Robust MAC-Lite and soft header recovery for packetized multimedia transmission

International audienceThis paper presents an enhanced permeable layer mechanism useful for highly robust packetized multimedia transmission. Packet header recovery at various protocol layers using MAP estimation is the cornerstone of the proposed solution. The inherently available intra-layer and inter-layer header correlation proves to be very effective in selecting a reduced set of possible header configurations for further processing. The best candidate is then obtained through soft decoding of CRC protected data and CRC redundancy information itself. Simulation results for WiFi transmission using DBPSK modulated signals over AWGN channels show a substantial (4 to 12 dB) link budget improvement over classical hard decision procedures. We also introduce a sub-optimal and hardware realizable version of the proposed algorithm

Joint Source-Protocol-Channel Decoding: Improving 802.11N Receivers

International audienceThis paper combines joint protocol-channel (JPC) and joint source-channel (JSC) decoding techniques within a receiver in the context of wireless data transmission. It assumes that demodulation and channel decoding at physical (PHY) layer can provide soft information about the transmitted bits. At each layer of the protocol stack, JPC decoding allows headers of corrupted packets to be reliably decoded and soft information on the corresponding payload to be forwarded to the correct upper layer. When reaching the application (APL) layer, packets may still contain errors and are JSC decoded, exploiting residual redundancy present in the compressed bitstream, to remove part of the residual errors. The main contribution of this paper is to show that these tools may be efficiently combined to obtain i) reliable protocol layers permeable to transmission errors and ii) improved source decoders. Performance is evaluated using an OMNET++ simulation for the transmission of compressed HTML files (HTTP 1.1) over a standard RTP/UDP-Lite/Ipv6/MACLite/802:11n-PHY protocol stack, only the receiver is modified. For a given packet error rate, the proposed scheme provides gains up to 2 dB in SNR compared to a standard receiver

ROBUST DECODING OF A 3D-ESCOT BITSTREAM TRANSMITTED OVER A NOISY CHANNEL

International audienceIn this paper, we propose a joint source-channel (JSC) decoding scheme for 3D ESCOT-based video coders, such as Vidwav. The embedded bitstream generated by such coders is very sensitive to transmission errors unavoidable on wireless channels. The proposed JSC decoder employs the residual redundancy left in the bitstream by the source coder combined with bit reliability information provided by the channel or channel decoder to correct transmission errors. When considering an AWGN channel, the performance gains are in average 4 dB in terms of PSNR of the reconstructed frames, and 0.7 dB in terms of channel SNR. When considering individual frames, the obtained gain is up to 15 dB in PSNR

Adaptive network abstraction layer packetization for low bit rate H.264/AVC video transmission over wireless mobile networks under cross layer optimization

Master'sMASTER OF ENGINEERIN

Flexible Macroblock Ordering for Context-Aware Ultrasound Video Transmission over Mobile WiMAX

The most recent network technologies are enabling a variety of new applications, thanks to the provision of increased bandwidth and better management of Quality of Service. Nevertheless, telemedical services involving multimedia data are still lagging behind, due to the concern of the end users, that is, clinicians and also patients, about the low quality provided. Indeed, emerging network technologies should be appropriately exploited by designing the transmission strategy focusing on quality provision for end users. Stemming from this principle, we propose here a context-aware transmission strategy for medical video transmission over WiMAX systems. Context, in terms of regions of interest (ROI) in a specific session, is taken into account for the identification of multiple regions of interest, and compression/transmission strategies are tailored to such context information. We present a methodology based on H.264 medical video compression and Flexible Macroblock Ordering (FMO) for ROI identification. Two different unequal error protection methodologies, providing higher protection to the most diagnostically relevant data, are presented

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

Mixed streaming of video over wireless networks

In recent years, transmission of video over the Internet has become an important application. As wireless networks are becoming increasingly popular, it is expected that video will be an important application over wireless networks as well. Unlike wired networks, wireless networks have high data loss rates. Streaming video in the presence of high data loss can be a challenge because it results in errors in the video.Video applications produce large amounts of data that need to be compressed for efficient storage and transmission. Video encoders compress data into dependent frames and independent frames. During transmission, the compressed video may lose some data. Depending on where the packet loss occurs in the video, the error can propagate for a long time. If the error occurs on a reference frame at the beginning of the video, all the frames that depend on the reference frame will not be decoded successfully. This thesis presents the concept of mixed streaming, which reduces the impact of video propagation errors in error prone networks. Mixed streaming delivers a video file using two levels of reliability; reliable and unreliable. This allows sensitive parts of the video to be delivered reliably while less sensitive areas of the video are transmitted unreliably. Experiments are conducted that study the behavior of mixed streaming over error prone wireless networks. Results show that mixed streaming makes it possible to reduce the impact of errors by making sure that errors on reference frames are corrected. Correcting errors on reference frames limits the time for which errors can propagate, thereby improving the video quality. Results also show that the delay cost associated with the mixed streaming approach is reasonable for fairly high packet loss rates

Transport Layer Optimizations for Heterogeneous Wireless Multimedia Networks

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.Ph.D.Committee Chair: Vijay Madisetti; Committee Member: Raghupathy Sivakumar; Committee Member: Sudhakar Yalamanchili; Committee Member: Umakishore Ramachandran; Committee Member: Yucel Altunbasa

