1.0.0 • Published 5 years ago
webrtcsmetrics v1.0.0
WebRTCMetrics
WebRTCMetrics is a JavaScript library that aggregates stats received from the WebRTC stack to help Web applications grabbing metrics and information in real time.
Install
Using NPM
$ npm install webrtcmetricsUsing Yarn
$ yarn add webrtcmetricsUsage
Once initialized the RTCPeerConnection, give it to the WebRTCMetrics instance created such in the following example:
import WebRTCMetrics from "webrtcmetrics";
// Somewhere in your code
const pc = new RTCPeerConnection(config);
const analyzer = new WebRTCMetrics({ pc });
analyzer.onmetrics = (metrics) => {
// Do something with the metrics received
};
// Start the analyzer
analyzer.start();Audio Information
| Name | Value | Description |
|---|---|---|
| input_codec | JSON | Description of the audio input codec and parameters used |
| input_level | Number | Level of the input sound (speakers) |
| output_codec | JSON | Description of the audio output codec and parameters used |
| output_level | Number | Level of the output sound (microphone) |
Audio Statistics
| Name | Value | Description |
|---|---|---|
| last_three_jitter | Array | Last 3 Jitter values received (in ms) |
| last_three_rtt | Array | last 3 RTT values received (in ms) |
| percent_packets_lost | Number | Percent of audio packet lost since the last statistic |
| total_packets_received | Number | Number of packers received since the begining of the call |
| total_packets_lost | Number | Number of packers lost since the begining of the call |
| delta_packets_received | Number | Number of packers received since the last statistic |
| delta_packets_lost | Number | Number of packers lost since last statistic |
Audio metrics
| Name | Value | Description |
|---|---|---|
| mos | Number | Audio quality indicator based on 'Monitoring VoIP Call Quality Using Improved Simplified E-model' |
Video Statistics
| Name | Value | Description |
|---|---|---|
| input_codec | JSON | Description of the video input codec and parameters used |
| input_size | Number | Size of the input video (from remote peer) |
| output_codec | JSON | Description of the video output codec and parameters used |
| output_size | Number | Size of the output video (own video) |
Network
| Name | Value | Description |
|---|---|---|
| infrastructure | Number | Infrastructure level (0: Eth, 3: Wifi, 5: 4G, 10: 3G) |
Metrics to add
cet (call establishment time) --> Network ? local_candidate {type: "host|srflx|relay", protocol: "udp|tcp, port: Number} --> Network ? remote_candidate_type {type: "host|srflx|relay", protocol: "udp|tcp, port: Number} --> Network ? candidates_number: Number --> Network
1.0.0
5 years ago