As we head into the autumn / fall season, we’re keen not to lose that CommCon magic that we felt in the summer, so we’re revisiting yet another of our awesome talks from this year’s first ever US event in San Francisco.
SipFront were our Gold sponsor for this year’s event, and we have valued their input on several of our events in the past; this year was no different. What they brought to the table was a really important discussion around how to automate WebRTC calls and be able to continuously monitor the audio quality on those calls.
Testing the quality of calls is not simple. The typical and most simple approach is to analyse bitrate, latency and packet loss. While these are useful in determining the quality of the connection, they still don’t give you the full picture of the quality perceived by the users. In this talk, SipFront demonstrate how they solve this problem by running automated tests.
SipFront talked about how they test WebRTC calls but they're more well known for testing typical SIP calls - it's all Real Time Media, just with a different transport mechanism after all.
We're keen to hear your thoughts on the future of Media Delivery, WebRTC and AI and what it can do to help boost our industry and our work. Come share your thoughts with us on our social media.
We're only 21 subscribers away from 1000 over on our Youtube channel, so if you haven't already... please go subscribe do we can hit that 1000 number!
- Team CommCon