At the moment when a call is initiated from Restcomm Web SDK, the call is established (200 Ok) once the call hits/reaches Restcomm, so the duration counter starts from the moment ring back tone starts playing which is totally fine for peer2peer implementations, however for real-world production use case with PSTN termination involved this will cause in-accurate billing/disputes with carriers when terminating the calls to the PSTN.
Suggested solution: Call established (200OK) should start once the call is connected to the B number, it should play RBT (Ring Back Tone) in early media mode, call state should be incomplete with SIP 100 if the B party did not pickup.
At the moment when a call is initiated from Restcomm Web SDK, the call is established (200 Ok) once the call hits/reaches Restcomm, so the duration counter starts from the moment ring back tone starts playing which is totally fine for peer2peer implementations, however for real-world production use case with PSTN termination involved this will cause in-accurate billing/disputes with carriers when terminating the calls to the PSTN.
Suggested solution: Call established (200OK) should start once the call is connected to the B number, it should play RBT (Ring Back Tone) in early media mode, call state should be incomplete with SIP 100 if the B party did not pickup.