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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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Integrate SIP.js with React Native

Is there a way to integrate SIP.js (both audio and video calling are needed) with React Native? Importing the library itself is easy enough, but the issues I'm running into are: WebRTC support: instead of using the browser's WebRTC functionality…
penguinrob
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How to use a JS SIP library using a non-websockets transport?

This question was asked by somebody on Quora here but there was no answer. Is there any work around solution for this, other than changing the source code of jsSip? The issue is that most JS SIP libraries that work with webRTC do so through…
Sunny
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upgrade/downgrade audio - video

I am trying to upgrade audio to video using re-negotiate toggleVideo(enableVideo) { const { session } = this.state; const constraints = { offerToReceiveVideo: enableVideo, }; const options = { useUpdate: true, …
android_learner
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A 40 sec delay of SIP call initiation using JSSIP / WebRTC

I am developing a JavaScript-based web SIP client communicating with Asterisk SIP server. The SIP client is using JSSIP 3.4.2, I'm testing on Chrome version 80. Both SIP client and SIP server are behind firewalls. I'm using STUN server…
vmayorow
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RTCPeerConnection addstream event not fired in firefox not focused or minimized

Does anyone know why the addstream event doesn't work when firefox is out of focus or minimized? im using jssip 3.2 library. In chrome this does not happen and the event callback is executed with chrome being minimized or without the need to have…
Dixon Pedraza
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Callback event when WebSocket connection to 'wss://phone.company.com:5063/' failed

We are trying to connect WebSocket using below code, found no issues when the internet is working fine. But when internet is down or due to some other N/W related issues this message getting displayed in the browser console: "WebSocket connection…
Cherry
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Chrome MediaRecorder API can not record REMOTE video from FreeSwitch, but local media can be recorded

Issue: When we record remote video (only) streams in Chrome by using the built-in MediaRecorder API, it returns blob objects; when we merge all blob files with the Blob API and create an object URL, it’s not playable with html5 players. I’m using…
Jack
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How to handle audio stream in JsSIP?

I'm creating React application that use JsSIP library to answer calls made via VoIP SIP provider. I've already created a page that have two buttons (Accept and Reject). It successfully register SIP client on SIP-server. It also successfully receive…
Michael Sivolobov
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JsSIP Firefox - InvalidSessionDescriptionError: Answer tried to set recv when offer did not set send

I was recently working with JsSIP. We tried to connect conference with one way (incoming) audio stream, but for Firefox was not working. Firefox error message: InvalidSessionDescriptionError: Answer tried to set recv when offer did not set…
Jack
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No audio /// WebRTC + Asterisk + jsSIP in Local Network

I'd appreciate a lot your help with this issue. I'm running a very basic script of JS with a jsSIP User Agent that uses a local Asterisk server for making voice calls. Everything is on a private network, and I don't get any warnings or errors from…
power.ponch
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How can I reduce the time it takes for echo cancellation to normalize the voice

I am working on a WebRTC project using the jsSIP library. One user connects from their browser, and the other connects by making a real phone call. While reviewing the audio recordings, I noticed a problem. If there is high background noise on the…
eren
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" 488 Incompatible SDP " when trying to send invite request to FreeSwitch with jssip library

Invite request always fails with 488 code. I tried to change the priority of codecs, but nothing helps. I think FreeSwitch is expecting another sdp parameters from what I'm sending to. But I can't figure it out. Here is the log from jssip debugger,…
Avdey
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How to print RTP and RTCP packet messages using JsSIP?

I am using JsSIP to make SIP calls. I am able to see Signalling messages like From, To, Via in the console. In the same way, can we see RTP and RTCP packet transmission messages?
Cherry
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Java gateway for webrtc <--> sip communication

I am implementing a java gateway for the compatibility between webrtc and sip. I have done sdp exchange part. But now i am stuck in media part. So far i assume that i need to implement dtls-srtp handshake and then the encryption, decryption part. So…
Rashed
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JsSIP Implementation

I'm working on a telecom company. We want to develop a SIP phone wit JsSIP library. But we can't do it. We create a basic form for test like this: var socket = new JsSIP.WebSocketInterface('*****'); var configuration = { sockets : [ socket ], …
J. Doe
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