How to run sip over websocket?

Question

FreeSWITCH

Hello. The situation is this. Recently we set the task to make it possible to call operators from CRM, i.e. straight from the browser. I dug in this thread, came to JsSIP library and OverSIP proxy. The proxy had to be set because our VoIP provider (on the side of which the virtual PBX was created) mango-office does not have the sip over websocket functionality, and does not want to add it to any. In general, the server was installed, configured, launched, I try to call from the browser. The call goes on, but when I pick up the phone on the phone, the call is cut off. In console: Failed to set remote answer sdp: Called with SDP without DTLS fingerprint. Those. the provider's media server does not support this DTLS. After that I decided to try FreeSwitch to use it as a sip proxy and media proxy. Created a getway, it normally connected to the provider. But when I try to log in via JsSIP, the friswitch says that the account was not found. As I understand it, it uses its own database of numbers, and does not want to connect to the provider's database (i.e. proxy). Please tell me how you can configure FreeSwitch so that it works as a proxy, forwarding SIP traffic through itself to the side of the SIP provider, and also transcoding Media traffic so that the browser can make calls normally? Well, or maybe there is some other alternative (not Flashphone). I will be very grateful. and was also engaged in transcoding Media traffic so that the browser could make calls normally? Well, or maybe there is some other alternative (not Flashphone). I will be very grateful. and was also engaged in transcoding Media traffic so that the browser could make calls normally? Well, or maybe there is some other alternative (not Flashphone). I will be very grateful.

Answer 1

FS user profiles are in conf/directory/default (valid for vanilla config)