The Impact of Bursty Packet Loss on Audio Quality in WebRTC
Ensuring high-quality audio in WebRTC encounters a pivotal challenge amidst less than ideal network conditions, predominantly driven by the burstiness of packet loss. This phenomenon is prevalent in congested networks, areas with low mobile coverage, and public Wi-Fi setups. Within the WebRTC framework, an array of strategies exists to mitigate packet loss, yet their efficacy varies depending on the specific network dynamics. Among the most prevalent techniques are: OPUS Forward Error Correction (FEC): Each audio packet incorporates low-bitrate data from preceding packet, facilitating potential recovery in the event of a single packet lost. Packet Retransmissions: Leveraging standard NACK/RTX mechanisms, the receiver requests retransmission upon detecting packet sequence gaps. Packet Duplication: Sending multiple instances of the same packet aims to compensate for potential losses. It is like sending preemptive retransmissions to mitigate the impact of potential packet loss. Redund