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Adding a SIP Gateway Functionality #4

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29d9c7f
Add SipUserAgentPlugin and mjsip-ua dependency
kinsleykajiva Mar 7, 2024
7017c05
Refactor WebsocketEndpoint and extract WebRTCGStreamerPlugInEventsHan…
kinsleykajiva Mar 8, 2024
8222c53
Add event handling features to WebRTC plugin
kinsleykajiva Mar 8, 2024
90ffc48
Refactor code for readability and consistency
kinsleykajiva Mar 8, 2024
f9ac9b4
Refractor and simplify the logic in various methods
kinsleykajiva Mar 9, 2024
15a649d
Add debug messages for registration and call events
kinsleykajiva Mar 11, 2024
a55160a
Add SIP User Agent and enhance plugin handling
kinsleykajiva Mar 11, 2024
86e11aa
Add SIP Gateway example and clean up Java code
kinsleykajiva Mar 12, 2024
c99ed71
Handle SIP_GATEWAY feature in WebsocketEndpoint
kinsleykajiva Mar 13, 2024
c3ff5e0
Enable debug logging and refine SIP plugin
kinsleykajiva Mar 13, 2024
f0062b2
Enable debug logging and refine SIP plugin
kinsleykajiva Mar 15, 2024
218dec2
Improve SIP plugin and server startup logic
kinsleykajiva Mar 15, 2024
080fc87
Refactor SDP parsing and building logic
kinsleykajiva Mar 16, 2024
914d704
Refactor SIP UserAgent and add SIP events
kinsleykajiva Mar 16, 2024
9b512f9
Refactor SipUserAgentPlugin to improve media handling
kinsleykajiva Mar 22, 2024
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7 changes: 7 additions & 0 deletions clients-examples/web/includes/left-nav.php
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Original file line number Diff line number Diff line change
Expand Up @@ -7,6 +7,13 @@
<i class="bi bi-grid"></i>
<span>G-Streamer</span>
</a>
<a class="nav-link collapsed" href="sipgateway-example.php">
<i class="bi bi-grid"></i>
<span>Sip Gateway</span>
</a>



</li><!-- End Dashboard Nav -->


Expand Down
127 changes: 114 additions & 13 deletions clients-examples/web/rippleApp.js
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Open in desktop
Original file line number Diff line number Diff line change
@@ -1,23 +1,23 @@
'use strict';
const RippleSDK = {
serverUrl : '',
serverName : '',
serverName : '',
timeZone : Intl.DateTimeFormat().resolvedOptions().timeZone,
clientID : '',
isDebugSession : false,
clientTypesAllowed : ['Firefox', 'Chrome'],
featuresAvailable : Object.freeze({VIDEO_ROOM: 'VIDEO_ROOM',AUDIO_ROOM: 'AUDIO_ROOM',VIDEO_CALL: 'VIDEO_CALL',G_STREAM:'G_STREAM',G_STREAM_BROADCAST:'G_STREAM_BROADCASTER',G_STREAM_BROADCAST_CONSUMER:'G_STREAM_BROADCAST_CONSUMER'}),
featuresAvailable : Object.freeze({VIDEO_ROOM: 'VIDEO_ROOM',AUDIO_ROOM: 'AUDIO_ROOM',VIDEO_CALL: 'VIDEO_CALL',G_STREAM:'G_STREAM',G_STREAM_BROADCAST:'G_STREAM_BROADCASTER',G_STREAM_BROADCAST_CONSUMER:'G_STREAM_BROADCAST_CONSUMER',SIP_GATEWAY:'SIP_GATEWAY'}),
app : {
iceServerArray : [{urls: 'stun:stun.l.google.com:19302'},{urls: "stun:stun.services.mozilla.com"}],
isAudioAccessRequired : false,
isVideoAccessRequired : false,
hasAccessToVideoPermission : false,
hasAccessToAudioPermission : false,
maxRetries: 10,
maxRetries : 10,
remindServerTimeoutInSeconds: 50 ,
reminderInterval : null,
mediaUI:{
renderGroupParentId:""
reminderInterval : null,
mediaUI :{
renderGroupParentId:""
},
webRTC:{
EVENT_NAMES : {
Expand Down Expand Up @@ -90,6 +90,7 @@ const RippleSDK = {
requestType: 'answer',
transaction: RippleSDK.utils.uniqueIDGenerator("transaction", 12),
threadRef : threadRef,
feature : RippleSDK.utils.threadRefsInUseMap.get(threadRef).feature,
answer : answer.sdp
};
RippleSDK.transports.websocket.webSocketSendAction(body);
Expand Down Expand Up @@ -119,6 +120,28 @@ const RippleSDK = {
});
RippleSDK.utils.log('consumeAnswer', 'answer consumed');

