Packet Permutation: A Robust Transmission Technique for Continuous Media Streaming Over the Internet

With the growing popularity of the Internet, there is an increasing demand to deliver continuous media (CM) streams over the Internet. However, packets may be damaged or lost during transmission over the current Internet. In particular, periodic network overloads often result in bursty packet losses, degrading the perceptual quality of CM streaming. In this paper, we focus on reducing the impact of this bursty loss behavior. We propose a novel robust end-to-end transmission scheme, referred to as packet permutation (PP), to deliver pre-compressed continuous media streams over the Internet. At the server side, PP permutes, prior to transmission, the normal packet delivery sequence of CM streams in a specific way. The packets are then re-permuted at the receiver side before they are presented to the application. In this way, the probability of losing a large number of packets within each CM frame can be significantly reduced. To validate the effectiveness of PP, a series of trace-driven simulations are conducted. Our results show that for a given quality of service (QoS) requirement of CM streaming, PP greatly reduces the overhead required by traditional error control schemes, such as forward error correction (FEC) and feedback/retransmission-based schemes.     

用于分层移动IPv6网络拥塞控制的新机制

针对在IPv6无线网络切换过程中, 短暂的链路连接断开会导致反复丢包; 同时在切换过程中, 由于带宽的改变, 在新的接入点就有可能发生丢包或资源浪费以及拥塞。提出一种以TCP协议为基础的路径损耗确认（TCP-PLACK）机制来代替TCP选择确认机制（TCP-SACK）。每当一个TCP接收方在断开或切换后而连接到一个新的接入点时, 上述的TCP-PLACK机制就会发送一个特殊的确认, 其中包含有在新接入点丢包的详细信息和可用带宽。收到这个确认, 发送方会重新发送丢失的包, 并在新接入点根据带宽的可用性调整发送速率。实验结果表明, 该机制有利于改善IPv6网络切换过程中的丢失恢复和速率控制问题。     

Numerical analysis of the power saving with a bursty traffic model in LTE-Advanced networks

We analyze the power saving operation called Discontinuous Reception (DRX) with a novel bursty packet arrival model in 3GPP Long Term Evolution (LTE)-Advanced networks. Typical analytical studies on the power saving operations in wireless networks have been carried out under the assumption that an expectation of exponentially distributed packet arrival intervals stays unchanged. However, practical packet arrival rate may change depending on time or typical Internet services may incur bursty packet arrivals. In either case, we need to evaluate the performance of the DRX operation. For this purpose, we develop a more realistic traffic arrival model considering packets may arrive in a bursty manner under the DRX operation. We, then, analyze the performance of the DRX operation in terms of power saving efficiency and average queuing delay, respectively. The analytical results are validated via comparisons with simulation results.     

Priority and delay aware packet management framework for real-time video transport over 802.11e WLANs

The rigid delay constraint is one of the most challenging issues in real-time video delivery over wireless networks. The expired video packets will become useless for the decoding and display even if they are received correctly at the receiver. Because the significance of each video packet is different, the schedulers have to take into account not only the urgency of the packet but also its importance in the real-time video applications. However, the existing QoS-based IEEE 802.11e MAC protocol leaves the urgency and the importance of video packets out of consideration. This paper proposes a Priority and Delay Aware Packet Management Framework (PDA-PMF) to improve the transmission quality of real-time video streaming over IEEE 802.11e WLANs. In the MAC layer, this framework estimates the delay of each video packet. Subsequently, video packets are sent or dropped according to both the significance of the video packets and the estimation value of the delay. Simulation results show that the proposed scheme can not only reduce the packet losses, but also protect the more important video packets, so as to improve the received video quality effectively.     

Playout buffering of speech packets based on a quality maximization approach

To combat jitter problems in voice streaming over packet networks, playout buffering algorithms are used at the receiver side. Most of the proposed solutions rely on two main operations: prediction of delay statistics for future packets; setting of the end-to-end delay so as to limit or avoid packet losses. In recent years, a new approach has been presented, which is based on using a quality model to evaluate the impact of both packet loss and delay on the voice quality. Such a model is used to find the buffer setting that maximizes the expected quality. In this paper, we present a playout buffering algorithm whose main contribution is the extension of the new quality-based approach to the case of voice communications affected by bursty packet losses. This work is motivated by two main considerations: most of IP telephony applications are characterized by bursty losses instead of random ones; the human perception of the speech quality is significantly affected by the temporal correlation of losses. To this purpose, we make use of the extensions proposed in the ETSI Tiphon for the ITU-T E-Model so as to incorporate the effects of loss burstiness on the perceived quality. The resulting playout algorithm estimates the characteristics of the loss process varying the end-to-end delay, weights the loss and the delay effects on the perceived quality, and maximizes the overall quality to find the optimal setting for the playout buffer. The experimental results prove the effectiveness of the proposed technique.     

