The congestion control problem from an RTC perspective, explaining the importance of a congestion control designed specifically for RTC, why it has to be delay-sensing to be able to keep latency low,? and why we can't use what's in?, e.g., TCP. It will touch on how a poorly designed system can have severe effects on audio quality, and how competing for bandwidth with a TCP flow can cause audio issues no matter what you do. I will explain why RTC calls over WiFi and mobile networks are hard to get right, and what we have done to better handle those.
The presentation will include graphs on how Chrome's congestion control performs, and examples of improvements we've made, for instance, to make it possible to reach HD quality within a second from the call starts, improvements for RTC calls over WiFi. It will end covering examples of what we're working on at the moment and what the next steps will be in this area, including optimizing for improved audio quality and better handling of mobile networks.
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