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Questions tagged [jssip]

Use this tag for questions related to the JavaScript SIP Library.

With JsSIP you can build a complete SIP user agent in your Web page:

  • Send and/or receive multimedia calls.
  • Send and/or receive text messages.

JsSIP is a SIP WebSocket client. It needs a SIP WebSocket capable server to which connect and exchange SIP messages.

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Module '"events"' has no exported member 'Listener' when using jssip in angular 14

I am getting following error in my angular project when using jssip library. Error: node_modules/jssip/lib/RTCSession.d.ts:1:23 - error TS2305: Module '"events"' has no exported member 'Listener'. I am using angular 14 and typescript version…
Dnyaneshwer Pendurkar
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JSSIP how to recived a video call without allow camera permission

I need to implement one way video call, both user can talk with audio. but the receiver will not show webcam and caller must show webcam. Are JSSIP have any option to disable some user camera ? or will I should do for workaround ?
Thanatcha pitithadakul
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Unable to make a call form JSSIP to my phone number and I have the CTI server credentials

I am working on CTI project I want to create cti phone for my company that can work fine with the company requirement, currently I am using following code for making call. const JsSIP = require('jssip'); const NodeWebSocket =…
shanjay
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Connect to specific port range in Asterisk Server from JsSIP front end

JsSIP is currently trying to connect to audio (after accepting incoming call) to random ports over UDP. How can I restrict it to a specific to a port range for the audio? Edit: Log from Asterisk Exp:m=audio 63485 UDP/TLS/RTP/SAVPF 8 107 0 101 (here…
Sanket Pandia
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422 Session Interval Too Small While trying to connect by jsSIP to Freeswitch

I am trying to have a call from the browser by jsSIP through Freeswitch. The registration is passed. When I call It sends an invitation packet. Freeswitch answers with an error. I've found a similar case on the internet. I've configured…
Victor Shelepen
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JsSIP How to detect call transfer successfullly

A holding session with B and calling on session with C. A transfer B to C by implementing code below. How to get info of B in C, i tried listening ua.on('newMessage', cb) and adding ua.sendMessage(currentSession?.remote_identity.uri, 'dataOfB') in A…
Yaokaoya
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The protocol is not available, Signalwire Relay Client Subscription

I am trying to subscribe to Signalwire phone numbers using Relay SDK with following code: this.client1 = new Relay({ project: 'myprojectid', token: jwtToken, }) this.client1.subscribe({ protocol: "WSS", …
Saira Nawaz
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How to get Status code for Trying and Ringing in JsSip

How to get Status code like 100 and 180 from response when i call with any number from below response in my console.console output for my call response as below so how to get status code for trying and ringing when i call with nay number. [1]:…
kevin ladani
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Not able to manage multiple jssip phone session request in react

Working one react project based on functional components only. Project contains Audio Call functions using JSSIP. Calling functions working fine with single call. When multiple call happens I can able to display multiple calling popup at bottom of…
Paresh Gami
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Bind mediaStream to

I've been trying to make a simple video-calling interface with JsSIP, so far I've only managed to init a videocall and the receiver gets my audio and video streams, but when I'm trying to add the stream (local or remote) to a
Eduardo Galeano
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Blind Transfer in JSSIP

I have been trying to make a blind transfer on a ongoing call. Below is the code i have implemented: transfersession(ext) { this.rtcSession.refer('sip:' + ext + '@' + serveraddress); } Can someone tell is there something more I have to…
Ram
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Javascript code to Register SIP endpoint from browser

I just want to Register a SIP or PJSIP endpoint of asterisk from browser. I know some libraries which can do this for me. But I want to know the core code in javascript to register a SIP end point. It is done like below in jssip.js library var…
Anup_Tripathi
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JsSip Answer addstream not called

We are using jssip 3.3.4 to handle incoming calls. We are seeing a few systems where the call is not answered when the user clicks the answer button. The answer button essentially…
Brett Sutton
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JSSIP How to switch between audio call to video call

I am new in JSSIP. I need to switch Audio call to video call in ongoing call. const session = userAgent.call(destinationNumber, { mediaConstraints: { audio: true, video: false }, pcConfig: { iceServers: [{…
Jibin Francis
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Passing CallerID for PSTN in From Header using JSSIP

I'm using JsSIP to connect to FreeSwitch and then to the PSTN. I'm looking to pass the callerID in the From header. I have my code set up somewhat like this: var TheCallerIDTest = '+33...number in E164 format'; var TheSipClient = new…
frenchie
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