WebRTC Metrics

A comprehensive overview of WebRTC statistics, derived indicators, and observable signals, to better understand call quality, connectivity, and user experience in rtcStats

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inbound-rtpinboundaudiovideo

packetsLost(%)

Percentage of RTP packets lost for this SSRC.

Description

Real number; as percentage

The percentage of RTP packets lost per second for this SSRC.

For this calculation, we divide packetsLost/s by inbound-rtp.packetsReceived/s.

Interpreting Values

Range	Description
0%	Perfect. No packets lost
<0.5%	Excellent. Losses hidden by FEC/concealment
0.5-1%	Acceptable for voice with FEC enabled
1-2%	Noticeable degradation
2-5%	Poor quality
>5%	Very poor

Video is more sensitive to loss than audio
Audio usually has better concealment mechanisms and Opus may have FEC enabled

Common Causes

Network congestion
WiFi instability
Cellular network handoffs
ISP issues
Overloaded local network

User Experience Impact

Audio: robotic voice, gaps, missing syllables
Video: blocky artifacts, frozen frames, resolution drops
Higher loss percentage always means worse quality. The relationship is roughly linear for audio but can be more abrupt for video (a single lost keyframe can freeze the stream)

Troubleshooting

Check if loss is bursty (causes freezes) or uniform (causes general degradation)
Enable FEC if not already enabled
Check if a wired connection helps - this isolates WiFi as the cause