We are running Asterisk mainly to proivde SIP services in a setting where many users have unpredictable bandwidth and where we have limited control over the client setup to say the least.
In other words, we frequently get reports with the typical level of detail like “yesterday I tried to call, but it did not work”. Quite often we find that for example users have not put in the STUN server and have only one way audio for that reason etc.
In order to take a more proactive approach to this, we would like to be able to log more technical details of a call. There is the CDR logging in Asterisk, but for our purpose, this is way too abstract.
What I would like to be able to see would be:
- SIP / SDP parameters. I want to see the IP addresses involved in the call, what codec got used, etc.
- actual throughput. Just counting TCP respectively UDP packets. This should give an indication of potential audio issues without actually tapping the call.
I know there is SIP debugging and I might be able to extract a lot of the Information I am looking for from SIP logs, but that would at least require some parsing and I’d assume this has been implemented already.
Also looking at the SIP and SDP level would still not tell me if data (audio) packages were exchanged or not.
Any pointers anyone?