Achieving inter-session fairness for layered video multicast

The Internet is increasingly used to deliver multimedia services. Since there are heterogeneous receivers and changing network conditions, it has been proposed to use adaptive rate control techniques such as layered video multicast to adjust the video traffic according to the available Internet resources. A problem of layered video multicast is that it is unable to provide fair bandwidth sharing between competing video sessions. We propose two schemes, layered video multicast with congestion sensitivity and adaptive join-timer (LVMCA) and layered video multicast with priority dropping (LVMPD), to achieve inter-session fairness for layered video multicast. Receiver-driven layered multicast (RLM), layer-based congestion sensitivity, LVMCA, and LVMPD are simulated and compared. Results show both proposed schemes, especially LVMPD, are fairer and have shorter convergence time than the other two schemes.     

A case for end system multicast

The conventional wisdom has been that Internet protocol (IP) is the natural protocol layer for implementing multicast related functionality. However, more than a decade after its initial proposal, IP multicast is still plagued with concerns pertaining to scalability, network management, deployment, and support for higher layer functionality such as error, flow, and congestion control. We explore an alternative architecture that we term end system multicast, where end systems implement all multicast related functionality including membership management and packet replication. This shifting of multicast support from routers to end systems has the potential to address most problems associated with IP multicast. However, the key concern is the performance penalty associated with such a model. In particular, end system multicast introduces duplicate packets on physical links and incurs larger end-to-end delays than IP multicast. We study these performance concerns in the context of the Narada protocol. In Narada, end systems self-organize into an overlay structure using a fully distributed protocol. Further, end systems attempt to optimize the efficiency of the overlay by adapting to network dynamics and by considering application level performance. We present details of Narada and evaluate it using both simulation and Internet experiments. Our results indicate that the performance penalties are low both from the application and the network perspectives. We believe the potential benefits of transferring multicast functionality from end systems to routers significantly outweigh the performance penalty incurred.     

Challenges and opportunities of delivering IP-based residential television services

This article describes the main challenges of implementing an end-to-end architecture for delivery of high-quality, IP-based residential TV services to residential customers. The IP-based approach can be implemented with an IP multicast overlay network with traditional routers or use IP-multicast-aware ATM switching systems. Both approaches use IP multicast to transport MPEG-2 broadcast video and can work on any access architecture, especially on copper-based architectures (xDSL) such as ASDL and VDSL. The main challenges met while implementing the IP-based architecture are competitive positioning relative to traditional CATV architectures, overall architecture, head-end architecture and quality issues, traffic engineering for stringent QoS requirements, IP multicast requirements, and business case considerations. The IP-based approach described leverages Internet technology advancements and capitalizes on the impacts of Internet on interactive entertainment. Video channel manipulation at the head-end is dependent on access bandwidth and affects video quality. Video traffic management to meet stringent QoS requirements is challenging at the IP layer. High-capacity, responsive IP multicasting is essential to achieving high service quality. Cost-effective IP multicasting is a critical component of the business case.     

一种新的逐跳TCP友好的主动分层多播拥塞控制方案

基于Internet的多媒体多播应用的迅猛发展对多播拥塞控制提出了要求.分层多播是适应网络异构性较有效的方案.针对现有分层多播存在的问题,将主动网技术思想引入到分层多播拥塞控制中,提出了一种逐跳TCP友好的主动分层多播拥塞控制方案(HTLMA),采用主动标记分层、逐跳探测TCP友好可用带宽,以及主动速率控制机制.仿真实验表明,HTLMA方案大大改进了分层多播拥塞控制的性能,具有较快的拥塞响应速度、较好的稳定性和TCP友好特性.     

