Latency in WebRTC refers to the time delay between data transmission from one endpoint to another, impacting real-time communication experiences. Minimizing latency is crucial for applications like video conferencing and live streaming. Factors contributing to internet latency include network congestion, packet loss, and jitter. Compared to other streaming protocols like HLS, RTMP, MPEG-DASH, and SRT, WebRTC often excels in minimizing latency due to its design for real-time communication. VideoSDK is a comprehensive solution that offers low-latency streaming capabilities, adaptive bitrate streaming, and real-time video optimization, making it an effective ally for developers aiming to mitigate latency challenges in WebRTC. By integrating VideoSDK into their applications, developers can unlock the full potential of real-time audio-video communication, providing users with seamless and immersive experiences.