I'm writing simple SIP-proxy application which stands between Astreisk and SIP client (any softphone). The purpose of the application is to calculate the duration of the call.
Below is example of regular flow:
On step 2 I assumes that call is started (e.g. rtp media flow is started). Then I wait for BYE message to calculate duration of the call. However I noticed that some clients never goes to step 3 and 4. No call end notification received from any parties after step 2. And duration of such call is infinitely.
What is the best way to find out start/stop time of the SIP call without sniffing RTP flow ? Should I wait for step 3 to mark the start of the call ? What if client omit ACK or what if UDP datagram with ACK is missed in the network ?
For now I used to think there is no reliable way to figure out that SIP call is started. Maybe I should use Astrisk channels API instead to track active calls.
Another option is to generate a RE-INVITE to test the existence of the Session. Dont have to negotiate a timer or anything. Just using re-invite could help. Reuse the SDP to ensure that no media changes happen. But then your application is moving out of being a Proxy in the path of the call to being a application server.
Also the duration can only be capped to nearest interval for this re-invite request and not necessarily the exact time the call was released.