Using SIP INVITE to Specify Test Call Duration

For test call endpoints where the remote SIP UA is the initiator of the test call (i.e., 'Call Party' parameter configured to Called), the duration of the test call can be determined by the incoming SIP INVITE message, instead of when the caller ends the call (i.e., sends a SIP BYE message).

The duration in the SIP INVITE message is specified using the 'duration=' parameter of the Request-URI, for example:

INVITE sip:3000@10.8.51.29;user=phone;duration=10000 SIP/2.0

The duration is in milliseconds, but the device rounds it off to the nearest seconds.

This feature is in accordance with RFC 4240 (Basic Network Services with SIP).

This feature is also used by the device for reporting MOS to WebRTC clients, as described in Reporting MOS Triggered by WebRTC Client.