Multimedia streaming gateway with jitter detection

This paper investigates a novel active buffer management scheme, "Jitter Detection" (JD) for gateway-based congestion control to stream multimedia traffics in packet-switched networks. The quality of multimedia presentation can be greatly degraded due to network delay variation or jitter when transported over a packet-switched network. Jitter degrades the timing relationship among packets in a single media stream and between packets from different media streams and, hence, creates multimedia synchronization problems. Moreover, too much jitter will also degrade the performance of the streaming buffer in the client. Packets received by the client will be rendered useless if they have accumulated enough jitter. The proposed active buffer management scheme will improve the quality of service in multimedia networking by detecting and discarding packets that accumulated enough jitter, such as to maintain a high bandwidth for packets within the multimedia stream's jitter tolerance. Simulation results have shown that the proposed scheme can effectively lower the average received packet jitter and increase the goodput of the received packets when compared to random early detection (RED) and DropTail used in gateway-based congestion control. Furthermore, simulation results have also revealed that the proposed scheme can maintain the same TCP friendliness when compared to that of RED and DropTail used for multimedia streams.     

一种适用于无线网络的流媒体传输机制

为保证无线网络中多媒体数据的传输质量,提出了一种适用于无线网络的流媒体传输机制（WMTCC）。该机制通过发送探测报文区分网络拥塞丢包和链路误码随机丢包,准确判断网络的拥塞状况,实施发送速率调节,保证了流媒体服务质量(QoS)。由于准确区分出无线链路误码丢包,该机制在链路误码率较高时能维持较高的网络吞吐量。仿真实验结果显示在高误码率无线网络中,该机制可以获得更高的吞吐量和更大的拥塞窗口,并且发送速率的变化更加平滑。     

A new approach for packet loss measurement of video streaming and its application

Packet loss is of great importance as a metric that characterizes the network’s performance, and is crucial for video applications, congestion control and routing. Most of existing measurement tools can indicate the packet loss of network links instead of the actual packet loss of individual application. On the other hand, because occurrence of packet loss behavior is relatively rare and its duration is short, active measuring methods need to inject a large number of packets and run for a long time for reporting accurate estimates, which would introduce additional intrusiveness to the network and perturb user traffic. In this paper, we present a new packet loss estimation technique by making use of user_data field of video, which is less intrusive since it does not affect video playing and does not need to inject extra probing stream. It can also provide the packet loss detailed information of I,P,B frames. The accuracy of the algorithm has been evaluated with both simulations and experiments over real-world Internet paths. In addition, we analyze the video quality distortion caused by packet loss of different frame types, and a real-time video quality monitoring system is built.     

基于累积时延的流媒体传输模糊拥塞控制

为保证网络流媒体传输质量,在流媒体的传输中需要采用有效的拥塞控制策略.结合流媒体数据对时延敏感的特点,提出了一种基于累积时延的模糊拥塞控制算法,该算法在流媒体数据流传输过程中检测和跟踪其时延,在转发分组数据前,根据容忍时延阈值,丢弃超时数据包,减少不必要的带宽浪费,并且对所到达的数据流按照累积时延进行优先级分类,把全局性缓冲区和各队列的局部性缓冲区按照正常、拥塞避免和拥塞的规则划分为3个具有交叉过渡域的阶段,然后采用整体和局部相结合的拥塞控制方法,实现队列调度过程中的模糊处理,从而对网络拥塞进行有效的控制.理论分析和实验结果表明,使用基于累积时延的模糊拥塞控制算法,能有效改善流媒体的传输性能,是解决流媒体传输拥塞控制的有效途径,并能对提高网络性能起到重要作用.     

