Performance analysis of transmission protocols for H.265 encoder

University essay from Blekinge Tekniska Högskola/Institutionen för kommunikationssystem

Author: Akella Umesh; [2015]

Keywords: H.265; HTTP; Network Emulator; RTSP; VoD; WebRTC;


In recent years there has been a predominant increase in multimedia services such as live streaming, Video on Demand (VoD), video conferencing, videos for the learning. Streaming of high quality videos has become a challenge for service providers to enhance the user’s watching experience. The service providers cannot guarantee the perceived quality. In order to enhance the user’s expectations, it is also important to estimate the quality of video perceived by the user. There are different video streaming protocols that are used to stream from server to client. In this research, we aren’t focused on the user’s experience. We are mainly focused on the performance behavior of the protocols.

In this study, we investigate the performance of the HTTP, RTSP and WebRTC protocols when streaming is carried out for H.265 encoder. The study addresses for the objective assessment of different protocols over VoD streaming at the network and application layers. Packet loss and delay variations are altered at the network layer using network emulator NetEm when streaming from server to client. The metrics at the network layer and application layer are collected and analyzed. The video is streamed from server to a client, the quality of the video is checked by some of the users.

The research method has been carried out using an experimental testbed. The metrics such as packet counts at network layer and stream bitrate at application layer are collected for HTTP, RTSP and WebRTC protocols. Variable delays and packet losses are injected into the network to emulate real world.

Based on the results obtained, it was found at the application layer that, out of the three protocols, HTTP, RTSP and WebRTC, the stream bitrate of the video transmitted using HTTP was less when compared to the other. Hence, HTTP performs better in the application layer. At the network layer, the packet counts of the video transmitted were collected using TCP port for HTTP and UDP port for RTSP and WebRTC protocols. The performance of HTTP was found to be stable in most of the scenarios. On comparing RTSP and WebRTC, the number of packet counts collected were more in number for RTSP when compared to WebRTC. This is because, the protocol and also the streamer are using more resources to transmit the video. Hence, both the protocols RTSP and WebRTC are performing better relatively. 

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