Multicast Support in DiffServ Using Mobile Agents

Many multicast applications, such as video‐on‐demand and video conferencing, desire quality of service (QoS) support from an underlying network. The differentiated services (DiffServ) approach will bring benefits for theses applications. However, difficulties arise while integrating native IP multicasting with DiffServ, such as multicast group states in the core routers and a heterogeneous QoS requirement within the same multicast group. In addition, a missing per‐flow reservation in DiffServ and a dynamic join/leave in the group introduce heavier and uncontrollable traffic in a network. In this paper, we propose a distributed and stateless admission control in the edge routers. We also use a mobile agents‐based approach for dynamic resource availability checking. In this approach, mobile agents act in a parallel and distributed fashion and cooperate with each other in order to construct the multicast tree satisfying the QoS requirements.     

基于网络控制的分层多点播送速率控制机制研究

本文在视频分层编码及分层传输协议的基础上，将基于网络控制与接收端控制机制相结合，提出一种新的分层多点播送速率控制机制。文中给出了该机制的拥塞检测、流行度权衡和数据层增加和丢弃算法。实验结果表明，该速率控制机制通史对拥塞做也快速响应，并在较好利用宽带的基础上保证高流行度的会话流具有较好的服务质量。     

Revisiting the fair queuing paradigm for end-to-end congestion control

Today, the dominant paradigm for congestion control in the Internet is based on the notion of TCP friendliness. To be TCP-friendly, a source must behave in such a way as to achieve a bandwidth that is similar to the bandwidth obtained by a TCP flow that would observe the same round-trip time (RTT) and the same loss rate. However, with the success of the Internet comes the deployment of an increasing number of applications that do not use TCP as a transport protocol. These applications can often improve their own performance by not being TCP-friendly, which severely penalizes TCP flows. To design new applications to be TCP-friendly is often a difficult task. The idea of the fair queuing (FQ) paradigm as a means to improve congestion control was first introduced by Keshav (1991). While Keshav made a fundamental step toward a new paradigm for the design of congestion control protocols, he did not formalize his results so that his findings could be extended for the design of new congestion control protocols. We make this step and formally define the FQ paradigm as a paradigm for the design of new end-to-end congestion control protocols. This paradigm relies on FQ scheduling with per-flow scheduling and longest queue drop buffer management in each router. We assume only selfish and noncollaborative end users. Our main contribution is the formal statement of the congestion control problem as a whole, which enables us to demonstrate the validity of the FQ paradigm. We also demonstrate that the FQ paradigm does not adversely impact the throughput of TCP flows and explain how to apply the FQ paradigm for the design of new congestion control protocols. As a pragmatic validation of the FQ paradigm, we discuss a new multicast congestion control protocol called packet pair receiver-driven layered multicast (PLM).     

A survey of active and router-assisted reliable multicast solutions

Group communications, so-called multicast communications, have been introduced as early as 1986 as an efficient way to distribute on a large-scale basis data packets on the whole Internet. With the emergence of new applications driven by the increase of bandwidth in the networks (such as grid computing, large database replication, etc.), reliable multicast is foreseen to be one of the most challenging technologies of the next generation Internet. However, group communications are much more complex than point-to-point communications and solutions that have been implemented and deployed in the unicast world for ensuring reliability can not be so simply applied to the multicast area. Recently, a new paradigm proposes to dynamically add additional functionalities into the routers, thus enabling a whole new range of distributed control mechanisms as opposed to the traditional end-to-end form of control. In this paper, we summarize the various active networking and router-assisted solutions for reliable multicast to avoid or limit the scalability problems of end-to-end approaches for loss recovery, congestion control and heterogeneity support. Then, with the hypothesis that active networking could be deployed on a large scale, we present how this technology could solve the remaining bottlenecks of multicast on wide-area networks.     