},requestUserMediaPermission:async ()=>{
if(RippleSDK.app.isAudioAccessRequired || RippleSDK.app.isVideoAccessRequired){
try {
if(RippleSDK.app.isAudioAccessRequired || RippleSDK.app.isVideoAccessRequired){
const stream = await navigator.mediaDevices.getUserMedia({
audio: RippleSDK.app.isAudioAccessRequired,
video: RippleSDK.app.isVideoAccessRequired
});
RippleSDK.app.webRTC.localStream = stream;
RippleSDK.app.callbacks.tellClientOnMediaAccessPermissionEvent(
RippleSDK.app.isAudioAccessRequired,
RippleSDK.app.isVideoAccessRequired,
stream
);
return stream;
}
return null;
} catch(err) {
RippleSDK.utils.error('requestUserMediaPermission', err);
return null;
}
}
},
createOffer:async (threadRef)=>{
if(!threadRef){
Expand All @@ -134,15 +157,64 @@ const RippleSDK = {
threadRef,
});
const peerConnection = RippleSDK.app.webRTC.peerConnectionsMap.get(threadRef);
await peerConnection.createOffer({offerToReceiveVideo: true, offerToReceiveAudio: true})
const feature = RippleSDK.utils.threadRefsInUseMap.get(threadRef).feature;
let offerObj = {
offerToReceiveVideo: true, offerToReceiveAudio: true
}
if(feature === RippleSDK.featuresAvailable.SIP_GATEWAY) {
offerObj.offerToReceiveVideo = false;
RippleSDK.app.isAudioAccessRequired = true;
RippleSDK.app.isVideoAccessRequired = false;
}
const stream=await RippleSDK.app.webRTC. requestUserMediaPermission();
function removeOpusCodec(sdp) {
// Split the SDP data into an array of lines
const sdpLines = sdp.split('\n');

// Define the start and end of the opus codec section
const opusStartIndex = sdpLines.findIndex(line => line.includes('a=rtpmap:111 opus/48000/2'));
const opusEndIndex = sdpLines.findIndex(line => line === 'a=rtpmap:63 red/48000/2');

// If both start and end indices exist, remove the opus codec section
if (opusStartIndex !== -1 && opusEndIndex !== -1) {
sdpLines.splice(opusStartIndex, opusEndIndex - opusStartIndex);
}

// Join lines back into a single SDP string
var newSdp = sdpLines.join('\n');

return newSdp;
}
if(stream && feature === RippleSDK.featuresAvailable.SIP_GATEWAY){
stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));
}

await peerConnection.createOffer(offerObj)
.then(async _sdp=>{
if(feature === RippleSDK.featuresAvailable.SIP_GATEWAY) {
let sdp_ = _sdp.sdp;
// Remove video codecs
sdp_ = sdp_.replace(/m=video[\s\S]*?(?=m=|$)/g, '');
// remove G722/8000 , PCMU/8000 , CN/8000 , CN/16000 , CN/32000 , CN/48000 , CN/96000 and keep PCMA/8000
sdp_ = sdp_.replace(/(a=rtpmap:(\d+)\s(G722|PCMU|CN)\/8000\r\n|a=rtpmap:(\d+)\sCN\/(16000|32000|48000|96000)\r\n)/g, '');
//sdp_ = sdp_.replace(/a=rtpmap:111 opus\/48000\/2\r\n/g, '');
/*sdp_=sdp_.split('\n')
.filter(line => !line.includes('opus'))
.join('\n');*/
console.log('1x',sdp_)
// sdp_ = removeOpusCodec(sdp_);
sdp_ = (sdp_).replace('a=rtpmap:111 opus/48000/2\n','');
console.log('1x',sdp_)
_sdp.sdp = sdp_;
}
await peerConnection.setLocalDescription(_sdp);
const body = {
clientID : RippleSDK.clientID,
requestType: 'offer',
transaction: RippleSDK.utils.uniqueIDGenerator("transaction", 12),
threadRef : threadRef,
offer : _sdp.sdp
offer : _sdp.sdp,
feature,
};
RippleSDK.transports.websocket.webSocketSendAction(body);
RippleSDK.utils.log('createOffer', 'offer sent to server',body);
Expand Down Expand Up @@ -255,6 +327,7 @@ const RippleSDK = {
requestType : 'iceCandidate',
transaction : RippleSDK.utils.uniqueIDGenerator("transaction",12),
threadRef : threadRef,
feature : RippleSDK.utils.threadRefsInUseMap.get(threadRef).feature,
candidate : ev.candidate.candidate,
sdpMid : ev.candidate.sdpMid,
sdpMLineIndex: ev.candidate.sdpMLineIndex
Expand Down Expand Up @@ -350,6 +423,9 @@ const RippleSDK = {

