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The world's first HTML5 SIP client (WebRTC)

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i use asterisk 13 how can i use sipml5 with realtime database mysql (i can register and make a call with static sip.conf extension but whene i use realtime dtabase...

User 1 makes a call to User2. User1 is in ringing state even after User2 attended the call.

Uncaught ReferenceError: mozRTCSessionDescription is not defined Uncaught ReferenceError: SIPml is not defined

Hello I need one quick help, When I use the call queue to connect a call, I can only hear audio from one side instead of both sides. Here, I...

Hi again. Has anyone tried to make changes to enable the api to work with mobile browsers - specifically Safari 15? Any tips pls?

When we dial a number, we also want to include the company's display name, as shown in the mobile version. But the display name "**o_uri_to"** is always null; how can...

Everything works fine till holding the call, voice from other side reach to panel , but voice from panel is not does reach to device android/IOS