Delay performance analysis of the new Internet services withguaranteed QoS

In addition to the traditional best effort Internet protocol (IP) service of the Internet, the Internet Engineering Task Force has defined two new services that provide quality of service guarantees on throughput and delay. In this paper, we analyze the delay performance that can be achieved with the service that provides the most firm guarantees-the guaranteed quality of service. Using a voice and a video application as examples, the end-to-end delay is calculated. Three different scenarios are presented: local, regional, and long distance, differing in the distance covered and the number of IP routers traversed. It is shown that even in the long-distance case, the achievable end-to-end delay for the guaranteed service can be kept sufficiently low to support interactive applications with strict delay requirements. It is concluded that for low-rate applications, the efficiency cannot be higher than about 60-75%. For connections traversing a large number of routers, the variable delay as advertised by the routers is much higher than the expected delay. The second new service, called the controlled-load network element service, provides less firm delay guarantees. The end-to-end delay for this service is estimated using similar techniques. The results suggest that this service is suitable for interactive applications in the local and regional scenarios only. For the long distance scenario, the end-to-end delay is expected to be below 300 ms     

Deployment issues for the IP multicast service and architecture

IP multicast offers the scalable point-to-multipoint delivery necessary for using group communication applications on the Internet. However, the IP multicast service has seen slow commercial deployment by ISPs and carriers. The original service model was designed without a clear understanding of commercial requirements or a robust implementation strategy. The very limited number of applications and the complexity of the architectural design-which we believe is a consequence of the open service model-have deterred widespread deployment as well. We examine the issues that have limited the commercial deployment of IP multicast from the viewpoint of carriers. We analyze where the model fails and what it does not offer, and we discuss requirements for successful deployment of multicast services     

Managing IP services over a PACS Packet Network

We have developed a system and network architecture to provide IP services in the Personal Access Communications System (PACS). IP datagrams are delivered to PACS users through the PACS packet-mode data service, achieving more efficient usage of wireless resources and supporting multimedia applications such as MBone audio and video. The architecture presented in this article augments the PACS voice network with IP routers and backbone links, called the PACS Packet Network (PPN), and is connected to the global Internet via gateways. Compared to the cellular digital packet data (CDPD) network, which employs its own network-layer mobility protocol and thus supports roaming within the CDPD network only, we have incorporated Mobile IP into the PACS handoff mechanism to further achieve global IP mobility. We have also developed native PACS multicast and a group management scheme to efficiently handle dynamic IP multicast and MBone connectivity. These features seamlessly integrate PACS into the global Internet and provide standard-conforming IP services with global mobility     

A survey on TCP-friendly congestion control

New trends in communication, in particular the deployment of multicast and real-time audio/video streaming applications, are likely to increase the percentage of non-TCP traffic in the Internet. These applications rarely perform congestion control in a TCP-friendly manner; they do not share the available bandwidth fairly with applications built on TCP, such as Web browsers, FTP, or e-mail clients. The Internet community strongly fears that the current evolution could lead to congestion collapse and starvation of TCP traffic. For this reason, TCP-friendly protocols are being developed that behave fairly with respect to coexistent TCP flows. We present a survey of current approaches to TCP friendliness and discuss their characteristics. Both unicast and multicast congestion control protocols are examined, and an evaluation of the different approaches is presented     

Light-weight multicast services (LMS): a router-assisted scheme for reliable multicast

Building on the success of unicast IP, IP Multicast adopted a simple, open, best-effort delivery model with many-to-many semantics. Despite several years of effort, a general, scalable and reliable end-to-end transport protocol analogous to TCP has proven elusive. Proposed solutions are either inflexible, or incur high control overhead. We present Lightweight Multicast Services (LMS), which enhance the IP Multicast model with simple forwarding services to facilitate scalable and efficient (compared to pure end-to-end) solutions to problems such as reliable multicast. In LMS, routers tag and steer control packets to preselected endpoints and perform fine-grain multicast to guide responses to a subset of the group without transport-level processing. LMS divides error control into transport and forwarding components, which allows the former to remain at the end-points while the latter is pushed to the routers, where it can be implemented very efficiently. The division is clean, resulting in significant gains in performance and scalability, while reducing application complexity. LMS reaches beyond reliable multicast to applications such as scalable collect, any-cast, and in general, any application that can benefit from a hierarchy congruent with the underlying topology.     

