WebRTC is an open-source real-time interactive audio and video communication framework. This paper discusses some of the mechanisms utilized in WebRTC to handle packet losses in the video communication path. Various system details are discussed and an adaptive hybrid NACK/FEC method with temporal layers is presented. Results are shown to quantify how the method controls the quality trade-offs for real-time video communication.

AV1 compression technology achieved superb efficiency over HEVC and VP9 by introducing advanced compression tools. With these new tools comes increased encoding complexity. On the other hand, real-time communication (RTC) applications require an encoder to operate with low latency, low bitrate, low encode time and often on low-end platforms. This makes usage of AV1 encoder in RTC tasks challenging. In this paper we discuss some of the techniques we used to make libaom AV1 encoder suitable for RTC applications: from encoding tool selection to reducing complexity of the existing encoding tools. By implementing these algorithms among others libaom AV1 encoder is now able to provide lowlatency real-time video encoding even on low-end mobile platforms in RTC applications.