Available bandwidth estimation for the network paths with multiple tight links and bursty traffic

Available bandwidth (ABW) estimation is useful for various applications such as network management, traffic engineering, and rate-based multimedia streaming. Most of the ABW estimation methods are based on the fluid cross-traffic model. Inevitably, their estimation accuracy is limited in the network environments with bursty cross-traffic. In this paper, we apply packet trains (a series of probing packets) and a modified Ping to probe the ABW of a network path. Our proposed probing method can identify several tight links along a path and can infer their individual ABWs. The ABW estimation algorithm developed in this study, GNAPP, is also based on the fluid traffic model, but it can effectively filter out probing noise incurred in networks that carry bursty traffic. The algorithm employs not only the gaps of any two consecutive probing packets but also those of nonadjacent probing packets for ABW estimation. Thus, the number of samples for ABW estimation increases significantly without resorting to sending more probing packets and the estimation efficiency and accuracy are improved. In addition, two-stage filtering and moving averages are used in GNAPP for reducing estimation errors. Numerical results demonstrate that the estimation scheme based on GNAPP can achieve good accuracy even when the traffic is bursty and there are multiple tight links on the path being observed. Thus, it outperforms other well-known ABW estimation tools.     

Congestion Adaptive Routing in Mobile Ad Hoc Networks

Mobility, channel error, and congestion are the main causes for packet loss in mobile ad hoc networks. Reducing packet loss typically involves congestion control operating on top of a mobility and failure adaptive routing protocol at the network layer. In the current designs, routing is not congestion-adaptive. Routing may let a congestion happen which is detected by congestion control, but dealing with congestion in this reactive manner results in longer delay and unnecessary packet loss and requires significant overhead if a new route is needed. This problem becomes more visible especially in large-scale transmission of heavy traffic such as multimedia data, where congestion is more probable and the negative impact of packet loss on the service quality is of more significance. We argue that routing should not only be aware of, but also be adaptive to, network congestion. Hence, we propose a routing protocol (CRP) with such properties. Our ns-2 simulation results confirm that CRP improves the packet loss rate and end-to-end delay while enjoying significantly smaller protocol overhead and higher energy efficiency as compared to AODV and DSR     

Priority-based rate control for service differentiation and congestion control in wireless multimedia sensor networks

In wireless multimedia sensor networks (WMSNs) a sensor node may have different types of sensor which gather different kinds of data. To support quality of service (QoS) requirements for multimedia applications having a reliable and fair transport protocol is necessary. One of the main objectives of the transport layer in WMSNs is congestion control. We observe that the information provided may have different levels of importance and argue that sensor networks should be willing to spend more resources in disseminating packets carrying more important information. Some applications of WMSNs may need to send real time traffic toward the sink node. This real time traffic requires low latency and high reliability so that immediate remedial and defensive actions can be taken when needed. Therefore, similar to wired networks, service differentiation in wireless sensor networks is also an important issue. We present a priority-based rate control mechanism for congestion control and service differentiation in WMSNs. We distinguish high priority real time traffic from low priority non-real time traffic, and service the input traffic based on its priority. Simulation results confirm the superior performance of the proposed model with respect to delays, delay variation and loss probability.     

自适应调整TCP阻塞控制窗口算法研究

提出了新的TCP速率调整算法.根据边缘路由器缓冲区中的输入数据报和输出数据报的变化,得到合理阻塞控制窗口,直接通过明确阻塞标记数据报返回到发送终端,从而改变了TCP发送速率.通过对仿真结果的分析,新算法可以明显地控制TCP的业务量,限制边缘路由器的队列的拥塞,大大降低数据报的丢失率,从而提高TCP的延迟性能和带宽分配的公平性.     

MTCP: scalable TCP-like congestion control for reliable multicast

We present MTCP, a congestion control scheme for large-scale reliable multicast. Congestion control for reliable multicast is important, because of its wide applications in multimedia and collaborative computing, yet non-trivial, because of the potentially large number of receivers involved. Many schemes have been proposed to handle the recovery of lost packets in a scalable manner, but there is little work on the design and implementation of congestion control schemes for reliable multicast. We propose new techniques that can effectively handle instances of congestion occurring simultaneously at various parts of a multicast tree.Our protocol incorporates several novel features: (1) hierarchical congestion status reports that distribute the load of processing feedback from all receivers across the multicast group, (2) the relative time delay concept which overcomes the difficulty of estimating round-trip times in tree-based multicast environments, (3) window-based control that prevents the sender from transmitting faster than packets leave the bottleneck link on the multicast path through which the sender's traffic flows, (4) a retransmission window that regulates the flow of repair packets to prevent local recovery from causing congestion, and (5) a selective acknowledgment scheme that prevents independent (i.e., non-congestion-related) packet loss from reducing the sender's transmission rate. We have implemented MTCP both on UDP in SunOS 5.6 and on the simulator ns, and we have conducted extensive Internet experiments and simulation to test the scalability and inter-fairness properties of the protocol. The encouraging results we have obtained support our confidence that TCP-like congestion control for large-scale reliable multicast is within our grasp.     