Transporting real-time video over the Internet: challenges andapproaches

Delivering real-time video over the Internet is an important component of many Internet multimedia applications. Transmission of real-time video has bandwidth, delay, and loss requirements. However the current Internet does not offer any quality of service (QoS) guarantees to video transmission over the Internet. In addition, the heterogeneity of the networks and end systems makes it difficult to multicast Internet video in an efficient and flexible way. Thus, designing protocols and mechanisms for Internet video transmission poses many challenges. In this paper, we take a holistic approach to these challenges and present solutions from both transport and compression perspectives. With the holistic approach, we design a framework for transporting real-time Internet video, which includes two components, namely, congestion control and error control. Specifically congestion control consists of rate control, rate-adaptive encoding, and rate shaping; error control consists of forward error correction (FEC), retransmission error resilience, and error concealment. For the design of each component in the framework, we classify approaches and summarize representative research work. We point out there exists a design space which can be explored by video application designers and suggest that the synergy of both transport and compression could provide good solutions     

基于分层多播的视频传输拥塞控制算法研究

分层多播以多速率方式解决了多播接收者异构性问题,对于提高网络服务质量具有重要意义。本文分析了分层多播传输特性,通过提取MPEG视频流中的I帧、P帧、B帧组成3个帧流,分别放到分层多播的基层、增强层1和增强层2上传输,并在中间节点采用优先级队列机制,提出了一种面向视频流传输的分层多播拥塞控制(VLMCC)算法。仿真实验表明,本文提出的VLMCC算法能够适应视频多播接收者的异构性,大大提高了视频多播传输质量。     

The impact of multicast layering on network fairness

Many definitions of fairness for multicast networks assume that sessions are single rate, requiring that each multicast session transmits data to all of its receivers at the same rate. These definitions do not account for multirate approaches, such as layering, that permit receiving rates within a session to be chosen independently. We identify four desirable fairness properties for multicast networks, derived from properties that hold within the max-min fair allocations of unicast networks. We extend the definition of multicast max-min fairness to networks that contain multirate sessions, and show that all four fairness properties hold in a multirate max-min fair allocation, but need not hold in a single-rate max-min fair allocation. We then show that multirate max-min fair rate allocations can be achieved via intra-session coordinated joins and leaves of multicast groups. However, in the absence of coordination, the resulting max-min fair rate allocation uses link bandwidth inefficiently, and does not exhibit some of the desirable fairness properties. We evaluate this inefficiency for several layered multirate congestion control schemes, and find that, in a protocol where the sender coordinates joins, this inefficiency has minimal impact on desirable fairness properties. Our results indicate that sender-coordinated layered protocols show promise for achieving desirable fairness properties for allocations in large-scale multicast networks     

Mobile and Multicast IP Services in PACS: System Architecture, Prototype, and Performance

Traditionally, wireless cellular communication systems have been engineered for voice. With the explosive growth of Internet applications and users, there is an increasing demand on providing Internet services to mobile users based on the voice-oriented cellular networks. However, Internet services add a set of radically different requirements on to the cellular wireless networks, because the nature of communication is very different from voice. It is a challenge to develop an adequate network architecture and necessary systems components to meet those requirements.This paper describes our experience on developing Internet services, in particular, mobile and multicast IP services, in PACS (Personal Access Communication Systems). Our major contributions are five-fold: (i) PACS system architecture that provides wireless Internet and Intranet access by augmenting the voice network with IP routers and backbone links to connect to the Internet; (ii) simplified design of RPCU (Radio Port Controller Unit) for easy service maintenance and migration to future IP standards such as IPv6; (iii) native PACS multicast to efficiently support dynamic IP multicast and MBone connectivity; (iv) optimization and incorporation of Mobile IP into PACS handoff mechanism to efficiently support roaming within a PACS network as well as global mobility between PACS networks and the Internet; (v) successful prototype design of the new architecture and services verified by extensive performance measurements of IP applications. Our design experience and measurement results demonstrate that it is highly feasible to seamlessly integrate the PACS networks into the Internet with global IP mobility and IP multicast services.