Měření Triple play služeb v hybridní síti

The master's thesis deals with a project regarding the implementation, design and the quality of IPTV, VoIP and Data services within the Triple Play services. In heterostructural networks made up of GEPON and xDSL technologies. Different lengths of the optical and metallic paths were used for the measurements. The first part of the thesis is theoretically analyzed the development and trend of optical and metallic networks. The second part deals with the measurement of typical optical and metallic parameters on the constructed experimental network, where its integrity was tested. Another part of the thesis is the evaluation of Triple play results, regarding the test where the network was variously tasked/burdened with data traffic and evaluated according to defined standards. The last part is concerned with the Optiwave Software simulation environment.Diplomová práce se zabývá návrhem, realizací a kvalitou služeb IPTV, VoIP a Data v rámci Triple play služeb v heterostrukturní sítí tvořené GEPON a xDSL technologiemi. Pro měření byli využity různé délky optické a metalické trasy. První části diplomové práce je teoreticky rozebrán vývoj a trend optických a metalických sítí. Druhá část se zaměřuje na měření typických optických a metalických parametrů na vybudované experimentální síti, kde byla následně testována její integrita. Dalším bodem práce je vyhodnocení výsledků Triple play, kde síť je různě zatěžována datovým provozem a následně vyhodnocována podle definovaných norem. Závěr práce je věnovaný simulačnímu prostředí Optiwave.440 - Katedra telekomunikační technikyvýborn

Multicast MAC extensions for high rate real-time traffic in wireless LANs

Nowadays we are rapidly moving from a mainly textual-based to a multimedia-based Internet, for which the widely deployed IEEE 802.11 wireless LANs can be one of the promising candidates to make them available to users anywhere, anytime, on any device. However, it is still a challenge to support group-oriented real-time multimedia services, such as video-on-demand, video conferencing, distance educations, mobile entertainment services, interactive games, etc., in wireless LANs, as the current protocols do not support multicast, in particular they just send multicast packets in open-loop as broadcast packets, i.e., without any possible acknowledgements or retransmissions. In this thesis, we focus on MAC layer reliable multicast approaches which outperform upper layer ones with both shorter delays and higher efficiencies. Different from polling based approaches, which suffer from long delays, low scalabilities and low efficiencies, we explore a feedback jamming mechanism where negative acknowledgement (NACK) frames are allowed from the non-leader receivers to destroy the acknowledgement (ACK) frame from the single leader receiver and prompts retransmissions from the sender. Based on the feedback jamming scheme, we propose two MAC layer multicast error correction protocols, SEQ driven Leader Based Protocol (SEQ-LBP) and Hybrid Leader Based Protocol (HLBP), the former is an Automatic Repeat reQuest (ARQ) scheme while the later combines both ARQ and the packet level Forward Error Correction (FEC). We evaluate the feedback jamming probabilities and the performances of SEQ-LBP and HLBP based on theoretical analyses, NS-2 simulations and experiments on a real test-bed built with consumer wireless LAN cards. Test results confirm the feasibility of the feedback jamming scheme and the outstanding performances of the proposed protocols SEQ-LBP and HLBP, in particular SEQ-LBP is good for small multicast groups due to its short delay, effectiveness and simplicity while HLBP is better for large multicast groups because of its high efficiency and high scalability with respect to the number of receivers per group.Zurzeit vollzieht sich ein schneller Wechsel vom vorwiegend textbasierten zum multimediabasierten Internet. Die weitverbreiteten IEEE 802.11 Drahtlosnetzwerke sind vielversprechende Kandidaten, um das Internet für Nutzer überall, jederzeit und auf jedem Gerät verfügbar zu machen. Die Unterstützung gruppenorientierter Echtzeit-Dienste in drahtlosen lokalen Netzen ist jedoch immer noch eine Herausforderung. Das liegt daran, dass aktuelle Protokolle keinen Multicast unterstützen. Sie senden Multicast-Pakete vielmehr in einer "Open Loop"-Strategie als Broadcast-Pakete, d. h. ohne jegliche Rückmeldung (feedback) oder Paketwiederholungen. In der vorliegenden Arbeit, anders als in den auf Teilnehmereinzelabfragen (polling) basierenden Ansätzen, die unter langen Verzögerungen, geringer Skalierbarkeit und geringer Effizienz leiden, versuchen wir, Multicast-Feedback bestehend aus positiven (ACK) und negativen Bestätigungen (NACK) auf MAC-Layer im selben Zeitfenster zu bündeln. Die übrigen Empfänger können NACK-Frames senden, um das ACK des Leaders zu zerstören und Paketwiederholungen zu veranlassen. Basierend auf einem Feedback-Jamming Schema schlagen wir zwei MAC-Layer-Protokolle für den Fehlerschutz im Multicast vor: Das SEQ-getriebene Leader Based Protocol (SEQ-LBP) und das Hybrid Leader Based Protocol (HLBP). SEQ-LBP ist eines Automatic Repeat reQuest (ARQ) Schema. HLBP kombiniert ARQ und paketbasierte Forward Error Correction (FEC). Wir evaluieren die Leistungsfähigkeit von ACK/NACK jamming, SEQ-LBP und HLBP durch Analysis, Simulationen in NS-2, sowie Experimenten in einer realen Testumgebung mit handelsüblichen WLAN-Karten. Die Testergebnisse bestätigen die Anwendbarkeit der Feedback-Jamming Schemata und die herausragende Leistungsfähigkeit der vorgestellten Protokolle SEQ-LBP und HLBP. SEQ-LBP ist durch seine kurze Verzögerung, seine Effektivität und seine Einfachheit für kleine Multicast-Gruppen nützlich, während HLBP auf Grund seiner hohen Effizienz und Skalierbarkeit im Bezug auf die Größe der Empfänger eher in großen Multicast-Gruppen anzuwenden ist