call-center/docs/roadmap/09-track14-browser-softphone-webrtc.md

Wave 2 / Track 14 - Browser Softphone (WebRTC, QA First)

Status: Implemented, QA validation pending

Track 14 adds the first browser-native operator media path on top of the accepted Track 11 and Track 12 voice baseline.

Goal

Keep the accepted bridge-backed live-call model and move the operator media path into the browser for the first QA slice.

Implemented direction:

• direct Asterisk WSS
• inbound MVP only
• browser softphone coexists with MicroSIP/Zoiper
• existing bridge claim/hangup/recent-calls remains authoritative

Deliverables

1. Bridge:
   • additive route: /asterisk/browser-softphone/config
   • additive status fields: webrtc_enabled, webrtc_ws_url
2. Operator shell:
   • Browser Softphone block
   • register/disconnect state
   • incoming call banner
   • Answer, Hang up, Mute/Unmute
   • microphone and speaker selectors
3. Asterisk lab assets:
   • deployment/asterisk/http.conf
   • deployment/asterisk/rtp.conf
   • updated deployment/asterisk/pjsip.conf

Boundaries

Track 14 does not introduce:

• production cutover
• ARI
• browser-only mandatory mode
• DB schema changes
• WebRTC backend service outside Asterisk

The accepted voice system of record remains:

• bridge live/recent calls
• bridge action log
• recording import and supervisor playback