Using DTMF with WebRTC - Web APIs
falling " + "back to the <strong>deprecated</strong> addstream() method..."); callerpc.addstream(stream); } if (callerpc.getsenders) { dtmfsender = callerpc.getsenders()[0].dtmf; } else { log("your browser doesn't support rtcpeerconnection.getsenders(), so " + "falling back to use <strong>deprecated</strong> createdtmfsender() " + "instead."); dtmfsender = callerpc.createdtmfsender(audiotracks[0]); } dtmfsender.ontonechange = handletonechangeevent; } after setting audiotracks to be a list of the audio tracks on the stream from the user's microphone, it's time to add the media to the caller's rtcpeerconnection.
... if getsenders() isn't available, we instead call rtcpeerconnection.createdtmfsender() to get the rtcdtmfsender object.

RTCPeerConnection - Web APIs
rtcpeerconnection.createdtmfsender() creates a new rtcdtmfsender, associated to a specific mediastreamtrack, that will be able to send dtmf phone signaling over the connection.