mediasoup/node/lib/rtpStreamStatsTypes.d.ts

Cutting Edge WebRTC Video Conferencing

export type RtpStreamRecvStats = BaseRtpStreamStats & {
    type: string;
    packetCount: number;
    byteCount: number;
    bitrate: number;
    bitrateByLayer: BitrateByLayer;
};
export type RtpStreamSendStats = BaseRtpStreamStats & {
    type: string;
    packetCount: number;
    byteCount: number;
    bitrate: number;
};
export type BaseRtpStreamStats = {
    timestamp: number;
    ssrc: number;
    rtxSsrc?: number;
    rid?: string;
    kind: string;
    mimeType: string;
    packetsLost: number;
    fractionLost: number;
    jitter: number;
    packetsDiscarded: number;
    packetsRetransmitted: number;
    packetsRepaired: number;
    nackCount: number;
    nackPacketCount: number;
    pliCount: number;
    firCount: number;
    roundTripTime?: number;
    rtxPacketsDiscarded?: number;
    score: number;
};
export type BitrateByLayer = {
    [key: string]: number;
};
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