Skype is currently the most popular VoIP software in the world. Existing research has indicated that Skype can provide better voice/video quality in adverse network conditions, when compared with other counterpart products (e.g. X-Lite and Windows Live Messenger). This is possibly due to Skype's built-in adaptation/control mechanism which can adapt its voice/video sender bit rate automatically in order to ease network congestion. Due to Skype's proprietary nature, details of adaptation and control mechanism within Skype are unknown.
This project aims to investigate how Skype adapts its video send bit rate in react to network congestions and what Skype's performance are (in terms of video Quality of Experience, or Mean Opinion Score) under different network conditions. The project mainly contains the following five stages.
1. Literature review on existing work on performance analysis for Skype on video quality, the latest video codec's used in Skype.
2. Setup a VoIP testbed which includes video sender, Network emulator, and video receiver. Nistnet or qdisc tools can be used for network emulation for creating different network conditions (e.g. packet loss, bandwidth). Wireshark can be used to capture and analyse the VoIP traffic.
3. Design and carry out experiments to investigate how Skype adapt to network condition changes in terms of video send bit rate, interpacket-gap and packet size.
4 User behaviour analyses for Skype, investigate how users react to video call quality degration, to which extent (in terms of packet loss), user will not tolerate the video call quality by dropping the call.
5. Thesis/research paper writing
Familar with Linux and Linux shell script, e.g. awk.