TITRE Adaptive Packet Video Streaming Over IP Networks - LaBRI

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packets. If the number of lost packets is not more than h i , then the decoder will be able to recover Ui. Otherwise, Ui is completely lost. Figure 4-16 shows the format of packets sent on the IP network. d bytes IP/UDP/RTP header 4 bytes FEC header seq n k RTP Payload Figure 4-16: Header information and packet format UEP augments the amount of the traffic sent in the network. To correctly control the volume of transmitted data, we have to provide a certain ratio of traffic overhead called r, for each level of priority score. We assume that moving from one priority score to other increases by a 10 percent ratio. Then, the ratio r can be defined by: r = 0 . 1× p (Eq.3) Therefore, the traffic overhead is limited to 10 percent (i.e., r=0.1) for the data flow of priority score 1, to 20 percent (i.e., r=0.2) for the data flow of priority 2, and so on. In order to find the efficient value n i for the (n i , k ) i RS code, we proceed as below: ∗ Let ς U i be the reserved byte-budget for error protection. It depends on the number of bytes used to send U i when no error protection is performed. It is given by: ( k ⋅ d m ) ς (Eq.4) ∗ U = r ⋅ i i + i Where d is the packet header size (i.e, when RTP/UDP/IP is used with the proposed UEP, d = (20+8+12+4) = 44 bytes). The relation between the real byte-budget spent on error protection, ς U i , and the RS code to be used can be stated as follows: The error margin between ( n − k ) ⋅ ( t d ) ς (Eq.5) U i = i i i + ς U i and ∗ Ui ∧ ∗ ς is ς ς U − ς = i Ui , that can be positive or negative. It cumulates along the data access unit arrivals. Using the formula (Eq.4) and (Eq.5), the fluctuation of the error margin can be written as: ∧ ς ( n ) i ⎧ ⎪ = ⎨ ⎪ ⎩ 0 , i = 1 ∧ ⎪( r + 1) ⋅ m − n ⋅ t + ( r − n + k ) ⋅ d + ς ( n ) i i i i i i−1 , i > 1 (Eq.6) To respect the constraint given on traffic overhead, the best value n i is the one that provides the smallest error margin in formula (Eq.6). Then, n i is obtained by: min ni ∧ ( ni ) such that ni ∈ℵ, ni ≥ ki , ∀mi , ki , ti ς (Eq.7) With the proposed UEP, the RS code evolves dynamically so the network bandwidth is correctly controlled according to video application requirement. 86
4.2.3 Performance Evaluation Intensive simulations are conducted to evaluate the performance of the proposed RTP payload. We have used the network simulator ns2 in which we have implemented the proposed RTP payload. An MPEG-4 Server (NS2 Agent) and an MPEG-4 client (NS2 Agent) are designed for this purpose. The server reads and sends the different MPEG-4 AVOs found in video trace files to the client though an IP network. 4.2.3.1 System and Network Models For the simulation, we used the network simulation models depicted in Figure 4-17 for evaluating and comparing our proposal with the classical approach described in IETF draft of the RTP Payload Format for Transport of MPEG-4 Elementary Streams [159]. MPEG-4 audio video streams are transported over an IP network using both approaches. Also, the network topologies and parameters are similar for both approaches. Links characteristics between the different network elements (i.e. the channel bit rate and the transfer delay) are illustrated in the Figure 4-17. FTP null 1 5Mbs 5mS 5Mbs 5mS 3 Serveur MPEG -4 S 5Mbs 5mS 2 5Mbs 5mS R1 5Mbs 5mS R2 5Mbs 5mS 4 5Mbs 5mS C Client MPEG-4 CBR Sink Figure 4-17: IP network model. The MPEG-4 source terminal is attached to the node “S”. It sends a customize MPEG-4 streams to the MPEG-4 destination terminal which is attached to the node “C” according to two scenarios (scenario 1 and scenario2). We include a constant-bit-rate (CBR) traffic over UDP to make the link between the nodes “R1” and “R2” congested A CBR sources allows loading the network differently each time in order to get further information of our packetization scheme. 4.2.3.1.1 Scenario 1 In the scenario 1, we demonstrate the effect of the multiplexing of our RTP payload. We transmit an MPEG-4 scene composed only of CELP stream (6Kbit/s). In order to optimize the bandwidth usage, we maximize the RTP payload exploitation for both approaches. The RTP packet size is fixed to 210 bytes. This permits encapsulating 12 AUs from a CELP stream into one RTP payload. The encapsulation takes care of the RTP payload header fields (i.e. SL packet header fields) using for either [159] or our proposal. In the classical encapsulation algorithm [159], the concatenation of several AUs, from a single CELP stream, into the RTP payload induces the transport of 12 AUs (i.e. 240 ms of voice) per RTP packet. With our encapsulation proposal, we achieve a multiplexing of 3 different CELP streams, which will result with the transport of 4 AUs (i.e. 80 ms of voice) from each multiplexed CELP stream. At the end, we simulate both of the approaches through a network varying conditions. 87
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UNIVERSITÉ DE VERSAILLES SAINT-QUE
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Résumé Nous constatons aujourd’
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3 Related Work ....................
