dtolb/jssip.md

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JsSip Demo

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jssip.md

JSSIP with Catapult API

⚠️ This has been updated at JsFiddle*

Prerequisites

Register for a Catapult (Bandwidth Application Platform) account here
Register a SIP domain
Create an endpoint/user
If you want to make calls to the PSTN (normal phones) you will need a server to handler events from Catapult
Make phone calls For a more in depth guide, view this article

Quick Start

Full docs are here CDN hosted library: minified not-minified JSFIDDLE Demo Client

Outbound Call

var callOptions = {
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};

var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'password': 'password', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443'
});
bwPhone.start();

bwPhone.on("registered", function(){
    bwPhone.call("222-333-4444", callOptions);    
});

bwPhone.on("newRTCSession", function(data){
    var session = data.session; // outgoing call session here
    var dtmfSender;
    session.on("confirmed",function(){
        //the call has connected, and audio is playing
        var localStream = session.connection.getLocalStreams()[0];
        dtmfSender = session.connection.createDTMFSender(localStream.getAudioTracks()[0])
    });
    session.on("ended",function(){
        //the call has ended
    });
    session.on("failed",function(){
        // unable to establish the call
    });
    session.on('addstream', function(e){
        // set remote audio stream (to listen to remote audio)
        // remoteAudio is <audio> element on page
        remoteAudio.src = window.URL.createObjectURL(e.stream);
        remoteAudio.play();
    });
    
    //play a DTMF tone (session has method `sendDTMF` too but it doesn't work with Catapult server right)
    dtmfSender.insertDTMF("1");
    dtmfSender.insertDTMF("#");

    //mute call
    session.mute({audio: true});

    //unmute call
    session.unmute({audio: true});

    //to hangup the call
    session.terminate();

    
});

Inbound Call

var callOptions = {
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};

var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'password': 'password', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443'
});
bwPhone.start();

bwPhone.on("newRTCSession", function(data){
    var session = data.session; 
    
    if (session.direction === "incoming") {
        // incoming call here
        session.on("accepted",function(){
            // the call has answered
        });
        session.on("confirmed",function(){
            // this handler will be called for incoming calls too
        });
        session.on("ended",function(){
            // the call has ended
        });
        session.on("failed",function(){
            // unable to establish the call
        });
        session.on('addstream', function(e){
            // set remote audio stream (to listen to remote audio)
            // remoteAudio is <audio> element on page
            remoteAudio.src = window.URL.createObjectURL(e.stream);
            remoteAudio.play();
        });
        
        // Answer call
        session.answer(callOptions);
        
        // Reject call (or hang up it)
        session.terminate();
    }
});

Passwordless connection (via auth token)


var authToken = "1234567890"; // you can get this token by POST http request to /v1/users/<userId>/domains/<domainId>/endpoints/<endpointId>/tokens

var authHeader = "X-Callsign-Token: " + authToken;

var callOptions = {
  extraHeaders: [authHeader], // set auth token here (it will be passed on making calls and answering incoming call) 
  mediaConstraints: {
    audio: true, // only audio calls
    video: false
  }
};

var bwPhone = new JsSIP.UA({
  'uri': 'sip:[email protected]', 
  'ws_servers': 'wss://webrtc.registration.bandwidth.com:10443',
});
bwPhone.registrator().setExtraHeaders([authHeader]); // set auth header on register

bwPhone.on('registered', function(){
     // ready to make calls and receive incoming calls
     // making a call
     bwPhone.call("222-333-4444", callOptions);
});

bwPhone.on("newRTCSession", function(data){
    var session = data.session; 
    
    if (session.direction === "incoming") {
        // answer incoming call
        session.answer(callOptions);
    }
});

bwPhone.start();

Supported Browsers

Firefox
Chrome
Opera You can check if browser supports WebRTC by code

    if (!JsSIP.rtcninja.hasWebRTC()) {
        alert("WebRTC is not supported. Please use another browser.");
    }

Thank you

Thank you, @GaborTorma for updating and answering questions.

