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«End-to-End QoS Provision over Heterogeneous IP and non IP Broadband Wired and Wireless Network Environments A dissertation submitted in satisfaction ...»

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mechanisms. The paper addresses the end-to-end QoS problem of scalable video streaming traffic delivery over a heterogeneous IP/UMTS network. It proposes and validates through a number of NS2-based simulation scenarios a framework that explores the joint use of packet prioritization and scalable video coding, by evaluating two FGS video encoders, together with the appropriate mapping of UMTS traffic classes to the DiffServ traffic classes. It is observed that, the scalable extension of H.264/MPEG-4 AVC can achieve better quality gains compared to MPEG-4 FGS, due to the applied motion-compensated prediction technique.

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End-to-End QoS Issues of MPEG-4 FGS video streaming traffic delivery in an IP/DVB/UMTS networking environment

5.1 Introduction The Fine Grain Scalability (FGS) feature of MPEG-4 is a promising scalable video solution to address the problem of guaranteed end-to-end QoS provision.

In the MPEG-4 FGS standard, a video is encoded into two bitstreams: the Base Layer (BL) and the Enhancement Layer (EL). The BL must be completely received to decode and display a basic quality video. The FGS EL can be cut anywhere at the granularity of bits and the received part can be decoded and improved upon the basic quality video. This FGS, which is achieved by a bitplane coding technique [2], allows the server to adapt the transmission rate finely to changing network conditions. In typical scenario for transmitting MPEG-4 FGS encoded videos over heterogeneous networks, the BL is transmitted with the high reliability (achieved through appropriate resource allocation and/or channel error protection) and the EL is transmitted with low reliability (e.g. in a best effort manner and without error control). However, scalable coding only solves part of the problem, and packet loss is very common with unpredictable channel conditions. To address this problem, both an efficient scalable video coding scheme with a flexible delivery technique and a scalable network management framework are needed. By using rate-allocation mechanism, prioritized packetization and differential forwarding, the application-layer QoS can be provided to the end user.

Concerning the network perspective, an all-IP setting seems to be able to resolve the inter-working amongst the diverse fixed core and wireless/mobile access technologies and the end-to-end QoS provision could be established through the appropriate mapping amongst the QoS traffic classes/services supported by the contributing underlying networking technologies. Building this concept, this work concerns a heterogeneous cluster of networks consisting of two DiffServaware, a DVB network acting as a trunk network and a UMTS network acting as an access network.

For the fixed networks, the Differentiated Services (DiffServ) [3] model, proposed by IETF, provides a less complicated and more scalable solution because Integrated Services (IntServ) [4] requires maintenance of the per-flow state across the whole path for resource reservation. In the DiffServ model, resources are allocated differently for various aggregated traffic flows based on a set of bits. DiffServ model support two different services: (1) the Expedited Forwarding (EF) [5] that supports low loss and delay/jitter, and (2) the Assured Forwarding (AF) [6] that provides QoS better than the best effort, but without guarantee. For streaming video applications, in which the encoding and decoding process is more resilient to packet loss and delay variations, besides EF, the AF can be employed.

In order to provide traffic differentiation in a Digital Video Broadcasting (DVB) [7] network, Bandwidth Management (BM) techniques can be applied on queues containing 188 byte long MPEG-2 TS packets. This technique is based on the dynamic uplink bandwidth reallocation into a number of independent virtual channels according to a predefined set of priority policies. The assignment of an IP flow at a virtual channel is achieved through a filtering mechanism, which is able to monitor traffic and based on some pre-defined filters (IP source and destination addresses, source and destination ports, protocol type, etc) to encapsulate that traffic to a specific virtual channel.