加密环境中的无线TCP性能优化

TCP是一种广泛使用的可靠传输协议,它的设计基于拥塞是引起丢包主要原因的假设。但在无线网络中,丢包通常是传输差错的缘故,这导致了TCP传输性能的急剧下降。现有不少无线TCP传输性能优化方法要求中间节点能够获得TCP连接信息,因此不适用于数据流被加密的情况。文中提出的丢包识别机制可应用于加密环境,它通过TCP发送端判断丢包原因并采取相应措施来提高传输速率,仿真结果证明该方法是有效的、鲁棒的,在局域网和广域网环境中都能明显提高TCP端到端传输性能。     

An admission control scheme for end-to-end statistical QoS provision in IP networks

This paper presents a distributed and scalable admission control scheme to provide end-to-end statistical QoS guarantees in IP network.The basic idea of the scheme is that the ingress routers make admission control decisions according to the network status information obtained by sending probing packets along the selected routing path.Each router passively monitors the arriving traffic and marks the probing packets with its network status.The performance of the presented scheme is evaluated with a variety of traffic models,QoS metrics and network topologies,The simulation results show that the proposed scheme can accurately control the admissible region and effectively improve the utilization of network resource.     

Providing MAC QoS for multimedia traffic in 802.11e based multi-hop ad hoc wireless networks

Ad hoc wireless networks with their widespread deployment, now need to support applications that generate multimedia and real-time traffic. Video, audio, real-time voice over IP, and other multimedia applications require the network to provide guarantees on the Quality of Service (QoS) of the connection. The 802.11e Medium Access Control (MAC) protocol was proposed with the aim of providing QoS support at the MAC layer. The 802.11e performs well in wireless LANs due to the presence of Access Points (APs), but in ad hoc networks, especially multi-hop ones, it is still incapable of supporting multimedia traffic.One of the most important QoS parameters for multimedia and real-time traffic is delay. Our primary goal is to reduce the end-to-end delay, thereby improving the Packet Delivery Ratio of multimedia traffic, that is, the proportion of packets that reach the destination within the deadline, in 802.11e based multi-hop ad hoc wireless networks.Our contribution is threefold: first we propose dynamic ReAllocative Priority (ReAP) scheme, wherein the priorities of packets in the MAC queues are not fixed, but keep changing dynamically. We use the laxity and the hop length information to decide the priority of the packet. ReAP improves the PDR by over 28% in comparison with 802.11e, especially under heavy loads. Second, we introduce Adaptive-TXOP (A-TXOP), where transmission opportunity (TXOP) is the time interval during which a node has the right to initiate transmissions. This scheme reduces the delay of video traffic by reducing the number of channel accesses required to transmit large video frames. It involves modifying the TXOP interval dynamically based on the packets in the queue, so that fragments of the same packet are sent in the same TXOP interval. A-TXOP is implemented over ReAP to further improve the performance of video traffic. ReAP with A-TXOP helps in reducing the delay of video traffic by over 27% and further improves the quality of video in comparison with ReAP without A-TXOP. Finally, we have TXOP-sharing, which is aimed at reducing the delay of voice traffic. It involves using the TXOP to transmit to multiple receivers, in order to utilize the TXOP interval fully. It reduces the number of contentions to the channel and thereby reduces the delay of voice traffic by over 14%. A-TXOP is implemented over ReAP to further improve the performance of voice traffic. The three schemes (ReAP, A-TXOP, and TXOP-sharing) work together to improve the performance of multimedia traffic in 802.11e based multi-hop ad hoc wireless networks.     

