Play dtmf sounds to user

[rhonda]

Hello.

I am trying to figure out how to have a sound played back to the user when they enter a dtmf, so there is audible feedback with each tone sent. Is this something that JsSIP already provides, or would I need to implement playing a sound on the local audio stream with each dtmf send? 

Here is what my call flow looks like from the RTC Session point of view:

07:56:27.519 common.js:113 JsSIP:UA call() +4s
07:56:27.520 common.js:113 JsSIP:RTCSession new +0ms
07:56:27.520 common.js:113 JsSIP:RTCSession connect() +0ms
07:56:27.524 common.js:113 JsSIP:RTCSession emit "peerconnection" +4ms
07:56:27.524 common.js:113 JsSIP:RTCSession newRTCSession() +0ms
07:56:27.590 common.js:113 JsSIP:RTCSession unhold() +66ms
07:56:27.591 common.js:113 JsSIP:RTCSession unmute() +1ms
07:56:27.635 common.js:113 JsSIP:RTCSession session connecting +44ms
07:56:27.635 common.js:113 JsSIP:RTCSession emit "connecting" +0ms
07:56:37.988 common.js:113 JsSIP:RTCSession createLocalDescription()
07:56:38.098 common.js:113 JsSIP:RTCSession emit "sdp"
07:56:38.098 common.js:113 JsSIP:RTCSession emit "sending"  
07:56:38.207 common.js:113 JsSIP:RTCSession receiveInviteResponse()
07:56:38.208 common.js:113 JsSIP:RTCSession receiveInviteResponse() +1ms
07:56:38.208 common.js:113 JsSIP:RTCSession session progress +0ms
07:56:38.208 common.js:113 JsSIP:RTCSession emit "progress" +0ms
07:56:38.221 common.js:113 JsSIP:RTCSession receiveInviteResponse() +13ms
07:56:38.222 common.js:113 JsSIP:Dialog new UAC dialog created with status CONFIRMED +0ms
07:56:38.222 common.js:113 JsSIP:RTCSession emit "sdp" +1ms
07:56:38.224 common.js:113 JsSIP:RTCSession session accepted +2ms
07:56:38.224 common.js:113 JsSIP:RTCSession emit "accepted" +0ms
07:56:38.242 common.js:113 JsSIP:RTCSession sendRequest() +18ms
07:56:38.243 common.js:113 JsSIP:RTCSession session confirmed +1ms
07:56:38.243 common.js:113 JsSIP:RTCSession emit "confirmed" +0ms
07:56:45.038 common.js:113 JsSIP:RTCSession sendDTMF() | tones: 1 +7s

When initiating the call, media constraints are set as such:

phone.call(phoneNumber, {

pcConfig: {

rtcpMuxPolicy: 'negotiate',

},

mediaConstraints: {

audio: true,

video: false,

},

rtcOfferConstraints: {

offerToReceiveAudio: 1,

offerToReceiveVideo: 0,

},

Playing of the remote stream upon receipt of Accepted event:

sipSession.on(RTCSessionEvents.ACCEPTED, () => {

console.log('ACCEPTED', callId);

if (!remoteAudio.current || !remoteStream.current) {

remoteAudio.current = document.createElement('audio');

remoteStream.current = new MediaStream([

sipSession.connection.getReceivers()[0].track,

]);

remoteAudio.current.srcObject = remoteStream.current;

remoteAudio.current

.play()

.catch(() => console.log('ACCEPTED: error playing remote audio'));

}

[Igor K]

You can play DTMF sound using AudioContext

https://developer.mozilla.org/en-US/docs/Web/API/AudioContext