ringcentral-softphone - npm Package Compare versions

Comparing version 0.1.7 to 0.1.8

demos/browser.js

@@ -20,6 +20,5 @@ import RingCentral from '@ringcentral/sdk'
   await rc.logout() // rc is no longer needed
-
+  const inputAudioStream = await navigator.mediaDevices.getUserMedia({ audio: true, video: false })
   const audioElement = document.getElementById('audio')
-  softphone.on('INVITE', sipMessage => {
-    softphone.answer()
+  softphone.on('INVITE', async sipMessage => {
     softphone.on('track', e => {
@@ -29,2 +28,3 @@ audioElement.srcObject = e.streams[0]
     })
+    await softphone.answer(inputAudioStream)
   })
@@ -31,0 +31,0 @@ softphone.on('BYE', () => {

demos/node.js

 import RingCentral from '@ringcentral/sdk'
 import fs from 'fs'
-import { nonstandard } from 'wrtc'
+import { nonstandard, MediaStream } from 'wrtc'
+import RTCAudioSource from 'node-webrtc-audio-source'
 
@@ -9,2 +10,4 @@ import Softphone from '../src/index'
 
+const rtcAudioSource = new RTCAudioSource()
+
 const rc = new RingCentral({
@@ -32,4 +35,3 @@ server: process.env.RINGCENTRAL_SERVER_URL,
   }
-  softphone.on('INVITE', sipMessage => {
-    softphone.answer()
+  softphone.on('INVITE', async sipMessage => {
     softphone.on('track', e => {
@@ -42,4 +44,10 @@ audioSink = new RTCAudioSink(e.track)
     })
+    const stream = new MediaStream()
+    const track = rtcAudioSource.createTrack()
+    stream.addTrack(track)
+    await softphone.answer(stream)
+    rtcAudioSource.start()
   })
   softphone.on('BYE', () => {
+    rtcAudioSource.stop()
     audioSink.stop()
@@ -46,0 +54,0 @@ audioStream.end()

dist/index.js

@@ -152,3 +152,3 @@ "use strict";
 
-  async answer() {
+  async answer(inputAudioStream = undefined) {
     const sdp = this.inviteSipMessage.body;
@@ -159,3 +159,3 @@ const remoteRtcSd = new _isomorphicWebrtc.RTCSessionDescription({
     });
-    const rtcpc = new _isomorphicWebrtc.RTCPeerConnection({
+    const pc1 = new _isomorphicWebrtc.RTCPeerConnection({
       iceServers: [{
@@ -165,8 +165,14 @@ urls: 'stun:74.125.194.127:19302'
     });
-    rtcpc.addEventListener('track', e => {
+    pc1.addEventListener('track', e => {
       this.emit('track', e);
     });
-    rtcpc.setRemoteDescription(remoteRtcSd);
-    const localRtcSd = await rtcpc.createAnswer();
-    rtcpc.setLocalDescription(localRtcSd);
+    pc1.setRemoteDescription(remoteRtcSd);
+
+    if (inputAudioStream) {
+      const track = inputAudioStream.getAudioTracks()[0];
+      pc1.addTrack(track, inputAudioStream);
+    }
+
+    const localRtcSd = await pc1.createAnswer();
+    pc1.setLocalDescription(localRtcSd);
     await this.send(new _ResponseSipMessage.default(this.inviteSipMessage, 200, 'OK', {
@@ -173,0 +179,0 @@ Contact: `<sip:${this.fakeEmail};transport=ws>`,

package.json

 {
   "name": "ringcentral-softphone",
-  "version": "0.1.7",
+  "version": "0.1.8",
   "license": "MIT",
@@ -32,5 +32,8 @@ "main": "dist/index.js",
     "jest": "^24.9.0",
+    "mic": "^2.1.2",
+    "node-webrtc-audio-source": "^0.1.1",
     "ramda": "^0.26.1",
     "standard": "^14.2.0",
     "webpack": "^4.40.2",
+    "webpack-cli": "^3.3.9",
     "webpack-dev-server": "^3.8.1",
@@ -37,0 +40,0 @@ "wrtc": "^0.4.2",

README.md

 # RingCentral Softphone SDK for JavaScript
 
-## Experimental
-
-This project is still in its early stage. It is not ready for production yet!
-
-
 ## What are the differences between ringcentral-web-phone and this project?
@@ -41,8 +36,11 @@
 await rc.logout()
+
+// const stream = <input-audio-stream-from-microphone>
+
 softphone.on('INVITE', sipMessage => {
-  softphone.answer()
+  softphone.answer(stream)
   softphone.on('track', e => {
     // phone call connected
     const { track, streams } = e
-    // you can audio data from track and streams
+    // you can get audio data from track and streams
   })
@@ -129,3 +127,2 @@ })
 
-- state machine
 - make outbound call

src/index.js

@@ -110,12 +110,16 @@ import uuid from 'uuid/v4'
 
-  async answer () {
+  async answer (inputAudioStream = undefined) {
     const sdp = this.inviteSipMessage.body
     const remoteRtcSd = new RTCSessionDescription({ type: 'offer', sdp })
-    const rtcpc = new RTCPeerConnection({ iceServers: [{ urls: 'stun:74.125.194.127:19302' }] })
-    rtcpc.addEventListener('track', e => {
+    const pc1 = new RTCPeerConnection({ iceServers: [{ urls: 'stun:74.125.194.127:19302' }] })
+    pc1.addEventListener('track', e => {
       this.emit('track', e)
     })
-    rtcpc.setRemoteDescription(remoteRtcSd)
-    const localRtcSd = await rtcpc.createAnswer()
-    rtcpc.setLocalDescription(localRtcSd)
+    pc1.setRemoteDescription(remoteRtcSd)
+    if (inputAudioStream) {
+      const track = inputAudioStream.getAudioTracks()[0]
+      pc1.addTrack(track, inputAudioStream)
+    }
+    const localRtcSd = await pc1.createAnswer()
+    pc1.setLocalDescription(localRtcSd)
     await this.send(new ResponseSipMessage(this.inviteSipMessage, 200, 'OK', {
@@ -122,0 +126,0 @@ Contact: `<sip:${this.fakeEmail};transport=ws>`,

No alert changes

Improved metrics

Total package byte prevSize

70119

increased by1.68%

Lines of code

979

increased by1.66%

Worsened metrics

Dev dependency count

20

increased by17.65%

Number of lines in readme file

127

decreased by-2.31%

No dependency changes