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When it comes to debugging SIP messages on WebRTC, there is a simple but hidden way to see all the messages passed between the server and the browser in plain text.
Updated WebRTC SIP Trace Capture (markdown)
Initial Home page
Destroyed Step 1 (Raspbian Setup) (markdown)
Destroyed Install and configure Asterisk 13 or 16 from source (markdown)
Destroyed Browser Phone (markdown)
Updated Browser Phone (markdown)
A fully featured browser based WebRTC SIP phone for Asterisk
Updated Home (markdown)
How to install Asterisk 13 or 16 and configure chan_sip or chan_pjsip for the browser phone
How to create a self signed Certificate Authority, and server certificate from that Root CA
How to setup and configure Raspbian OS with some initial tools and services