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5.5.1.2 Simulation Results.........
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List of Figures Figure 1-1: Structu
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Figure 5-15: End-to-end packet tran
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Acknowledgments The first person I
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[11] Toufik Ahmed, Guillaume Burida
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d’accès. Les interactions de QoS
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Métrique MAX_DELAY PREF_MAX_DELAY
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1.1.2 Proposition dun Protocole d'E
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des paquets RTP. Reste toujours le
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Nous pouvons distinguer trois possi
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1.1.2.5.2 Multiplexage des Flux El
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l’horloge est par défaut égale
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La valeur R TCP représente le déb
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Pour ces raisons, nous nous intére
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Telles que illustrés par la Figure
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3. DMIF2SIP vérifie si son propre
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utilisée optionnellement pour loca
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Chapter 2 2 Introduction 2.1 Motiva
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conditions. Based on end-to-end fee
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Chapter 3 3 Related Work Nowadays,
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• multicasting: it takes the stre
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expensive, fixed delivery and recep
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Page 54 and 55: environment because they require a
Page 56 and 57: H.261 is called video codec for aud
Page 58 and 59: VideoScene (VS) VideoObject (VO) Vi
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Page 62 and 63: Temporal scalability involves parti
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Page 70 and 71: correctly received block. The missi
Page 72 and 73: epresent user traffic. The delay ca
Page 74 and 75: 3.2.5 IP Signaling Protocols for Pa
Page 76 and 77: two H.323 endpoints or between an e
Page 79 and 80: Chapter 4 4 Adaptive Packet Video T
Page 81 and 82: 4.1.1 Video Classification Model Pr
Page 83 and 84: attributes can refer to the QoS par
Page 85 and 86: In RBF, 2 D = X − is the distance
Page 87 and 88: Figure 4-6: Experiment testbed In t
Page 89 and 90: 190 180 170 End-to-end delay MPEG-4
Page 91 and 92: Sync Layer. The SL-packetized strea
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Page 95 and 96: the same value. This timing informa
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Page 105 and 106: 0.3 0.25 our proposal classical app
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Page 111 and 112: The video server adds audio-visual
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Page 115 and 116: VO1 VO2 AVO Scenario Audio BL EL1 E
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Page 125 and 126: Chapter 5 5 A Dynamic Packet Video
Page 127 and 128: The TCM algorithm is based on a tok
Page 129 and 130: BEGIN Initialize the classification
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The proposed marking algorithm bett
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Chapter 6 6 A SIP/MPEG-4 Multimedia
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performed by many ways like SLA (Se
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6.1.1.4 SIP Communication Model SIP
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• It assign a value (network sess
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6.2.1 SIP2DMIF Subsystem The SIP2DM
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Step 1: The MPEG-4 Terminal passes
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1. Set the result C to the empty se
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TCP/UDP port, our implementation us
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Control & Data Plan Video Rate Cont
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characteristics and relevance of ea
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References [1] Kazaa Media Desktop
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[43] B. Schuster, “Fine granular
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[79] D. Bansal and H. Balakrishnan,
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[115] N. Laoutaris, I. Stavrakakis,
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[154] W. Almesberger, J. H. Salim,
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Appendix A A Appendix A: Overview o
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Media aware Delivery unaware ISO/IE
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A.3.3 Scene Description Streams Sce
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This appendix presents structure ex
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