JsSIP

Guys, I think the problem is due to the audio stream not being assigned to the HTML audio object.

Has anyone solved this problem? It's really urgent 在2023年5月23日星期二 UTC+8 11:45:32<Cheng Lin Yu

You can just roll your own, lots of open source options. From: <js...@googlegroups.com> on

当频繁刷新浏览器去重新连接通信命令服务器时，接收到电话通话，这个时候经常会遇到服务中断websocket断开连接的情况，导致无法正常去answer接听通话，知道是什么原因导致的这个结果吗，或者如何规避

The JWT is sent in the Authorization SIP header of the messages. El mié, 5 abr 2023 a las 17:32, Mike

Does Jssip work with react native anyway? who could help with that? I read some comments here in the

if local reject the calling,`faild` event's callback function return { message:null } but in the

u need to handle get permission first if u want i can help you i had this issue before On Thu, Apr 13

Hello All, I just wanted to check if we can declare an optional parameter to get information from

Morning folks. A quick question about the setter for JsSIP UA for authorization_jwt. If I call that

Hello, My asterisk got a log like this below : "NOTICE [22152] [C-0000000c]: res_rtp_asterisk.c:

Looking for some ideas. My application has long lived calls, where we hit RTCSession runSessionTimer(

Hello! I'm trying to place a one way video call. My CallOptions are: { rtcConstraints: {

Sorry, jssip unmute is working fine, the problem seems to be with the "react-native-callkeep

Hello! I am using the JSSip (Version 3.10.0) library on my project. And I'm trying to set up an

Hello Ronald, So you're able to run JsSIP in RN 0.70.7? Any plans on publishing a PR to react-

Are you behind NAT? You might need to have STUN/TURN for audio On Mon, Mar 27, 2023 at 12:29 PM

Hi Alex, Thank you for that! Have you any idea how https://tryit.jssip.net/ on Chrome manages to

During the call, we try to refer it, we get RequestSucceded and Accepted in response, but the user to

Hello! How can i add a optional field with call statistics on the 200 response when i receive an

Reloading the page close the wesoket and the active call will be down, so i do not think is possible.

Hello! i've been using this library with Asterisk 18 and we noticed a peculiar issue, if this

Hi All, I use Jssip connect with legacy SIP phone by using rtpengine to be the media proxy. 1. jssip

Hello! The newDTMF event is fired only for local sent DTMF's. If remote party sends DTMF during

Absolute Bill! Thanks again. On Friday, February 3, 2023 at 3:20:47 PM UTC-8 bker...@heavylogic.com

Hello! I need a help How add DTMF in this situation? Init: function initUA() { window.ua = new JsSIP.

Hi there, in a project, I just wanted to listen the other side without having to give access or even

when we create a release apk it doesnot register

hello everyone. I want to send an audio file to the other end of the call this is what i try, const

sip debug <— SIP read from UDP:10.11.63.77:5068 —> INVITE sip:20...@10.12.39.223;user=phone SIP/