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

Applications of satellite technology to broadband ISDN networks

Two satellite architectures for delivering broadband integrated services digital network (B-ISDN) service are evaluated. The first is assumed integral to an existing terrestrial network, and provides complementary services such as interconnects to remote nodes as well as high-rate multicast and broadcast service. The interconnects are at a 155 Mbs rate and are shown as being met with a nonregenerative multibeam satellite having 10-1.5 degree spots. The second satellite architecture focuses on providing private B-ISDN networks as well as acting as a gateway to the public network. This is conceived as being provided by a regenerative multibeam satellite with on-board ATM (asynchronous transfer mode) processing payload. With up to 800 Mbs offered, higher satellite EIRP is required. This is accomplished with 12-0.4 degree hopping beams, covering a total of 110 dwell positions. It is estimated the space segment capital cost for architecture one would be about $190M whereas the second architecture would be about$250M. The net user cost is given for a variety of scenarios, but the cost for 155 Mbs services is shown to be about $15-22/minute for 25 percent system utilization

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

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

D13.1 Fundamental issues on energy- and bandwidth-efficient communications and networking

Deliverable D13.1 del projecte europeu NEWCOM#The report presents the current status in the research area of energy- and bandwidth-efficient communications and networking and highlights the fundamental issues still open for further investigation. Furthermore, the report presents the Joint Research Activities (JRAs) which will be performed within WP1.3. For each activity there is the description, the identification of the adherence with the identified fundamental open issues, a presentation of the initial results, and a roadmap for the planned joint research work in each topic.Preprin

Decodificació de canal assistida per protocol

The purpose of this project is to demonstrate how the channel decoder at physical layer can benefit from the knowledge of the protocol used. Making this possible by exploiting the redundancy present in the protocol using known bits as pilot bits during the decoding process.This document describes how the channel decoder at Physical layer of a communication system can benefit from information brought either by careful examination of the standard, or by already received packets. In all communication systems we have a stablished protocol. That helps communication between sender and receiver more understanding. This fixed structure usually leads to the systematic use of headers and other repetitive fields in the frame structure. What if we can take profit of all this redundancy of the standard to improve the decoding process? That's the main idea of Protocol-Assisted Channel Decoding.La presente tesis describe como el decodificador de canal en la capa física de un sistema de comunicación se puede beneficiar de información obtenida por un detenido estudio del estándar, o por paquetes recibidos anteriormente. En todo sistema de comunicación existe un protocolo establecido. Esto ayuda a que haya un entendimiento entre el emisor y el receptor durante la comunicación. Esta estructura fija normalmente conlleva el uso sistemático de cabeceras y campos repetitivos en la estructura de los paquetes. ¿Y si podemos aprovechar de esa redundancia del estándar para mejorar el proceso de decodificación? Esa es la principal idea de la Decodificación de Canal Asistida por Protocolo.La present tèsi descriu com el decodificador de canal a la capa física d’un sistema de comunicació es pot beneficiar de l’informació obtinguda per un rigurós estudi de l’estàndard, o de paquets rebuts anteriorment. A tot sistema de comunicació hi existeix un protocol establert. Això ajuda a que hi hagi una entesa entre l’emissor i el receptor durant la comunicació. Aquesta estructura fixa normalment implica l’ús sistemàtic de capçaleres i camps repetitius a l’estructura dels paquets. I si podem aprofitar aquesta redundància de l’estàndard per tal de millorar el procés de decodificació? Aquesta és la principal idea de la Decodificació de Canal Assistida per Protocol

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

Security Enhancements in Voice Over Ip Networks

Voice delivery over IP networks including VoIP (Voice over IP) and VoLTE (Voice over LTE) are emerging as the alternatives to the conventional public telephony networks. With the growing number of subscribers and the global integration of 4/5G by operations, VoIP/VoLTE as the only option for voice delivery becomes an attractive target to be abused and exploited by malicious attackers. This dissertation aims to address some of the security challenges in VoIP/VoLTE. When we examine the past events to identify trends and changes in attacking strategies, we find that spam calls, caller-ID spoofing, and DoS attacks are the most imminent threats to VoIP deployments. Compared to email spam, voice spam will be much more obnoxious and time consuming nuisance for human subscribers to filter out. Since the threat of voice spam could become as serious as email spam, we first focus on spam detection and propose a content-based approach to protect telephone subscribers\u27 voice mailboxes from voice spam. Caller-ID has long been used to enable the callee parties know who is calling, verify his identity for authentication and his physical location for emergency services. VoIP and other packet switched networks such as all-IP Long Term Evolution (LTE) network provide flexibility that helps subscribers to use arbitrary caller-ID. Moreover, interconnecting between IP telephony and other Circuit-Switched (CS) legacy telephone networks has also weakened the security of caller-ID systems. We observe that the determination of true identity of a calling device helps us in preventing many VoIP attacks, such as caller-ID spoofing, spamming and call flooding attacks. This motivates us to take a very different approach to the VoIP problems and attempt to answer a fundamental question: is it possible to know the type of a device a subscriber uses to originate a call? By exploiting the impreciseness of the codec sampling rate in the caller\u27s RTP streams, we propose a fuzzy rule-based system to remotely identify calling devices. Finally, we propose a caller-ID based public key infrastructure for VoIP and VoLTE that provides signature generation at the calling party side as well as signature verification at the callee party side. The proposed signature can be used as caller-ID trust to prevent caller-ID spoofing and unsolicited calls. Our approach is based on the identity-based cryptography, and it also leverages the Domain Name System (DNS) and proxy servers in the VoIP architecture, as well as the Home Subscriber Server (HSS) and Call Session Control Function (CSCF) in the IP Multimedia Subsystem (IMS) architecture. Using OPNET, we then develop a comprehensive simulation testbed for the evaluation of our proposed infrastructure. Our simulation results show that the average call setup delays induced by our infrastructure are hardly noticeable by telephony subscribers and the extra signaling overhead is negligible. Therefore, our proposed infrastructure can be adopted to widely verify caller-ID in telephony networks