JsSIP

Hi Usman, Did you find a way to achieve this? On Tuesday, 22 June 2021 at 08:58:55 UTC+12 usmana...@

how to capture the nonce of a 401 response in order to generate the ha1 and be able to authenticate?

Running jssip 3.9.1. Freepbx has a feature where it can send a prompt over early media and expects a

Hi all! I would like to get some recommendations on guides of how to implement e2e SIP. From the

40 seconds ? Look likes ICE gathering delay. On Wednesday, June 15, 2022 at 9:21:25 PM UTC+3 bker...@

Hello all, We have built a application using Jssip 3.5 version and since 1 week there are seen

Hi Gustavo, JsSIP makes use of the WebRTC implementation of the browser, hence all the metrics you

On Tue, 17 May 2022 13:12:21 +0200 'José Luis Millán' via JsSIP <jssip@googlegroups.com

Reconnection will happen in random interval times between the given config values: https://jssip.net/

Ok it seems that my pbx needs an ACL to allow media to start. I modified the example code to show to

Looking for a few ideas of where to start looking. Node.js application, not running in a browser,

Just for completness... Finaly I have found a solution that seem work. I used the terminate method

Hi, Our initial idea to connect JsSIP WebRTC client to a SIP Server / IMS was using Asterisk. However

Hi. I'm trying to create a functionality when one softphone register, another one with the same

ICE Gathering fail on chrome 100.0.4896.127 (MAc) Issue: When we try to make calls we are getting an

I have two numbers in my trunk line. Is there any method or header to set up the caller id of my

Hello, I am looking for some guidance on attended transfer Refer-To, and whether the below might be a

Yeah… There's WebRTC, and then there's the outermost “user interface” to WebRTC, which sadly

Hello guys I have a question about a setup done with asterisk. I receive every 15 seconds an empty

Hello, Looking through codebase and examples I've got the impression that usage of Trickle ICE in

Application is running in node.js. JSSIP sends INVITE with local candidates, and receives 200 OK from

Hello, First of all, thank you so much to all contributors, I´ve been using the jssip library for a

Thanks, SDP in 200 OK had extra JsSIP:Transport a=rtcp-fb:111 transport-cc After removing that, the

Hi Ryan, Did you find the solution yet? I am having the same error. Can you please describe the

Hi, I am experiencing a problem that during a phone call, JsSip re-register with a new call-id, so my

Did some more digging into the source code of JsSIP. The condition that triggers the error message is

Did you manage to solve this problem? воскресенье, 19 сентября 2021 г. в 03:48:11 UTC+3, Alpt: Hi All

Is it possible to test the websocket connection in an app like postman before implementing in JsSip I

That was quick ! Thank you ! Will try. El jueves, 17 de marzo de 2022 a las 13:40:37 UTC+1, abal...@

You can play DTMF sound using AudioContext https://developer.mozilla.org/en-US/docs/Web/API/