科研成果详情

针对下一代网络中多媒体信息的网络传输方式正从UDP向TCP回归,研究了基于TCP的实时流媒体传输方法的性能.结合TCP快重传,快恢复的特性和拥塞窗口的变化规律,从微观角度分析了TCP播放缓冲区对流媒体播放质量的影响,通过进一步分析语音流的组帧传输与视频流的分帧传输过程,推导证明了在Internet上使用TCP传输实时音频的不可行,并提出了实时视频传输达到可接受的播放质量的必要条件是视频帧的发送延时应小于播放缓冲区延时与网络往返延时一半的差值.Ns2环境下的模拟实验验证了所提论断的正确性,该论断对于流媒体传输技术的发展具有指导意义和参考价值.

In view of the trend that the transmission control protocol (TCP) is increasingly used instead of the user datagram protocol (UDP) for multimedia information transmission in the next generation networks, this paper studies the performance of the real-time multimedia stream transmitting over TCP. Based on the overall consideration of the characteristics of TCP including fast retransmission, fast recovery and changing regularity of the congestion window, the impacts of TCP buffer on the playback quality are analyzed from a micro perspective. Based on a further study made on the process of assembling audio frames and disassembling video frames, the conclusion that it is unfeasible to transmit real-time audio on the Internet using TCP is proved. Also, it is presented that to transmit real-time video via TCP with the acceptable playing performance, the sending-delay of a video frame should be less than the difference between the play buffer delay and half of the round trip time (RTT). The conclusions are evaluated to be significant and referable for developing the streaming media delivery technology through the extensive simulations using the Ns2 simulator.