A client-driven media synchronization mechanism for RTP packet-based video streaming

Media synchronization is used to correctly playback a video stream with its associated audio. To support synchronization between video and audio streams transported over IP networks, an RTP/RTCP protocol suite is usually employed. In conventional server-driven media synchronization, the server needs to periodically transmit an RTCP sender report (SR) packet to provide the client with a UTC time in NTP format corresponding to the RTP timestamp carried by each RTP packet. In this paper, we propose a precise client-driven media synchronization mechanism for an RTP packet-based multimedia streaming service. In the proposed method, the server does not need to send any RTCP SR packets for synchronization. Instead, the client device derives the precise normal play time (NPT) for each video and audio stream from the received RTP packets containing an RTP timestamp. Simulations show that the proposed client-driven synchronization method can provide accurate media synchronization without employing an RTCP SR packet and accordingly reduce the initial synchronization delay, the processing complexity at the client device, the number of required user datagram protocol ports, and the amount of control traffic injected into the network.     

基于RTP的MPEG-4视频监控系统的设计

简述了MPEG-4标准的特性和RTP/RTCP协议的概念,详细说明了将MPEG-4码流映射到RTP净荷的方法,最后在此基础上提出了一种基于网络的远程视频监控系统的设计方案.该方案首先给出了系统的设计框架,然后分别阐述了发送端和接收端的具体实现流程.实验结果表明,采用RTP/RTCP协议传输MPEG-4码流能有效利用网络带宽,并且图像质量有较大的提高.     

分布式多媒体交互系统音频／视频流传输的设计与实现

本文简要介绍了RTP／RTCP协议和IP Multicast技术,结合分布式多媒体交互系统中音频、视频在网络上的多点实时传输的实现,详细地介绍了RTP／RTCP协议在音频和视频数据流传输及传输控制中的实现过程,以及利用IP Multicast技术实现多媒体多点传输过程中多用户之间的信息管理。     

基于丢包率的多播网络拓扑推断算法

在已有的网络拓扑推断算法的基础上,提出一种改进的基于丢包率的多播网络拓扑推断算法。依据各层次节点及接收节点的层次信息,利用海明距离和节点接收的探测包数目,能够同时推断网络拓扑结构和链路丢包率,并且在计算链路丢包率的估计值时动态地调整拓扑推断的判决门限值,从而提高了推断的准确性。仿真实验证明,与现有算法相比,该算法具有更好的性能。     

基于丢包率的多播网络拓扑推断算法

在分析现有的网络拓扑推断算法的基础上,提出一种改进的基于丢包率的多播网络拓扑推断算法。结合接收节点的层次信息、汉明距离及节点接收的探测包数量,能够同时推断网络拓扑结构和链路丢包率,并根据链路丢包率的估计值动态地调整拓扑推断的判决门限值,提高了推断的准确性。仿真实验证明,与现有算法相比,该算法具有更好的性能。     

无线传感器网络的能量均衡路由算法

在分析现有的网络拓扑推断算法的基础上,提出一种改进的基于丢包率的多播网络拓扑推断算法。结合接收节点的层次信息、汉明距离及节点接收的探测包数量,能够同时推断网络拓扑结构和链路丢包率,并根据链路丢包率的估计值动态地调整拓扑推断的判决门限值,提高了推断的准确性。仿真实验证明,与现有算法相比,该算法具有更好的性能。     

远程多点多媒体实时传输系统的设计与实现

介绍了 IP组播技术、实时传输和实时控制协议 ( RTP/ RTCP) ,以及利用 Winsock规范开发远程多点多媒体实时传输系统 ,并进一步地说明了其它各部分的组成、性能及实现了多点音频、视频多媒体信息在网络上实时多点传输的方法。     

LION: Layered Overlay Multicast With Network Coding

Recent advances in information theory show that the throughput of a multicast session can be improved using network coding. In overlay networks, the available bandwidth between sender and different receivers are different. In this paper, we propose a solution to improve the throughput of an overlay multicast session with heterogeneous receivers by organizing the receivers into layered data distribution meshes and sending substreams to each mesh using layered coding. Our solutions utilize alternative paths and network coding in each mesh. We first formulate the problem into a mathematical programming, whose optimal solution requires global information. We therefore present a distributed heuristic algorithm. The heuristic progressively organizes the receivers into layered meshes. Each receiver can subscribe to a proper number of meshes to maximize its throughput by fully utilizing its available bandwidth. The benefits of organizing the topology into layered mesh and using network coding are demonstrated through extensive simulations. Numerical results indicate that the average throughput of a multicast session is significantly improved (up to 50% to 60%) with only slightly higher delay and network resource consumption.     

A sensitive network jitter measurement for covert timing channels over interactive traffic

In order to reflect the network transmission quality, some network state feedback mechanisms are provided in the network protocol. In the RTP, the jitter of the packet transmission delay is fed back through the jitter field in the RTCP packet. This feedback value is a very important reference data when the covert timing channel is established. However, the sending frequency of the RTCP packet is low and the feedback value of the RTCP packet are only the jitter value of the last RTP packet associated with this RTCP packet when it is sent. Therefore, the jitter feedback mechanism in the existing RTCP protocol has the problem of lack of feedback on the network state during the period between two RTCP data packets. As a result, the feedback value is highly susceptible to extreme values, which prevents it from providing an accurate numerical reference for establishing covert channels. Therefore, in this paper, a buffer was established between the last RTCP packet and the current RTCP packet. And we choose to set the interval is n RTP packets and record the corresponding position jitter value in the buffer. The data in the buffer is averaged, and the mean value is weighted and averaged with the jitter value of the current RTCP packet as a new jitter feedback value. The effect of the extreme value on the feedback value is reduced, thereby it contribute to the improvement of the feedback energy for the state of the network. In addition, the bit error rate generated by establishing a simple covert timing channel for data transmission under different network conditions is compared with the change of two jitter feedback values. It is verified that there is a positive correlation between the feedback value of the new feedback mode and the error rate. through the comparison It is verified that the new feedback method can provide a more accurate reference for the establishment of covert channels.