; the Test Agents will then register with the SIP server once at the beginning, but in each test cycle they will only make calls, without unregistering and re-registering.
The number of calls in each test cycle is configurable. The calls will be made sequentially.
- Duration (seconds): The duration of this test step in seconds. Min: 30 s. Max: 604800 s. Default: 60 s.
- Fail threshold (seconds): The maximum number of errored seconds (ES) that may occur without triggering a fail for this test step. Default: 0.
- Wait for ready: Time to wait before starting the test. The purpose of inserting a wait is to allow all Test Agents time to come online and acquire good time sync. Min: 1 min. Max: 24 hours. Default: "Don't wait", i.e. zero wait time.
- Hub: The Test Agent that acts as hub in the test or monitoring. The client Test Agents will make calls towards this hub.
- SIP account: After selecting the hub, you are prompted to select what SIP account to associate with the hub. You can choose among the SIP accounts that are available under Account > SIP accounts.
- Clients: The Test Agents that will make calls towards the hub.
- SIP accounts: After selecting a client, you are prompted to select what SIP account to associate with that client. You can choose among the SIP accounts that are available under Account > SIP accounts.
- Registration during test cycles: This setting determines whether the Test Agent should do SIP registration during testing.
- Yes: The Test Agent will unregister and re-register in each test cycle.
- Only once at the beginning: The Test Agent will register only once at the beginning of the test.
- No: The Test Agent will never register.
- Number of calls per test cycle: The number of SIP calls to make during a test cycle. Default: 1.
- Time to keep a call/registration: The duration of calls and registrations in seconds. Min: 1 s. Default: 10 s.
Thresholds for errored seconds (ES)
- SIP response time (ms): Maximum allowed completion time for SIP operations (registration, unregistration, invite, and hang-up). If any of these operations takes longer to complete, an errored second is triggered. Min: 1 ms. Default: 400 ms.
- MOS: Lowest allowed Mean Opinion Score (MOS) during VoIP calls. If during a call the MOS value drops below this level, an errored second is triggered. Range: 1 ... 5. Default: 4.
- DSCP/IPP: The Differentiated Services Code Point (DSCP) or IP Precedence to be used in the IP packet headers, for SIP signaling as well as media streams. Range: 0 ... 63. Default: 0.
- Transport: Transport protocol to be used for SIP messages: UDP or TCP. The default is UDP.
- Codec: Audio codec to be used for RTP media streams: GSM, G.711 A-law (PCMA), or G.711 µ-law (PCMU). Default: GSM.
- Delayed start (s): (Tests only) Time by which to delay the start of the test within a test step. Default: 0 s.
SLA thresholds (monitorings only)
- SLA Good: Threshold for good fulfillment of service level agreement. Default: 99.95%.
- SLA Acceptable: Threshold for acceptable fulfillment of service level agreement. Default: 99.5%.
- Rate, loss, misorders, jitter, MOS, ES total, ES response time, ES MOS, SLA