An integrated protocol stack for efficient resources management in wireless networks

The area of ATM multihop radio networks has recently become an issue of interest especially with the growing interest in portable multimedia units. Multimedia communication requires a deterministic delivery of isochronous traffic with predefined QoS requirements. In this paper we propose an integrated protocol stack which provides deterministic delivery bounds and efficiently utilizes channel backwidth. By integrating the access, routing and congestion control protocols, a solution is provided which improves bandwidth utilization, maintains shortest route packet delivery and leads to congestion avoidance. A tractable approximate analytical model is developed and verified for networks with finite storage capacity. For large multihop networks the analytical solution is complemented by simulation. Evaluation shows that the proposed protocol integration yields a significant reduction in end-to-end delay, which together with bounded access time at the channel access level, provide two essential features for the support of isochronous traffic in multihop wireless environments.     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

A link layer adaptive pacing scheme for improving throughput of transport protocols in wireless mesh networks

In this paper, we study the performance of a static multihop wireless network, specifically that of the backhaul network of a two-tier Wireless Mesh Network (WMN) operating on IEEE 802.11 Medium Access Control (MAC) protocol. The performance of an IEEE 802.11 based backhaul network is greatly affected by the MAC contention and congestion in the network. If the sources pump data into the network than can be supported, loss rate increases due to MAC contention and congestion in the network. This also leads to the problem of unfairness among flows. In this paper, we propose a Link Layer Adaptive Pacing (LLAP) scheme that adaptively controls the offered load into the network. This improves the performance of higher layer protocols without any modifications to them. Our LLAP scheme estimates the four hop transmission delay in the network path without incurring any additional overhead (Control packets) and accordingly paces the packet transmissions to reduce MAC contentions in the network. We implement the LLAP scheme in ns-2.29 network simulator and extensively study its performance for both User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) traffic in different network scenarios. In all the cases, our scheme shows a significant improvement in the performance of both UDP and TCP traffic.     

基于ECN的实时多媒体的实时多媒体流TCP友好控制

随着Internet上多媒体应用的日益增加,实时多媒体流的TCP友好控制成为当前的研究热点。该机制基于RTP/RTCP协议,以ECN的方式将拥塞状况通知发送端,在路由器中采用RED队列管理策略,在端主机采用TCP友好的速率调节机制。ECNBCC机制具有TCP友好的特性并且可以对网络早期拥塞作出反应,从而降低丢包率和网络延时,该机制也可用于无线网络多媒体流的拥塞控制。     

Dejittering in the transport of MPEG-2 and MPEG-4 video

When multimedia information is transported over a packet-switched network, the quality of presentation can be degraded due to network delay variation or jitter. This paper presents a dejittering scheme that can be used in the transport of MPEG-4 and MPEG-2 video to absorb any introduced network jitter, thus preserving the presentation quality of transported media streams. The dejittering scheme is based on the statistical approximation of delay variation in the arrival times of video packets carrying encoded clock reference values and a filtering and re-stamping mechanism. In addition, a brief overview of the MPEG-4 system is presented.     

一种基于路由协议的拥塞控制策略

提出了一种基于OSPF路由协议的拥塞控制策略。利用OSPF协议的链路状态更新报文（LSA）中的空闲位,增加路由器的拥塞状态和流量状态的描述,随LSA报文的传播将路由器的拥塞情况告知其他路由器,利用OSPF的快速收敛及时得知网络拥塞状况并进行早期的拥塞避免。仿真模拟表明,该方案能够控制网络拥塞,减小延迟,达到网络负载平衡。     

一种报文大小可变的TFRC改进算法

基于Internet的实时多媒体数据传输是一种报文发送速率固定,报文大小变化的应用。该文分析了这类应用对TFRC的影响,通过对TFRC协议的扩展,提出了一种支持报文大小可变应用的改进TFRC拥塞控制算法。这种算法在接收方采用了对报文数量进行加权的方法来计算丢失事件率以支持报文大小变化的应用。同时在网络仿真器ns2中实现了这种改进算法。仿真实验表明：这种改进算法能够支持报文大小变化,报文发送速率固定的应用,并且具有TCP友好性,与TCP相比具有较平缓的流量抖动。