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callstats.js: Integrating with your WebRTC App



You can track your integration progress from your callstats dashboard at:

Step 1: Include callstats.js

Everything in the callstats.js is scoped under the callstats namespace, hence create the object.

Add the callstats.js in the HEAD tag.

<script src=""></script>

After adding it, you can begin using callstats

var callstats = new callstats();

If you are using require.js, please refer to the following callstats.js: require.js support section.

Step 2: Initialize() with AppSecret

After the user is authenticated with the origin server (or when the page loads), call initialize() with appropriate parameters see callstats.js API Reference. Check the callback for errors. If the authentication succeeds, callstats.js will receive a valid authentication token to make subsequent API calls.

//initialize the app with application tokens
  var AppID     = "YOUR APPLICATION ID";

  //localUserID is generated or given by the origin server
  callstats.initialize(AppID, AppSecret, localUserID, csInitCallback, csStatsCallback, configParams);

ALTERNATIVE: If you are interested in using the third-party authentication, see the details described in Other Authentication Mechanisms and API section.

For more information on callbacks, please refer to csInitCallback and csStatsCallback. Also have a look at step 8 for csStatsCallback data handling.

Step 3: addNewFabric()

When creating a PeerConnection, call addNewFabric() with appropriate parameters (see API section). It is important to make the request only after the PeerConnection is created. The PeerConnection object MUST NOT be "undefined" or NULL because callstats.js uses getStats() to query the metrics from the browser internals. The application SHOULD call addNewFabric() immediately after the PeerConnection object is created.

Time stamp of addNewFabric() is used as a reference point to calculate fabric failure delay or fabric setup delay:



  • Fabric failure delay = timestamp of fabricSetupFailed - timestamp of addNewFabric
  • Fabric setup delay = timestamp of fabricSetup - timestamp of addNewFabric

In any WebRTC endpoint, where multiple PeerConnections are created between each participant (e.g., audio and video sent over different PeerConnections or a mesh call), the addNewFabric() MUST be called for each PeerConnection.

//adding Fabrics
  var pc_config = {"iceServers": [{url: ""}]};
  var pcObject = new RTCPeerConnection(pc_config);

  function pcCallback (err, msg) {
    console.log("Monitoring status: "+ err + " msg: " + msg);

  // pcObject is created, tell callstats about it
  // pick a fabricUsage enumeration, if pc is sending both media and data: use multiplex.

  var usage = callstats.fabricUsage.multiplex;
  var fabricAttributes = {
    remoteEndpointType:   callstats.endpointType.peer,
    fabricTransmissionDirection:  callstats.transmissionDirection.sendrecv

  //remoteUserID is the recipient's userID
  //conferenceID is generated or provided by the origin server (webrtc service)
  callstats.addNewFabric(pcObject, remoteUserID, usage, conferenceID, fabricAttributes, pcCallback);

Step 4: reportError()

Sometimes WebRTC endpoints fail to establish connectivity, this may occur when user-agents and/or bridges implement differing flavors of the Session Description Protocol (SDP) or may not support some features that others implement.

The WebRTC APIs either have a callback or a Promise associated to them. Since callstats.js ver. 3.2.x, WebRTC applications can use reportError() to capture at which stage the negotiation fails and pass on the DomError returned by the callback or Promise to The failure reason will appear both in the conference time-line and aggregate on the main dashboard. See Section enumerating WebRTC functions for details. The example below reports error when creating an SDP offer:

//adding Fabrics
  var pc_config = {"iceServers": [{url: ""}]};
  var pcObject = new RTCPeerConnection(pc_config);

  function pcCallback (err, msg) {
    console.log("Monitoring status: "+ err + " msg: " + msg);

  function createOfferError(err) {
    callstats.reportError(pcObject, conferenceID, callstats.webRTCFunctions.createOffer, err);

  // remoteUserID is the recipient's userID
  // conferenceID is generated or provided by the origin server (webrtc service)
  // pcObject is created, tell callstats about it
  // pick a fabricUsage enumeration, if pc is sending both media and data: use multiplex.

  var usage = callstats.fabricUsage.multiplex;
  callstats.addNewFabric(pcObject, remoteUserID, usage, conferenceID, pcCallback);

  // let the "negotiationneeded" event trigger offer generation
  pcObject.onnegotiationneeded = function () {
    // create offer



You have now completed the basic integration steps, read more for advanced features!

Step 5: (OPTIONAL) sendFabricEvent()

During the conference, users might perform several actions impacting the measurements and conference analysis. The user might mute the audio or switch off the camera or do screen sharing during a conference. These events can directly impact the measurement data (For example, you can see a significant drop in throughput when camera is switched off). For the list of all possible conference events, please refer here

Send the appropriate fabricEvent via sendFabricEvent().

  • fabricSetup and fabricSetupFailed has been deprecated in v2.1.0 and v3.10.0,
    respectively, these events are now generated automatically by the JS library.

  • send fabricTerminated when an endpoint or participant disconnects from
    the conference, it notifies callstats.js to stop monitoring
    the local PeerConnection. Depending on the implementation of the
    hangup in your WebRTC application (may have to rely on signaling), the
    remote endpoint sends a fabricTerminated event before destroying its
    local PeerConnection object. monitors each
    PeerConnection in real-time, and generates summary statistics when the
    participant leaves. The summary of statistics for each conference is aggregated
    when all the participant have left. If no fabricTerminated event is received, will summarize
    and aggregate the summary statistics 30 seconds after the last measurement
    for a conference is received.

