webrtc

Notes on the telephony layer of a conference-bridge platform built with a small team. FreeSWITCH in Docker, SIP plus WebRTC, a dialplan that curls into the API on every call, and a recording pipeline driven by post-process hooks.

Updated OPUS codec patch now supports Asterisk 11.11.0 and 12.2.5, tested with VP8 pass-through.

Tonight, I tried two WebRTC clients (JsSIP and sipML5) with Asterisk 11. I was able to get both of them working for echo test calls with the ulaw (g711u) codec. However, when I called from WebRTC to the SIP softphone, there was only one-way audio.