The QoS provision in Universal Mobile Telecommunications System (UMTS) [8] is achieved through the concept of bearers. A bearer is a service providing a particular QoS level between two defined points invoking the appropriate schemes for either the creation of QoS guaranteed circuits, or the enforcement of special QoS treatments for specific packets. The selection of bearers with the appropriate characteristics constitutes the basis for the UMTS QoS provision. Each UMTS bearer is characterized by a number of quality and performance factors. The most important factor is the bearers Traffic Class; four traffic classes have been defined in the scope of the UMTS framework (i.e., Conversational, Streaming, Interactive and Background). The appropriate mapping of UMTS traffic classes to the aforementioned DiffServ service classes could offer a vehicle for the end-toend QoS provision over a heterogeneous DiffServ/UMTS network. In this chapter, I evaluate two different mapping approaches of traffic classes for the end-to-end QoS provision over a heterogeneous DiffServ/UMTS network [9][10].

To address the end-to-end QoS guarantees across heterogeneous network, like DiffServ/DVB/UMTS, the paper proposes and validates through a number of experimental scenarios an architecture that explores the joint use of rate adaptation with scalable coding, packet prioritization, together with the appropriate mapping of UMTS traffic classes to the DiffServ traffic classes.





This chapter is organized as follows. In Section 5.2, the proposed video coding and prioritization framework for providing QoS guarantees for MPEG-4 FGS

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video streaming traffic delivery over a heterogeneous DiffServ/DVB/UMTS network is presented, in which key components such as the scalable video coding and differential forwarding across different heterogeneous network domains, including fixed and wireless/mobile networks are employed. The testbed configuration details for the media delivery experimental studies and the results of these studies are discussed in Sections 5.3 and 5.4, respectively. Finally, Section 5.5 draws conclusions and discusses directions for further work and improvements.

5.2 Proposed Arrhitecture

The proposed framework is focused on the integration of rate allocation within MPEG-4 FGS video streaming; prioritized packetization based on content and heterogeneous QoS-aware network systems for providing end-to-end QoS over IP/DVB/UMTS systems. The proposed framework is shown in Figure 5.2.

This work deals with the following key components: (1) scalable source encoding with constant quality rate allocation, (2) prioritized packetization, and (3) differential forwarding across heterogeneous network domains. They are briefly

described below:

• Scalable Coding with rate allocation The video sequence is encoded using MPEG-4 FGS codec, where the estimated minimal bandwidth, provided by the network monitoring system, gives the rate constraint for BL. Then the rate-allocation module scales the EL stream based on the feedback of the available bandwidth, to preserve constant quality by referring to R-D samples, produced by the video analysis of the video sequence.

• Prioritized Packetization Fixed length packetization scheme is adopted, to packetized BL and EL bit-streams, as proposed by MPEG-4 [11]. It applies priorities based on the loss impact of each packet to the end-to-end video quality.

• Differential Forwarding - The focus of network-level QoS mapping is to ensure the vertical QoS continuity across different network domains. Basically, the application, network, and data link layers are involved in this mapping. The main motivation is to assign different priorities to parts of a video bit stream that represents the content on the application layer. The BL in case of scalable bit-stream, is regarded as most important for the decoding process and, therefore, should be transmitted with a higher priority than less important EL, and so on. These priorities at the application layer are then mapped to different Differentiated Service Code Points (DSCPs) [12] at the network layer. That is, packets containing important parts of the bitstream receive a higher packet priority than packets containing less prioritized parts. This can be realized by using different QoS classes that differ only in the drop probability (e.g., AF11 for high priority packets and AF12 for low priority packets). These different priorities at the network layer may be mapped to QoS mechanisms available at DVB BM virtual channels and the UMTS traffic classes. UMTS offers four different classes, which can be used for service differentiation between real-time traffic (e.g., video streaming) and best-effort traffic. Authors propose the mapping of DSCPs to the BM virtual channels and UMTS traffic classes in order to ensure the vertical QoS continuity across different network domains.