lavkushtyagi commented Nov 8, 2023 •

edited

Loading

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createAnswer: function createAnswer() { [native code] },
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createDTMFSender: function createDTMFSender() { [native code] },
createOffer: function createOffer() { [native code] },
currentLocalDescription: null,
currentRemoteDescription: null,
dispatchEvent: function dispatchEvent() { [native code] },
getConfiguration: function getConfiguration() { [native code] },
getLocalStreams: function getLocalStreams() { [native code] },
getReceivers: function getReceivers() { [native code] },
getRemoteStreams: function getRemoteStreams() { [native code] },
getSenders: function getSenders() { [native code] },
getStats: function getStats() { [native code] },
getTransceivers: function getTransceivers() { [native code] },
iceConnectionState: "new",
iceGatheringState: "new",
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ontrack: null,
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remoteDescription: null,
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restartIce: function restartIce() { [native code] },
sctp: null,
setConfiguration: function setConfiguration() { [native code] },
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setRemoteDescription: function setRemoteDescription() { [native code] },
signalingState: "stable"
}
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"%cJsSIP:RTCSession %ccreateLocalDescription()%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
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body: "v=0
o=- 2164683428557940697 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
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a=candidate:2979755658 1 tcp 1518280447 10.10.10.135 9 typ host tcptype active generation 0 network-id 1
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a=ice-pwd:atUYKfRy9vhp0nMT847h2Fbt
a=ice-options:trickle
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a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
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a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
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a=rtpmap:63 red/48000/2
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a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
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a=rtpmap:126 telephone-event/8000
a=ssrc:531608100 cname:XCbvh2SbM0Pf6D7b
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",
call_id: "0fp8ni4redbckebe3q6j",
cseq: 6701,
extraHeaders: ["Contact: <sip:[email protected];transport=ws;ob>", "Content-Type: application/sdp", "Session-Expires: 90"],
from: {
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_uri: { ... }
},
headers: {
Call-ID: ["0fp8ni4redbckebe3q6j"],
CSeq: ["6701 INVITE"],
From: ["<sip:[email protected]>;tag=4o2j6p6jqg"],
Max-Forwards: [69],
To: ["<sip:[email protected]>"],
Via: [""]
},
method: "INVITE",
ruri: {
_headers: { ... },
_host: "sip01.sipprovider.com",
_parameters: { ... },
_port: undefined,
_scheme: "sip",
_user: "9026951168"
},
to: {
_display_name: null,
_parameters: { ... },
_uri: [circular object Object]
},
transaction: null,
ua: {
_applicants: { ... },
_cache: { ... },
_closeTimer: null,
_configuration: { ... },
_contact: { ... },
_data: { ... },
_dialogs: { ... },
_dynConfiguration: { ... },
_error: null,
_events: { ... },
_eventsCount: 2,
_maxListeners: undefined,
_registrator: { ... },
_sessions: { ... },
_status: 1,
_transactions: { ... },
_transport: { ... },
addListener: function addListener(type, listener) {
return _addListener(this, type, listener, false);
},
emit: function emit(type) {
var er, handler, len, args, i, events;
var doError = (type === 'error');

events = this._events;
if (events)
  doError = (doError && events.error == null);
else if (!doError)
  return false;

// If there is no 'error' event listener then throw.
if (doError) {
  if (arguments.length > 1)
    er = arguments[1];
  if (er instanceof Error) {
    throw er; // Unhandled 'error' event
  } else {
    // At least give some kind of context to the user
    var err = new Error('Unhandled "error" event. (' + er + ')');
    err.context = er;
    throw err;
  }
  return false;
}

handler = events[type];

if (!handler)
  return false;

var isFn = typeof handler === 'function';
len = arguments.length;
switch (len) {
    // fast cases
  case 1:
    emitNone(handler, isFn, this);
    break;
  case 2:
    emitOne(handler, isFn, this, arguments[1]);
    break;
  case 3:
    emitTwo(handler, isFn, this, arguments[1], arguments[2]);
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    emitThree(handler, isFn, this, arguments[1], arguments[2], arguments[3]);
    break;
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  default:
    args = new Array(len - 1);
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      args[i - 1] = arguments[i];
    emitMany(handler, isFn, this, args);
}

return true;

},
eventNames: function eventNames() {
return this._eventsCount > 0 ? Reflect.ownKeys(this._events) : [];
},
getMaxListeners: function getMaxListeners() {
return $getMaxListeners(this);
},
listenerCount: function listenerCount(type) {
var events = this._events;

if (events) {
  var evlistener = events[type];

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listeners: function listeners(type) {
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on: function addListener(type, listener) {
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this.on(type, _onceWrap(this, type, listener));
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return _addListener(this, type, listener, true);
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this.prependListener(type, _onceWrap(this, type, listener));
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rawListeners: function rawListeners(type) {
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removeAllListeners: function removeAllListeners(type) {
var listeners, events, i;

    events = this._events;
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    // not listening for removeListener, no need to emit
    if (!events.removeListener) {
      if (arguments.length === 0) {
        this._events = objectCreate(null);
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      } else if (events[type]) {
        if (--this._eventsCount === 0)
          this._events = objectCreate(null);
        else
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    // emit removeListener for all listeners on all events
    if (arguments.length === 0) {
      var keys = objectKeys(events);
      var key;
      for (i = 0; i < keys.length; ++i) {
        key = keys[i];
        if (key === 'removeListener') continue;
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      this._events = objectCreate(null);
      this._eventsCount = 0;
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    listeners = events[type];

