WebRTC Call Recording Issues in Desktop Dialer (JsSIP) We are working on a desktop VoIP dialer built using JsSIP and WebRTC and are facing multiple issues related to audio call recording that need to be analyzed and fixed. Issues Observed 1. Conference Call Recording Controls o User-facing Start / Stop recording buttons do not work reliably during conference calls. o Recording sometimes does not include all participants or does not stop correctly. 2. Critical: Background Audio Recording o Audio recording sometimes continues or restarts without explicit user action or permission. o This is a privacy and browser policy compliance issue and must be handled strictly via user consent and proper MediaStream cleanup. 3. Recording Failure During Multiple Calls o When handling multiple or consecutive calls, call recording intermittently fails. o In some cases, recordings are empty, corrupted, or not generated at all. Technical Stack • JsSIP • WebRTC (RTCPeerConnection) • MediaStream / MediaStreamTrack • MediaRecorder API • Desktop application (Electron or browser-based) Expected Outcome • Reliable user-controlled start/stop recording • Proper conference audio mixing • Recording works consistently across single, multiple, and conference calls • No background or unauthorized recording • Correct MediaStream and permission lifecycle management Required Expertise • Strong experience with WebRTC audio pipelines • MediaStream lifecycle and track management • Conference call audio mixing • Browser permission and privacy compliance