Design and analysis of new generation protocols for triple-play networks

Abstract

Abstract Transmission Control Protocol (TCP)'s proven stability and scalability has made it the most widely used transport layer protocol for more than twenty years. However, as multimedia applications become ubiquitous over the internet, TCP has been found incapable of meeting their requirements, which place more emphasis on timeliness than on reliability. Because of that, many multimedia applications turn to UDP as their underlying transport protocol. However, the majority of video on demand and live broadcast applications predominantly use TCP over UDP (User Datagram Protocol), due to UDP's unresponsiveness to network conditions and problems with firewalls and NATs (Network Address Translations). TCP's poor performance in delivering real-time media is due to the following reasons: 1) TCP's emphasis on reliable in-order delivery causes frame jitter that interrupt media play out. 2) TCP's coarse-grained retransmission timeout (RTO) and its back-off mechanism is detrimental to any real-time based application. In this study, we propose a new variant of TCP with an early retransmission scheme as an enhancement to make it more suitable for streaming media. We call this new protocol TCP-ER. We performed extensive NS-2 simulations to show that: 1) the early retransmission scheme can reduce the number of retransmission timeouts in a variety of network environments, which results in a considerable decrease in number of retransmission timeouts and packet delay jitter. 2) Under same network conditions, constrained streaming over TCP-ER has a considerably lower number of late packets than its normal TCP counterpart. 3) TCP-ER has a higher throughput in severely congested network conditions, whereas it stays relatively fair with typical TCP implementations (specifically TCP-SACK) as congestion gets alleviated.