Supported codecs right now include G.711, G.722, iLBC, and iSAC. Video uses the VP8 codec.
Media is sent and received using RTP with a preference for SRTP.
STUN and ICE are used to determine the type of NAT the browser is behind and how it behaves. If direct communication is not possible a TURN server can be used to relay the media traffic.
JSEP (specification can be found here) is used as the API to process and create session descriptions. The session descriptions are formatted as SDP and contain codec details, ICE details, and more.
If you would like to view the API the latest specification is available here.
As WebSocket has become the current method of exchanging session information I propose implementing a WebSocket server within Asterisk itself. This would allow WebRTC users instant access to all the protocols Asterisk has to offer with minimal work required on their side.