Submit Session Network Stats

Accept a client-side WebRTC quality rollup for a session.

No auth required — session_id acts as the authorization, same as join endpoints. Called from the browser at end-of-call (and periodically as a backup) with a pre-aggregated rollup of inbound/outbound RTP audio stats sampled via RTCPeerConnection.getStats().

Last write wins: each call overwrites the stored rollup. The client is expected to send cumulative/aggregate values, not deltas.