},
features:{
sipGateway:{
makeCall:(phoneNumber)=>{ },
},
streaming:{
threads:[],
functions:{
Expand Down Expand Up @@ -601,10 +677,21 @@ const RippleSDK = {
messageObject = JSON.parse(messageObject);
}
const eventType = messageObject.eventType;
const feature = messageObject.feature;
const success = messageObject.success;
const plugin = messageObject.plugin;
let pluginEventType = plugin ? plugin.eventType: null;
if (success && pluginEventType) {
if (pluginEventType === 'sipRegistration') {
RippleSDK.app.callbacks.tellClientOnMessage(plugin);

// maybe we can create a pear connection here or create when its required!
// RippleSDK.app.webRTC.createPeerConnection(messageObject.position);

RippleSDK.app.webRTC.peerConnectionsMap.set(messageObject.position, RippleSDK.app.webRTC.createPeerConnection(messageObject.position));
await RippleSDK.app.webRTC.createOffer(messageObject.position);
}

if (pluginEventType === 'webrtc') {
RippleSDK.app.webRTC.remoteOfferStringSDPMap.set(messageObject.position, plugin.clientSDP);
await RippleSDK.app.webRTC.createAnswer(messageObject.position);
Expand Down Expand Up @@ -666,10 +753,22 @@ const RippleSDK = {
if (success && eventType) {
switch (eventType) {
case "newThread":
RippleSDK.app.features.streaming.threads.push(messageObject.position);
RippleSDK.app.features.streaming.functions.startBroadCast(messageObject.position);
// renderThreadUI
RippleSDK.app.features.streaming.functions.renderThreadUI(messageObject.position);
RippleSDK.utils.threadRefsInUseMap.set(messageObject.position, {
feature: messageObject.feature,
accessAuth: messageObject.accessAuth,
});
if (feature === RippleSDK.featuresAvailable.G_STREAM_BROADCAST ||
feature === RippleSDK.featuresAvailable.G_STREAM) {

RippleSDK.app.features.streaming.threads.push(messageObject.position);
RippleSDK.app.features.streaming.functions.startBroadCast(messageObject.position);
// renderThreadUI
RippleSDK.app.features.streaming.functions.renderThreadUI(messageObject.position);
}
if (feature === RippleSDK.featuresAvailable.SIP_GATEWAY) {

}

break;
case "register":
RippleSDK.app.startToRemindServerOfMe();
Expand All @@ -683,6 +782,7 @@ const RippleSDK = {
},
tellClientOnWebRtcEvents:eventMessage=>{},
tellClientOnStreamUIUpdates:eventMessage=>{},
tellClientOnMediaAccessPermissionEvent:(hasAudio,hasVideo,stream)=>{},
tellClientOnConnected:null,
onConnected:()=>{
RippleSDK.utils.log('onConnected');
Expand Down Expand Up @@ -731,7 +831,7 @@ const RippleSDK = {
}
if(RippleSDK.transports.websocket.socket.readyState !== 1){
RippleSDK.utils.log('webSocketSendAction', 'not ready');
return;

}else{
messageObject. isDebugSession= RippleSDK.isDebugSession,
messageObject.transaction = RippleSDK.utils.uniqueIDGenerator("transaction",12);
Expand Down Expand Up @@ -798,6 +898,7 @@ const RippleSDK = {
info : console.info.bind(console),
trace : console.trace.bind(console),
assert : console.assert.bind(console),
threadRefsInUseMap:new Map(),
convertToWebSocketUrl: (url) => {
if (url.startsWith('https://')) {
return url.replace('https://', 'wss://');
Expand Down
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