基于集成FEC的可靠多目广播及性能评价

对点对点通信，数据的传输常使用自动重复求技术，而前向纠错多用于半可靠实时传输。然而，ＡＲＱ用于多目广播的性能不佳。     

A multirate MAC protocol for reliable multicast in multihop wireless networks

Many multicast applications, such as audio/video streaming, file sharing or emergency reporting, are becoming quite common in wireless mobile environment, through the widespread deployment of 802.11-based wireless networks. However, despite the growing interest in the above applications, the current IEEE 802.11 standard does not offer any medium access control (MAC) layer support to the efficient and reliable provision of multicast services. It does not provide any MAC-layer recovery mechanism for unsuccessful multicast transmissions. Consequently, lost frames cannot be detected, hence retransmitted, causing a significant quality of service degradation. In addition, 802.11 multicast traffic is sent at the basic data rate, often resulting in severe throughput reduction.In this work, we address these issues by presenting a reliable multicast MAC protocol for wireless multihop networks, which is coupled with a lightweight rate adaptation scheme. Simulation results show that our schemes provide high packet delivery ratio and when compared with other state-of-the-art solutions, they also provide reduced control overhead and data delivery delay.     

MPRN多播数据可靠传送协议的设计与研究

在移动分组无线网环境中,多播路由协议起着非常重要的作用。由于MPRN受到节点移动和带宽的限制,已有多播路由协议的可靠性不高。文章提出基于Gossip的传输协议(GBTP),旨在提高多播路由协议的数据分组递交率。在MFGRP上进行了GBTP的模拟和性能分析,研究了GBTP对分组递交成功率的影响。仿真实验显示GBTP协议可以有效地提高MFGRP协议的分组递交率。     

A hybrid scheme for multicast authentication over lossy networks

For multicast communication, authentication is a challenging problem, since it requires that a large number of recipients must verify the data originator. Many of multicast applications are running over IP networks, in which several packet losses could occur. Therefore, multicast authentication protocols must resist packet loss. Other requirements of multicast authentication protocols are: to perform authentication in real-time and to have low communication and computation overheads. In the present paper, a hybrid scheme for authenticating real-time data applications, in which low delay at the sender is acceptable, is proposed. In order to provide authentication, the proposed scheme uses both public key signature and hash functions. It is based on the idea of dividing the stream into blocks of m packets. Then a chain of hashes is used to link each packet to the one preceding it. In order to resist packet loss, the hash of each packet is appended to another place in the stream. Finally, the first packet is signed. The proposed scheme resists packet loss and is joinable at any point. The proposed scheme is compared to other multicast authentication protocols. The comparison shows that the proposed scheme has the following advantages: first, it has low computation and communication overheads. Second, it has reasonable buffer requirements. Third, the proposed scheme has a low delay at the sender side and no delay at the receiver side, assuming no loss occurs. Finally, its latency equals to zero, assuming no loss occurs.     

一种用于无线局域网可伸缩视频传输的跨层结构与SBED算法*

为了增强802.11e无线局域网上的视频传输质量,提出了一种跨层结构与自适应映射算法。将H.264可伸缩视频编码（SVC）的层次信息与802.11e MAC层的访问类（AC）相结合,基于重要性早期检测（SBED）策略将SVC数据包动态映射到合适的AC上,在基础层损失率与平均重要性损失度间实现良好的平衡。仿真表明,该方案的PSNR性能明显优于现有的静态与随机映射方案。     

Internet多媒体分层组播JSCC差错控制率失真优化

针对Internet多媒体群组通信中同时存在的带宽异构性和包丢失率异构性，文中将分层组播和接收者驱动的思想扩展到FEC差错控制中，提出一种分层FEC组播差错控制方法LM-FEC．LM-FEC通过不同的组播组发送信源编码层和各信源层的FEC校验数据，为接收者根据信道带宽和数据包丢失率实施差错控制提供更加灵活的选择．文中用FH-MDP模型描述接收者行为，通过JSCC率失真优化确定编码层内和编码层间的速率分配，JSCC率失真优化采用变量替换和动态规划算法求解．实验表明，该文提出的差错控制方法能够有效改善重建多媒体信号的回放质量．     

