Modify Kurento group call example to support only audio - webrtc

Question

I need to modify the Kurento group call example from Link
to send only audio if one participant has no camera. Right now only audio is received when a camera is used. When only a microphone is available I receive a DeviceMediaError.
I managed to filter whether a camera device is connected or not and then send only audio, but this doesn't work. Maybe the participant should've an audio tag instead of a video tag?
EDIT: It's only working on Firefox and not in Chrome. Any ideas?

Answer 1

in file - https://github.com/Kurento/kurento-tutorial-java/blob/master/kurento-group-call/src/main/java/org/kurento/tutorial/groupcall/UserSession.java.
change following line -
sender.getOutgoingWebRtcPeer().connect(incoming, MediaType.AUDIO);
and set offer media constraints to video:false in browser js file.
updated code -
let constraints = {
audio: true,
video: false
};
let localParticipant = new Participant(sessionId);
participants[sessionId] = localParticipant;
localVideo = document.getElementById('local_video');
let video = localVideo;
let options = {
localVideo: video,
mediaConstraints: constraints,
onicecandidate: localParticipant.onIceCandidate.bind(localParticipant),
configuration : { iceServers : [
{"url":"stun:74.125.200.127:19302"},
] }
};
localParticipant.rtcPeer = new kurentoUtils.WebRtcPeer.WebRtcPeerSendonly(options, function(error) {
if (error) {
return console.error(error);
}
localVideoCurrentId = sessionId;
localVideo = document.getElementById('local_video');
localVideo.src = localParticipant.rtcPeer.localVideo.src;
localVideo.muted = true;
this.generateOffer(localParticipant.offerToReceiveVideo.bind(localParticipant));
});
server.js code
function join(socket, room, callback) {
let userSession = userRegister.getById(socket.id);
userSession.setRoomName(room.name);
room.pipeline.create('WebRtcEndpoint', {mediaProfile : 'WEBM_AUDIO_ONLY'}, (error, outgoingMedia) => {
if (error) {
console.error('no participant in room');
if (Object.keys(room.participants).length === 0) {
room.pipeline.release();
}
return callback(error);
}
// else
outgoingMedia.setMaxAudioRecvBandwidth(100);
add media profile parameter on server side while joining room.
function getEndpointForUser(userSession, sender, callback) {
if (userSession.id === sender.id) {
return callback(null, userSession.outgoingMedia);
}
let incoming = userSession.incomingMedia[sender.id];
if (incoming == null) {
console.log(`user : ${userSession.id} create endpoint to receive video from : ${sender.id}`);
getRoom(userSession.roomName, (error, room) => {
if (error) {
return callback(error);
}
room.pipeline.create('WebRtcEndpoint', {mediaProfile : 'WEBM_AUDIO_ONLY'}, (error, incomingMedia) => {
if (error) {
if (Object.keys(room.participants).length === 0) {
room.pipeline.release();
}
return callback(error);
}
console.log(`user: ${userSession.id} successfully create pipeline`);
incomingMedia.setMaxAudioRecvBandwidth(0);
incomingMedia.getMaxAudioRecvBandwidth(0);
add media profile parameter when accepting call.
hope this helps.