interceptor
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pion
Pluggable RTP/RTCP processors for building real time communication
According [to the specs](https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter) the jitter should be measured in seconds. The current stats interceptor instead [uses timestamp units](https://github.com/pion/interceptor/blob/d6aad58f3f3947d1b9411ad242be8c74e8472152/pkg/stats/stats_recorder.go#L140).
The issue covers everything we want to do around Congestion Estimation and Pion. This covers a Bandwidth Estimator in Pion, and generating feedback for remote peers. @mengelbart is the owner...
When the controller decides to increase the rate, it should not decrease, even if the measured transmission rate was low. The rate could be lower because there was not enough...
In pkg/gcc/arrival_group_accumulator.go, interDepartureThreshold is 5 * time.Millisecond, whereas google C++ implementation uses 0 here. https://webrtc.googlesource.com/src//+/d090952628606228afa47c842812927f15227290/modules/congestion_controller/goog_cc/inter_arrival_delta.cc?autodive=#125 If we believe performance is an issue, we should use a very small value, like...
This interceptor adds the playout delay header extension following https://webrtc.googlesource.com/src/+/refs/heads/main/docs/native-code/rtp-hdrext/playout-delay
This is based on #117 and adds a simple adapter to make the NADA implementation available to the generic CC interceptor.
This change exposes an API for reading the loss rate.
This change adds the NADA congestion control implementation. The binding to cc is left TODO.
### Your environment. - Version: v0.1.11 - Browser: Firefox 108 ### What did you do? Sending a single audio track from Firefox to pion webrtc, acting as SFU. It's also...
