Robust Header Compression (ROHC) in Next-Generation Network Processors

Robust Header Compression (ROHC) provides for more efficient use of radio links for wireless communication in a packet switched network. Due to its potential advantages in the wireless access area andthe proliferation of network processors in access infrastructure, there exists a need to understand the resource requirements and architectural implications of implementing ROHC in this environment. We presentan analysis of the primary functional blocks of ROHC and extract the architectural implications on next-generation network processor design for wireless access. The discussion focuses on memory space andbandwidth dimensioning as well as processing resource budgets. We conclude with an examination of resource consumption and potential performance gains achievable by offloading computationally intensiveROHC functions to application specific hardware assists. We explore the design tradeoffs for hardware as-sists in the form of reconfigurable hardware, Application-Specific Instruction-set Processors (ASIPs), andApplication-Specific Integrated Circuits (ASICs)

Unidirectional and bidirectional optimistic modes IP header compression for real-time video streaming

Communication over Internet Protocol (IP) networks, has become crucial component of day everyday activities. They are utilized over the Internet to support a wide range of services. The flexibility of this kind of transmission relies on the IP User Datagram Protocol (UDP), IP/UDP/Real-time Transport Protocol (RTP) and IP/Transmission Control Protocol (TCP). Unfortunately, the weight of encapsulated protocol headers affects the transmission efficiency. This research aims at improving a technique that reduce the packets header size by compression. Performance analysis of the enhanced efficient techniques in both unidirectional and bidirectional optimistic modes applied to real-time video streaming traffic for UDP/IP and HTTP/TCP flows over free error channel has been conducted. The finding shows that the header compression ratio in each case is good and better than the previous studies. The technique achieved a reduction up to 90% for RTP/UDP/IP, 89% for UDP /IP and 77.5 to 86.5 % for TCP/IP profile. This research contribution is restricted to compression gain and saving for 0x0000, 0x0001, 0x0002 and 0x0006 profiles in the unidirectional and bidirectional optimistic mode

Hardware Acceleration of the Robust Header Compression (RoHC) Algorithm

With the proliferation of Long Term Evolution (LTE) networks, many cellular carriers are embracing the emerging eld of mobile Voice over Internet Protocol (VoIP). The robust header compression (RoHC) framework was introduced as a part of the LTE Layer 2 stack to compress the large headers of the VoIP packets before transmitted over LTE IP-based architectures. The headers, which are encapsulated Real-time Transport Protocol (RTP)/User Datagram Protocol (UDP)/Internet Protocol (IP) stack, are large compared to the small payload. This header-compression scheme is especially useful for ecient utilization of the radio bandwidth and network resources. In an LTE base-station implementation, RoHC is a processing-intensive algorithm that may be the bottleneck of the system, and thus, may be the limiting factor when it comes to number of users served. In this thesis, a hardware-software and a full-hardware solution are proposed, targeting LTE base-stations to accelerate this computationally intensive algorithm and enhance the throughput and the capacity of the system. The results of both solutions are discussed and compared with respect to design metrics like throughput, capacity, power consumption, chip area and exibility. This comparison is instrumental in taking architectural level trade-o decisions in-order to meet the present day requirements and also be ready to support future evolution. In terms of throughput, a gain of 20% (6250 packets/sec can be processed at a frequency of 150 MHz) is achieved in the HW-SW solution compared to the SW-Only solution by implementing the Cyclic Redundancy Check (CRC) and the Least Signicant Bit(LSB) encoding blocks as hardware accelerators . Whereas, a Full-HW implementation leads to a throughput of 45 times (244000 packets/sec can be processed at a frequency of 100 MHz) the throughput of the SW-Only solution. However, the full-HW solution consumes more Lookup Tables (LUTs) when it is synthesized on an Field-Programmable Gate Array (FPGA) platform compared to the HW-SW solution. In Arria II GX, the HW-SW and the full-HW solutions use 2578 and 7477 LUTs and consume 1.5 and 0.9 Watts, respectively. Finally, both solutions are synthesized and veried on Altera's Arria II GX FPGA

Header Compression and Signal Processing for Wideband Communication Systems.