    if (typeof listeners === 'function') {
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    } else if (listeners) {
      // LIFO order
      for (i = listeners.length - 1; i >= 0; i--) {
        this.removeListener(type, listeners[i]);
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    return this;
  },
removeListener: function removeListener(type, listener) {
    var list, events, position, i, originalListener;

    if (typeof listener !== 'function')
      throw new TypeError('"listener" argument must be a function');

    events = this._events;
    if (!events)
      return this;

    list = events[type];
    if (!list)
      return this;

    if (list === listener || list.listener === listener) {
      if (--this._eventsCount === 0)
        this._events = objectCreate(null);
      else {
        delete events[type];
        if (events.removeListener)
          this.emit('removeListener', type, list.listener || listener);
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      for (i = list.length - 1; i >= 0; i--) {
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}
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}, "color: #99CC00"
"%cJsSIP:Transport %csend()%c +5s", "color: #FF0066", "color: inherit", "color: #FF0066"
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INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS 526h81p4pd99.invalid;branch=z9hG4bK2918698
Max-Forwards: 69
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6701 INVITE
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
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Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.4.2
Content-Length: 1489

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t=0 0
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a=ice-pwd:atUYKfRy9vhp0nMT847h2Fbt
a=ice-options:trickle
a=fingerprint:sha-256 E3:0B:40:B6:8A:B1:9D:19:6C:C6:E8:1B:C0:D2:BF:11:E7:C6:BE:A2:25:D7:1E:43:34:C0:37:9D:C1:64:AF:2B
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
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%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:WebSocketInterface %csend()%c +5s", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +48ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/WSS 526h81p4pd99.invalid;received=103.106.194.125;rport=48134;branch=z9hG4bK2918698
To: <sip:[email protected]>;tag=5944.8e4b8ba743f000761380c53bf8df60de
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6701 INVITE
Proxy-Authenticate: Digest realm="sip01.sipprovider.com", nonce="71Cnw3v3ZEwaGFMYLshtwX2VmewZVuU7lhN+vVuEsPgA", qop="auth"
Server: OpenSIPS (3.3.0 (x86_64/linux))
Content-Length: 0

%c +48ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csend()%c +1ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csending message:

ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS 526h81p4pd99.invalid;branch=z9hG4bK2918698
Max-Forwards: 69
To: <sip:[email protected]>;tag=5944.8e4b8ba743f000761380c53bf8df60de
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6701 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.4.2
Content-Length: 0

%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:WebSocketInterface %csend()%c +1ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:DigestAuthentication %cauthenticate() | using qop=auth [a2:"%s"]%c +31s", "color: #FF9933", "color: inherit", "INVITE:sip:[email protected]", "color: #FF9933"
"%cJsSIP:DigestAuthentication %cauthenticate() | response generated%c +0ms", "color: #FF9933", "color: inherit", "color: #FF9933"
"%cJsSIP:Transport %csend()%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csending message:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WSS 526h81p4pd99.invalid;branch=z9hG4bK5677102
Max-Forwards: 69
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 INVITE
Proxy-Authorization: Digest algorithm=MD5, username="x245akwD", realm="sip01.sipprovider.com", nonce="71Cnw3v3ZEwaGFMYLshtwX2VmewZVuU7lhN+vVuEsPgA", uri="sip:[email protected]", response="c134a1af7854e245e5cffa497b44e049", qop=auth, cnonce="f2f2h6km1ib5", nc=00000001
Contact: <sip:[email protected];transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSIP 3.4.2
Content-Length: 1489

%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:WebSocketInterface %csend()%c +1ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:InviteClientTransaction %cTimer D expired for transaction z9hG4bK2918698%c +14s", "color: #FF3333", "color: inherit", "color: #FF3333"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +50ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

SIP/2.0 100 Giving it a try
Via: SIP/2.0/WSS 526h81p4pd99.invalid;received=103.106.194.125;rport=48134;branch=z9hG4bK5677102
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 INVITE
Server: OpenSIPS (3.3.0 (x86_64/linux))
Content-Length: 0

%c +51ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:RTCSession %creceiveInviteResponse()%c +103ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +100ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 526h81p4pd99.invalid;rport=48134;received=103.106.194.125;branch=z9hG4bK5677102
Record-Route: <sip:103.163.40.230;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
Record-Route: <sip:103.163.40.230:443;transport=wss;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
From: <sip:[email protected]>;tag=4o2j6p6jqg
To: <sip:[email protected]>;tag=as67cf82c0
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 INVITE
Server: Asterisk PBX 18.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0

%c +100ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:RTCSession %creceiveInviteResponse()%c +100ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:Dialog %cnew UAC dialog created with status EARLY%c +14s", "color: #33CC33", "color: inherit", "color: #33CC33"
"%cJsSIP:RTCSession %csession progress%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "progress"%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +711ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

