RRB-SIMD: RTP Rate-Based SIMD Protocol for Media Streaming Applications over the Internet

Congestion control is vital in the streaming of a video sequence or clip, as network traffic varies unpredictably requiring constant adjustment of the transmission rate. Standard TCP-Friendly Rate Control (TFRC) wastes bandwidth and may react to congestion only when packet loss has already occurred. This paper presents a unicast transport protocol named RRB-SIMD for video streaming over the Internet, that provides better quality of service (QoS) support than the TCP-friendly rate control. RRB-SIMD operates on top of Real-time Transport Protocol (RTP) and takes advantage of Real-time Transport Control Protocol (RTCP) reports to multiplicatively decrease the transmission rate in response to congestion and quadratically increase the transmission rate in the absence of congestion. Since packet loss is not a reliable indicator of congestion, RB-SIMD uses in addition the cumulative jitter as a control criterion to detect incipient congestion prior to loss of a packet. The cumulative jitter scheme is reinforced with a delay factor that measures on a per round basis the buffer occupancy at the bottleneck path between the sender and the receiver. This is done to reduce the risk of unnecessary decrease of the transmission rate every time that incipient congestion is reported through the cumulative jitter scheme. The performance evaluation results using both network-related metrics and video quality measurement shows that RRB-SIMD exhibits a better performance with respect to lost frames ratio, delay and cumulative jitter, and hence an improved quality display than the standard TFRC.