WebRTC API tricks

Streams and peers association

When multiple participants join a room and publish their audio and video, a way to determine which peer each each stream belongs to is needed.

By using the API provided by the webrtc module, mediasoup stores each receiving msid value into the userParameters of the corresponding RtpReceiver. Such an object is copied verbatim into the corresponding RtpParameters of all the associated RtpSender instances.

The msid value has the following syntax:

xxxxxxxx yyyyyyyy

Where xxxxxxxx matches the id of the MediaStream source, and yyyyyyyy matches the id of the specific MediaStreamTrack.

For each RTCPeerConnection in a room, the Node.js application can retrieve all its RtpSender instances and the msid associated to them:

// Get the underlying Peer instance.
let peer = peerconnection.peer;

// Map of Peer names indexed by
let mediaStreamIdToPeerName = {};

for (let rtpSender of peer.rtpSenders) {
  // Get the name of the associated Peer instance.
  let associatedRtpReceiver = rtpSender.associatedRtpReceiver;
  let senderPeerName =;

  // Get the
  let msid = rtpSender.rtpParameters.userParameters.msid;
  let mediaStreamId = msid.split(/\s/)[0];

  // Store in the map.
  mediaStreamIdToPeerName[mediaStreamId] = senderPeerName;

The Node.js app can signal such a mediaStreamIdToPeerName map to the browser. The JavaScript in the browser application can then inspect the remote MediaStream objects in its local RTCPeerConnection and look for their id in the map, obtaining the associated peer name.

Limiting the sending bitrate

mediasoup implements REMB based congestion control. This feature lets the client adapt its sending bitrate to the available network bandwidth.

However, in many scenarios it may be useful for the Node.js app to artificially limit the sending bitrate of a specific participant (for example, when his video stream is being rendered into a very small <video> into the browser application).

The setMaxBitrate() method of the Transport class allows the Node.js application to dynamically change the maximum sending bitrate of a peer. Within the exposed RTCPeerConnection object, it can be used as follows:

// Get the transport of the RTCPeerConnection.
// NOTE: Just a single transport is created.
let transport = peerconnection.transports[0];

// Ensure it exists.
if (!transport)

// Set 64kbps as maximum sending bitrate.

The maximum sending bitrate can also be set within the RTCPeerConnection constructor by setting the maxBitrate option.