%c +712ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:RTCSession %creceiveInviteResponse()%c +711ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %csession progress%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "progress"%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +792ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS 526h81p4pd99.invalid;rport=48134;received=103.106.194.125;branch=z9hG4bK5677102
Record-Route: <sip:103.163.40.230;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
Record-Route: <sip:103.163.40.230:443;transport=wss;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
From: <sip:[email protected]>;tag=4o2j6p6jqg
To: <sip:[email protected]>;tag=as67cf82c0
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 INVITE
Server: Asterisk PBX 18.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 693

v=0
o=root 64994987 64994987 IN IP4 103.163.40.241
s=Asterisk PBX 18.6.0
c=IN IP4 103.163.40.241
t=0 0
m=audio 13584 UDP/TLS/RTP/SAVPF 111 0 8 126
a=maxptime:20
a=mid:0
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=sendrecv
a=rtcp:13584
a=rtcp-mux
a=setup:active
a=fingerprint:sha-256 01:F4:35:DC:48:F7:89:D3:A0:48:31:41:56:A0:67:8C:74:47:FC:97:B2:8A:A3:D1:D0:15:1E:ED:1B:36:14:33
a=ptime:20
a=ice-ufrag:ZScToNgg
a=ice-pwd:tGC7ADo18G2WBrgq2ksSpX8ne7
a=ice-options:trickle
a=candidate:jI2oon5gu1OTmegP 1 UDP 2130706431 103.163.40.241 13584 typ host
a=end-of-candidates

%c +791ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:RTCSession %creceiveInviteResponse()%c +792ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %csession progress%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "progress"%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "sdp"%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +11s", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 526h81p4pd99.invalid;rport=48134;received=103.106.194.125;branch=z9hG4bK5677102
Record-Route: <sip:103.163.40.230;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
Record-Route: <sip:103.163.40.230:443;transport=wss;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
From: <sip:[email protected]>;tag=4o2j6p6jqg
To: <sip:[email protected]>;tag=as67cf82c0
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 INVITE
Server: Asterisk PBX 18.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 90;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 693

%c +11s", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:RTCSession %creceiveInviteResponse()%c +11s", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:Dialog %cdialog 0fp8ni4redbckebe3q6j4o2j6p6jqgas67cf82c0 changed to CONFIRMED state%c +13s", "color: #33CC33", "color: inherit", "color: #33CC33"
"%cJsSIP:RTCSession %cemit "sdp"%c +1ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %csession accepted%c +2ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "accepted"%c +1ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %csendRequest()%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:Transport %csend()%c +6ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csending message:

ACK sip:[email protected]:5060 SIP/2.0
Route: <sip:103.163.40.230:443;transport=wss;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
Route: <sip:103.163.40.230;r2=on;lr;ftag=4o2j6p6jqg;did=f87.d2e83441>
Via: SIP/2.0/WSS 526h81p4pd99.invalid;branch=z9hG4bK1933423
Max-Forwards: 69
To: <sip:[email protected]>;tag=as67cf82c0
From: <sip:[email protected]>;tag=4o2j6p6jqg
Call-ID: 0fp8ni4redbckebe3q6j
CSeq: 6702 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSIP 3.4.2
Content-Length: 0

%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:WebSocketInterface %csend()%c +6ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:RTCSession %csession confirmed%c +1ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:RTCSession %cemit "confirmed"%c +0ms", "color: #99CC00", "color: inherit", "color: #99CC00"
"%cJsSIP:WebSocketInterface %creceived WebSocket message%c +12s", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:Transport %creceived text message:

OPTIONS sip:[email protected];transport=ws SIP/2.0
Via: SIP/2.0/WSS 103.163.40.230:443;branch=z9hG4bK572.edb8810000003501
From: sip:[email protected];tag=07b883d3
To: sip:[email protected];transport=ws
Call-ID: [email protected]
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0

%c +12s", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csend()%c +1ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:Transport %csending message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 103.163.40.230:443;branch=z9hG4bK572.edb8810000003501
To: sip:[email protected];transport=ws;tag=nsfmc92lc3
From: sip:[email protected];tag=07b883d3
Call-ID: [email protected]
CSeq: 1 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Accept: application/sdp, application/dtmf-relay
Supported: outbound
Content-Length: 0

%c +0ms", "color: #FF0066", "color: inherit", "color: #FF0066"
"%cJsSIP:WebSocketInterface %csend()%c +1ms", "color: #66CC33", "color: inherit", "color: #66CC33"
"%cJsSIP:NonInviteServerTransaction %cTimer J expired for transaction z9hG4bK572.edb8810000003501%c +30s", "color: #6633CC", "color: inherit", "color: #6633CC"

the attached log after call pickup successfully

lavkushtyagi commented Nov 8, 2023

Generally it is codec, NAT or route problem.

I want to made audio call
the attached log after successfully call made and pickup