Key Metrics in chrome://webrtc-internals

The chrome://webrtc-internals tool is an essential diagnostic page in Google Chrome used by developers to troubleshoot real-time communication issues. This article provides a direct overview of the specific diagnostic metrics you can extract from this tool, focusing on connection quality, audio/video performance, and bandwidth utilization.

Connection and ICE Metrics

These metrics help diagnose signaling, connection establishment, and network routing issues. They are primarily found under the RTCIceCandidatePair and RTCPeerConnection statistics.

• currentRoundTripTime: Displays the latency of the network connection in seconds. High values indicate lag.
• localCandidateType and remoteCandidateType: Identifies how the peers are connected (e.g., host for local network, srflx for NAT traversal, or relay when using a TURN server).
• bytesSent and bytesReceived: Total data volume transferred. Sudden plateaus in these graphs suggest a frozen connection.
• requestsSent and responsesReceived: Measures STUN/TURN binding requests, indicating whether firewall issues are blocking ICE candidates.

Media Quality Metrics (Inbound and Outbound RTP)

These statistics assess the quality of the media streams (audio and video) being sent and received.

• packetsLost: The total number of RTP packets lost during transmission. High packet loss directly causes audio clipping and video artifacts.
• jitter: Measures the packet arrival delay variation in seconds. High jitter indicates network congestion, leading to choppy media playback.
• framesDecoded vs. framesDropped: For inbound video, a high ratio of dropped frames to decoded frames points to rendering bottlenecks or CPU exhaustion on the receiver’s device.
• framesEncoded: For outbound video, this tracks how many frames the sender’s encoder is producing.
• frameWidth and frameHeight: Tracks the active resolution of the video stream. Frequent drops in resolution indicate that the system is adapting to poor network conditions.

Bandwidth and Congestion Control

These metrics show how WebRTC adapts to changing network environments.

• availableOutgoingBitrate: The estimated uplink bandwidth available for sending media, determined by the Bandwidth Estimation (BWE) algorithm.
• targetBitrate: The bitrate the encoder aims to output based on the estimated available bandwidth.
• qualityLimitationReason: Indicates what is currently limiting the outbound video quality. Values include none, cpu (processor bottleneck), bandwidth (network capacity reached), or other.