可靠多媒体多播传输协议

对于大规模多媒体多播应用来说,一个有效发现和修正传输错误的可靠多媒体多播协议是必要的。该文研讨了多媒体多播应用中的允许延迟和分组丢失率问题,提出了一种基于协议转接概念的可靠多播传输协议(RMTP)。RMTP协议聚焦在允许延迟上提供多媒体服务质量保证,转接节点放置在多播树上,数据恢复在两个转接节点之间进行。RMTP协议转接能立即满足重传需求和减少分组的复制数量。最后,给出了RMTP协议的性能分析及与SRM协议的比较。研究表明RMTP协议为多媒体多播传输提供了一种新的有效途径。     

移动Ad Hoc网络中基于NACK的可靠多播协议

如何采用有效的机制来保证多播数据的可靠传输是移动Ad Hoc网络中的一个难题.针对IEEE 802.11 MAC层进行改进,提出了一种基于使用NACK消息的可靠多播协议,该协议仅当数据分组传输出错时才发送NACK消息要求重传数据.仿真结果表明使用该协议能提高多播路由传输数据的吞吐量,同时降低了平均时延.     

基于无线TCP的簇生丢失重传协议

为提高TCP在无线网络中的传输性能,提出一种局部数据链路层重传协议,簇生丢失重传协议CLRP(Clustered-Loss Retransmission Protoc01)．针对无线链路上突发丢失性强、分组丢失率高的特点,CLRP协议与移动主机端TCP相结合,一方面为重发提供明确的分组丢失种类和高效的无线多分组丢失信息;另一方面提出了更为完善的无线丢失重发控制机制．此外,本协议无需对固定主机端TCP做任何改动．     

Performance evaluation of AL-FEC LDPC codes for push content applications in wireless unidirectional environments

FEC (Forward Error Correction) mechanisms improve IP content transmission reliability through the recovery of packets lost in transmission. Opposite to ARQ (Automatic Repeat Request), FEC mechanisms are especially suited to unidirectional environments or to multicast environments where multiple receivers perceived different channel losses, thus making difficult the implementation of mechanisms based on feedback information. Among the different types of FEC codes, this paper presents a thorough performance evaluation of LDPC (Low Density Parity Check) codes, based on an implementation developed by the authors, according to the specifications defined by RFC 5170 for the usage of LDPC codes by push content applications based on the FLUTE protocol. LDPC codes provide a good trade-off between performance and complexity, hence, they are appropriate for mobile applications. Contributions of this paper include tests conducted with commercial mobile phones connected to the push content download server over a Wi-Fi network. The evaluation highlights the advantages of using packet level FEC encoding in file transmission over unidirectional networks and provides with a comparison between two kinds of LDPC structures: Staircase and Triangle. This is accomplished by calculating the inefficiency ratio of these LDPC structures in different environments. Results show that the implemented LDPC codes can provide inefficiency ratios close to one when the different coding parameters (as the code rate or the number of blocks) are configured to an optimal value that depends on the packet loss rate.     

基于转接节点的可靠多媒体多播协议

研讨了多媒体多播应用中的允许延迟和分组丢失率问题,提出了一种基于转接节点概念的可靠多播协议（RMPRM）。RMPRM协议聚焦在允许延迟上提供多媒体服务质量保证,转接节点放置在多播树上,数据恢复在两个转接节点之间进行。RMPRM协议转接能满足重传需求和减少分组的复制数量。给出了RMPRM协议与不可靠多播协议的比较。仿真实验表明,该协议具有较高的传输率和较低的端到端的传输延迟。     

一种优化的多播报文认证机制

提出了一种有效的多播报文认证机制，该机制结合了Hash树和Hash链两种方法的特点。在发送一组多播报文时，首先将其划分为大小相等的多个子组，子组的大小由预计抵御的突发丢包发生次数确定。然后为每个子组内的报文建立一棵Hash树，并将每棵Hash树的树根附加于之前的若干个报文中，从而构成了Hash链。该文使用了两种丢包模型对这种机制的性能进行了分析和模拟，其结果表明该机制在达到相同校验率的情况下，可以降低通信开销。