JsSIP

What's the problem? You get the remote audio track from the corresponding PeerConnection, you

Hi friends I install freeswitch, write test code for register jssip-ua on freeswitch js register with

Hello, I need change the SDP, specifically in the m= line by the audio and RTP/AVP. Additionally, i

Hi Team, I'd like to make a conference between a security camera (It has a built-in softphone -

Hi, The session instance has a terminate() method: https://jssip.net/documentation/3.9.x/api/session/

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Oh yes, in jssip.net. El El jue, 28 jul 2022 a las 9:19, MD. TAHIDUR RAHMAN <tahidur.cse@gmail.com

Hi. I have problems with some requests, for example BYE does not work and I have the error message

Hi, I need to manipulate the SDP of an answer to an Invite request before it is sent out.

Hi, Perhaps this is not really a question about JsSIP, but I'm hoping you might be able to help.

Good morning. I am trying to register, I receive the 401 Unauthorized message but immediately after I

The tryit.jssip.net server is down and isn't working. Can someone try to get it working again? I

Hello, Excuse me, by any chance do you have the example code so that you can provide me with the

Hello. When making the connection, the server responds with a 401, I have tried to make a second

Are you using an old jssip version or the newest? `sendDTMF()`[1] does not consider that state[2]

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

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