IOS Swift WebRtc insertDtmf issue - swift

Question

I am building an app that works with janus gateway via websocket and webrtc. everything works fine, I can send and receive voice calls successfully but insertDtmf metod doesnt send my dtmf to other peer.
Same account and same codes in android works fine.
Here is where I prepare webrtc
private func prepareWebRtc( callbacks:PluginHandleWebRTCCallbacksDelegate) {
if (pc != nil) {
if (callbacks.getJsep() == nil) {
createSdpInternal(callbacks: callbacks, isOffer: isOffer)
} else {
let jsep = callbacks.getJsep()!
let sdpString:String = jsep["sdp"] as! String
let type:RTCSdpType = RTCSessionDescription.type(for: jsep["type"] as! String)
let sdp:RTCSessionDescription = RTCSessionDescription.init(type: type, sdp: sdpString)
pc.setRemoteDescription(sdp) { (err) in}
}
} else {
trickle = callbacks.getTrickle() != nil ? callbacks.getTrickle()! : false
streamsDone(webRTCCallbacks: callbacks)
}
}
private func streamsDone(webRTCCallbacks:PluginHandleWebRTCCallbacksDelegate) {
let rtcConfig = RTCConfiguration.init()
rtcConfig.iceServers = server.iceServers
rtcConfig.bundlePolicy = RTCBundlePolicy.maxBundle
rtcConfig.rtcpMuxPolicy = RTCRtcpMuxPolicy.require
rtcConfig.continualGatheringPolicy = RTCContinualGatheringPolicy.gatherContinually
rtcConfig.sdpSemantics = .planB
let source :RTCAudioSource = sessionFactory.audioSource(with: audioConstraints)
let audioTrack:RTCAudioTrack? = sessionFactory.audioTrack(with: source, trackId: AUDIO_TRACK_ID)
let stream:RTCMediaStream? = sessionFactory.mediaStream(withStreamId: LOCAL_MEDIA_ID)
if (audioTrack != nil){
stream!.addAudioTrack(audioTrack!)
myStream = stream
}
if (stream != nil){
onLocalStream(stream: stream!)
}
// pc.addTrack(audioTrack, mediaStreamLabels);
pc = sessionFactory.peerConnection(with: rtcConfig, constraints: audioConstraints, delegate: nil)
if (myStream != nil){
pc.add(myStream)
}
if let obj:[String:Any] = webRTCCallbacks.getJsep(){
let sdp:String = obj["sdp"] as! String
let type:RTCSdpType = RTCSessionDescription.type(for: obj["type"] as! String)
let sessionDescription:RTCSessionDescription = RTCSessionDescription(type: type, sdp: sdp)
print(" STREAMS DONE JSEP NULL DEĞİL")
// pc.setRemoteDescription(WebRtcObserver(webRTCCallbacks), sessionDescription);
pc.setRemoteDescription(sessionDescription) { (err) in
}
}else{
createSdpInternal(callbacks: webRTCCallbacks, isOffer: isOffer)
print(" STREAMS DONE JSEP NULL ");
}
/* } catch (Exception ex) {
webRTCCallbacks.onCallbackError(ex.getMessage());
}*/
}
and here where I try to send dtmf
public func insertDTMF(_ tone:String){
if(pc != nil){
if let dtmfSender = pc.senders.first?.dtmfSender{
dtmfSender.insertDtmf(tone, duration: 200, interToneGap: 70)
}
//Here the timers are in ms
}
}

Answer 1

In my case, this is how I have handled insert DTMF functionality.
a - First filter out audio RTCRtpSender track:
var audioSender: RTCRtpSender?
for rtpSender in pc.senders {
if rtpSender.track?.kind == "audio" {
audioSender = rtpSender
}
}
b - And then use the same filtered audioSender object to insert the tone using OperationQueue
if let audioSender = audioSender {
let queue = OperationQueue()
queue.addOperation({
audioSender.dtmfSender?.insertDtmf(dtmfTone, duration: TimeInterval(0.1),interToneGap: TimeInterval(0.5))
})
}
Note: you can modify duration and interToneGap as per your requirement.
Hope this solution works for you as well.
The original answer can be found here: https://stackoverflow.com/a/60148372/4515269