I am currently working on a performance monitoring dashboard for our regional SIP trunks. I want to track the ‘Jitter’ and ‘Packet Loss’ metrics for our BYOC Cloud trunks in real time to proactively identify voice quality issues before the agents start complaining. I see that these metrics are available in the ‘Interaction Detail’ view, but I want to pull them via the API for all active calls. Is there an Analytics or Telephony API endpoint that provides real-time QoS metrics for active SIP legs?
Greetings. I am a security officer and I have been reviewing our SIP infrastructure for our SOC2 audit. To answer your question, Genesys Cloud does not expose real-time QoS metrics (like jitter and packet loss) via a polling API endpoint for active interactions. These metrics are calculated at the end of each media segment and are only available in the ‘Interaction Detail’ records after the call has finished. If you need real-time monitoring, you must use a third-party SIP probe or an SBC that can stream RTCP-XR reports to your dashboard.
Hello everyone! I am a supervisor for thirty agents and I would love to have a dashboard like this! My agents are always telling me about ‘choppy audio’ but I never have any data to show the IT team. Che75, if you cannot get the data from Genesys Cloud, maybe you can at least show the number of ‘Dropped Calls’ per hour? That would still be very helpful for me to see if there is a pattern during our peak times!