Showing posts from December, 2018

Sending Packet Loss Feedback in WebRTC SFUs

One of the responsibilities of WebRTC SFUs is to receive and send RTCP packets.  RTCP packets include different types of feedback about audio and video streams and one of the most important RTCP packets is the Receiver Report (RR) . RR packets are sent from the receiver of the media stream towards the sender of that media stream.  In case of an SFU the RR are generated and sent from the SFU to the media stream Sender and also from every stream Receiver to the SFU (Figure 1). The feedback sent inside RR packets include fields to calculate the round-trip-time delay, the jitter and the packet loss introduced by the network. The packet loss reported in these RR packets is important because the audio and video being sent will be adjusted based on that parameter: In case of audio streams the packet loss in the network modifies the level of robustness of the OPUS codec.   In presence of high packet loss the sender increases the level of redundancy of the forward error correction