  • send fabricHold or fabricResume whenever the user holds and resumes the call.
    This is usually done when a user gets multiple incoming conference calls, and has
    to stop transmitting (hold) on one conference call to transmit on the other, and
    then returns to earlier call to resume transmitting (unhold).

  • send dominantSpeaker when a particular userID appears to be the only participant
    speaking. Typically, each endpoint calculates the dominant speaker over the set of
    participants in a sliding time-window (say, 10 seconds). Then the endpoint that
    notices that it is the dominant speaker sends the event.

  • send activeDeviceList whenever the audio input, audio output, or video input
    device changes.

// send fabricEvent: videoPause
callstats.sendFabricEvent(pcObject, callstats.fabricEvent.videoPause, conferenceID);

// devices are returned by the
var devices = navigator.mediaDevices.getUserMedia({
  audio: true,
  video: true
//a typical device looks like:
//device= {  "deviceId":"default","kind":"videoinput","label":"FaceTime HD Camera","groupId":"2004946474"}

var eventData = {
  deviceList: devices // array of active device

// send fabricEvent: activeDeviceList
callstats.sendFabricEvent(pcObject, callstats.fabricEvent.activeDeviceList, conferenceID, eventData);

Step 6: (OPTIONAL) associateMstWithUserID()

When interacting with the conference server, the developer is most likely going to use the name or identifier associated with the conference server as the remoteUserID. A typical conference bridge (for example, Jitsi Videobridge) transmits multiple media stream tracks within a peer connection. In which case, using a remote participant’s userID is impractical as there maybe several participants.

Since callstats.js ver. 3.3.x, we allow mapping Synchronization Source Identifier (SSRC) for a mediastreamtrack to a userID (both local and remote). By default the local and remote MediaStreamTracks are automatically mapped to the localUserID and remoteUserID. With associateMstWithUserID(), you can override the actual local and remote userIDs to the correct association. If the DOM identifiers of the video tags associated to each participant, callstats.js will calculate better quality scores for each participant. The code example shows how the API can be integrated:

Associationg userID with MSTAssociationg userID with MST More discussion related to the motivation of `associateMstWithUserID()` is covered in the following [blog post](/2015/07/17/api-update-handling-multiple-media-stream-tracks-callstats/).
// After O/A is complete, i.e., onAddStream is fired
  var localUserID  = "Alice";
  var remoteUserID = "Bob";
  var conferenceID = "AliceAndBobAndCharlie";
  var mstLabel = "front-camera";
  // SSRC1 is the SSRCs of the local video stream
  // SSRC2 is the SSRC of the remote video stream, usually received in the remote SDP
  // mstLabel is a developer provided string that lets them identify
  // various tracks (e.g., front-camera, back-camera, without looking at the
  // configurations of the individual MSTs). In this example, we assume it is the
  // front-camera.
  callstats.associateMstWithUserID(pc, localUserID, conferenceID, ssrc1, mstLabel);
  callstats.associateMstWithUserID(pc, remoteUserID, conferenceID, ssrc2, mstLabel);

Step 7: (OPTIONAL) sendUserFeedback()

The developers are expected to design an appropriate UI to get user input on quality at the end of the call. Typically, services collect user feedback based on the Mean Opinion Score (MOS). However, it is not neccessary to use all values of the MOS scale, for example a service using only 2 point scale: it can associate 1 and 5 to bad and excellent, respectively and not use the values 2 to 4.

var overallRating = 4; // 1-5 rating, typically according to MOS scale.
  var feedback = {
    "userID": localUserID, //mandatory
    "overall": overallRating, //mandatory
  callstats.sendUserFeedback(conferenceID, feedback, pcCallback);

Step 8: (OPTIONAL) Handling stats from csStatsCallback()

The developers can handle the stats received from csStatsCallback function in a way suitable to their application. It can be used for displaying bitrate or based on the conference quality indicators applications can change their settings etc. For more details check this blog post.

Step 9: (OPTIONAL) Submitting application logs

The developers can send application error logs using reportError() API and track them on dashboard. The logs will help in debugging the corresponding conferences. The error can be an object or a string.

var error1 = {
    message: "Error message",
    error: "Error 1",
    stack: "stack trace for the error"

callstats.reportError(pc, confID, callstats.webRTCFunctions.applicationLog, error1);


var error2 = "application error ";

callstats.reportError(pc, confID, callstats.webRTCFunctions.applicationLog, error2);


Error log size is capped

Please note that the application log size is limited to 20KB. Any application log greater than 20KB will be truncated to 20KB and a warning message will be displayed on the console log.

Step 10: (OPTIONAL) Obtaining the default configuration

You can obtain the default configuration you have set on the dashboard via default configuration callback. The default configuration can be set for the Peer Connection or the media. You can set the configuration by visiting the "App Settings" page and navigating to the "configuration" tab. The section RTCPeerConnection configuration and Media constraints can be used for entering the default configuration in JSON format.

// example: default configuration callback

function csDefaultConfigCallback(config) {
    var pc = new RTCPeerConnection(config.peerConnection);
// Setting the default configuration callback

callstats.on(“defaultConfig”, csDefaultConfigCallback);

Updated about a month ago

callstats.js: Integrating with your WebRTC App

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