5.2.1 Rate allocation with scalable video coding

The video sequence is encoded based on the estimated minimal bandwidth, provided by the network monitoring system, which gives the bandwidth requirements for BL. The encoding is being performed based on the collected statistics, generated by the video sequence analysis. For the EL, the generated R-D samples are stored either in the user data of each Video Object Plane (VOP) or as metadata in a separate file. Then, the rate allocation module truncates the EL stream, according to the feedback of the available bandwidth in order to increase quality by referring to information, provided by the generated R-D samples.

To make practical and effective use of MPEG-4 FGS encoding, a rate control algorithm is needed to transfer the rate constraint into the rate assigned to each frame, and also to minimize the variation quality. A simple method is constant quality rate allocation (CBR) but the usage of this method does not achieve high results in overall video quality due to quality fluctuations. In order to tackle this problem, variable bit-rate (VBR) allocation is proposed for constant quality reconstruction by allocating rate according to the complexity of each frame [13].

Authors in [14] propose an optimal rate allocation using an exponential model.

In [15], constant quality rate allocation is proposed that minimizes the sum of absolute differences of qualities between adjacent frames under the rate constraint.

However the optimality of this approach depends on the initial condition, which is computed based on the assumption that the average distortion of CBR rate allocations is close to the distortion of the constant quality rate allocation. In fact, the two distortions must be within the same R-D sample interval for all frames in order to have a valid solution to the set of linear equations. According to piecewise linear interpolation, described in [15], the rate allocation can be calculated by

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where Ctotal is the available bandwidth, N is the total number of frames, Ri denotes the source frame rate, and Ri is the optimal rate that should allocated to i frames in order to achieve the constant distortion D. Consider Rmi, Dmi and Rni, Dni to be two ajdacent R − D points, such that Dmj D Dni and Rni. In the above equations, ∆Ri = Rni −Rmi and ∆Di = Dmi −Dnmi Rmj R represent the difference in rate and distortion at adjacent R-D points, respectively.

5.2.2 Prioritized Packetization Scheme

In order to packetize MPEG-4 video streams, fixed-length packetization scheme is adopted, where video packets of similar length are formed. The packet size of video stream is also related to efficiency and error resiliency because a smaller packet size for example requires a higher overhead but has a better performance in error prone networks. Then, each packet is identified by a particular priority in accordance with its impact on end-to-end visual quality. For different service preferences in terms of loss and delay, the priority can be further divided into the RLI and RDI, as authors proposed in [16] [17].

To determine packet priority with low computational complexity is an active research area today. Several features, such as initial error strength, propagation via motion vectors, and the spatial filtering effect were used to develop a corruption model in [18] to determine packet priority in terms of loss impact. For BL packets, we use a fixed Equal Error Protection (EEP) scheme, where all packets are high priority and they are transmitted using the EF class.

The packet loss within the EL only affects a single frame, and it does not propagates, the incurred distortion from each EL packet can be accurately calculated

within each frame, and the packet priority can be calculated as:

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where ∆Di represents the incurred distortion due to the specified loss and ∆Ri is the rate of the packet concerned. Furthermore, packet dependency must be considered to that if packets containing a more significant bitplane get lost, packets containing a less significant bitplane in the same region get discarded anyway. By using the piecewise-linear R-D model for each bitplane, the priority of EL packets can be easily calculated online during the packetization procedure.

However, for simplicity reasons, a simpler QoS mapping policy in this framework is adopting, by using direct mapping of packets to DiffServ classes. All packets are formed into three groups, according the type of context that they contain, and each group of packet is mapped to one DiffServ class. Table 5.1 depicts the relation between the type of the EL content and the corresponding DiffServ classes. The first digit of the AF class indicates forwarding priority and

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the second indicates the packet drop precedence.

5.2.3 DVB Domain - BM Implementation The bandwidth reallocation among the IP virtual channels of a DVB MPEG-2 TS uplink is based on a set of predefined priority policies [19]. The paper implements

three priority policies, which are:



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