Real Time Communications Bits

Bandwidth estimation is probably the most critical component in the video engine of WebRTC. The bandwidth estimation (BWE) module is responsible for deciding how much video* traffic you can send without congesting the network to prevent degradation of the video quality. In the past bandwidth estimation algorithms used to be very rudimentary and based mostly on packet loss. Basically we used to start increasing slowly the video bitrate until we detected packets being lost.  To detect the packet loss you use the standard RTCP feedback mechanisms where the receiver side reports packet loss periodically using RTCP Receiver Report (RR) messages. Modern bandwidth estimation algorithms are more advanced and try to detect congestion before it is bad enough to make routers discard packets. These algorithms predicts congestion analyzing the delay between packets. The idea is that when you start having some congestion, the buffers in the routers will start filling and the delay will be

It is a common request from WebRTC developers and customers to know how they can differentiate WebRTC traffic from other type in their networks.  Usually the goal is to be able to prioritize RTC traffic over other types of less important traffic. By prioritizing the WebRTC traffic in the edge routers it is possible to improve the quality in some network scenarios .  The most common cases where this marking may help are: Congested broadband uplink where the router can discard other type of traffic instead of WebRTC traffic when queues get full. Congested local wireless network One obvious way to do this is forcing all the traffic to be relayed through a TURN or SFU server and se the priority based on IP addresses.   That's easy and always work, but the problem is that it is very difficult to maintain when your infrastructure changes often (scaling elastically, new datacenters...). Other way to accomplish this is to use  Differentiated Services Code Point (DSCP) marking. Wi