Evaluation of skype's audio and video quality: a technical analysis of skype's audio and video quality, including codec selection, bitrate, and packetization

Recently, video telephony has gained popularity and is now commonly used by end users. However, research on the effects of video calls on networks and user Quality-of-Experience (QoE) in various network scenarios is somewhat limited. The rate control and video quality of Skype video chats are examined in this work. First, we use a controlled network testbed to observe how Skype video conversations behave. We see how Skype modifies its rates, FEC redundancy, and video quality by altering the packet loss rate, bandwidth, and propagation delay. We discover that Skype can effectively use the available network bandwidth and is resilient to slight packet losses and propagation delays. Additionally, we discover that Skype uses an excessively forceful FEC protection approach. We create rate control, FEC, and video quality models for Skype based on the measurement data. We use numerical analysis to investigate Skype's network implications by extrapolating from the models. We show that user back-offs in response to quality degradation function as a useful user-level rate control mechanism. Additionally, we demonstrate that Skype video conversations are TCP-friendly and swiftly adapt to network congestion. Two important performance metrics for video and audio conferences in streaming video call services are Quality of Service (QoS) and Quality of Experience (QoE). Given that Skype functions and develops with the goal of providing the highest quality service possible across all operational environments, it is critical to look into how Skype has changed to accommodate this cross-platform scope, what encoding enhancements have been made over the years, and how effective these changes are both objectively and subjectively.