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