This thesis is dedicated to the investigation, development and practical verification of header compression and signal processing techniques over TErrestrial Trunked RAdio (TETRA), TETRA Enhanced Data Services (TEDS) and Power Line Communication (PLC). TETRA release I is a narrowband private mobile radio technology used by safety and security organizations, while TEDS is a widebandsystem. With the introduction of IP support, TEDS enables multimedia based applications and services to communicate across communication systems. However the IP extension for TEDS comes at a cost of significant header contributions with the payload. With small application payloads and fast rate application traffic profiles, the header contribution in the total size of the packet is considerably more than the actual application payload. This overhead constitutes the considerable slot capacity at the physical layer of TEDS and PLC. Advanced header compression techniques such as Robust Header Compression (RoHC) compress the huge header sizes and offer significant compression gain without compromising quality of service (QoS). Systems can utilize this bandwidth to transmit more information payload than control information. In this study, the objective is to investigate the integration of RoHC in TEDS and design a novel IPv6 enabled protocol stack for PLC with integrated RoHC. The purpose of the study is also to investigate the throughput optimization technique such as RoHC over TEDS and PLC by simulating different traffic profile classes and to illustrate the benefit of using RoHC over TEDS and PLC. The thesis also aims to design and simulate the TEDS physical layer for the purpose of investigating the performance of higher order modulation schemes. Current TEDS, standards are based on the transmission frequencies above 400MHz range, however with delays in the standardization of broadband TETRA, it is important to explore all possible avenues to extend the capacity of the system. The research concludes the finding of the application of RoHC for TEDS and PLC, against different traffic classes and propagation channels. The benefit of using RoHC in terms of saving bandwidth, slot capacity and other QoS parameters is presented along with integration aspects into TEDS and PLC communication stacks. The study also presents the TEDS physical layer simulation results for modulation schemes and transmission frequency other than specified in the standard. The research results presented in this thesis have been published in international symposiums and professional journals. The application of the benefits of using RoHC for TEDS has been proposed to the ETSI TETRA for contribution to the TETRA standard under STF 378. Simulation results for the investigation of characteristics of ?/4 DQPSK performance below 200 MHz have also been also presented to ETSI TETRA as a contribution to the existing TEDS standard. The Results presented for the design of IPv6 enabled stacked with integrated RoHC have been submitted as deliverable under the FP-7 project DLC+VIT4IP. All the results, simulations and investigations presented in the thesis have been carried out through the platform provided by HW Communication Ltd

SIP COMPRESSION

The wired line network has been well studied and widely used for a long time. Most of its protocols are so successful that passed the test of time. There are many similar tasks in mobile and wired line environment, and we would like to achieve compatible, inter-working solutions. So it is a plausible idea to use the protocols of the wired line network in mobile environment too. However, the mobile and wired line environment differ significantly; mainly the bandwidth is different in the two networks. Although the difference is going to be smaller with the help of new generation of mobile networks, it will still remain significant. An acceptable solution is to compress these protocols. We have not found such a solution in the literature so our opinion is that this article is the first dealing with SIP compression. We have created a demonstration system, which connects two SIP user agents to each other and ensures the compression and decompression of the messages between them. In this article we show our development about adapting various compressing algorithms for SIP compression, and we evaluate them

Improving the robustness of CELP-like speech decoders using late-arrival packets information : application to G.729 standard in VoIP

L'utilisation de la voix sur Internet est une nouvelle tendance dans Ie secteur des télécommunications et de la réseautique. La paquetisation des données et de la voix est réalisée en utilisant Ie protocole Internet (IP). Plusieurs codecs existent pour convertir la voix codée en paquets. La voix codée est paquetisée et transmise sur Internet. À la réception, certains paquets sont soit perdus, endommages ou arrivent en retard. Ceci est cause par des contraintes telles que Ie délai («jitter»), la congestion et les erreurs de réseau. Ces contraintes dégradent la qualité de la voix. Puisque la transmission de la voix est en temps réel, Ie récepteur ne peut pas demander la retransmission de paquets perdus ou endommages car ceci va causer plus de délai. Au lieu de cela, des méthodes de récupération des paquets perdus (« concealment ») s'appliquent soit à l'émetteur soit au récepteur pour remplacer les paquets perdus ou endommages. Ce projet vise à implémenter une méthode innovatrice pour améliorer Ie temps de convergence suite a la perte de paquets au récepteur d'une application de Voix sur IP. La méthode a déjà été intégrée dans un codeur large-bande (AMR-WB) et a significativement amélioré la qualité de la voix en présence de <<jitter » dans Ie temps d'arrivée des trames au décodeur. Dans ce projet, la même méthode sera intégrée dans un codeur a bande étroite (ITU-T G.729) qui est largement utilise dans les applications de voix sur IP. Le codeur ITU-T G.729 défini des standards pour coder et décoder la voix a 8 kb/s en utilisant 1'algorithme CS-CELP (Conjugate Stmcture Algebraic Code-Excited Linear Prediction).Abstract: Voice over Internet applications is the new trend in telecommunications and networking industry today. Packetizing data/voice is done using the Internet protocol (IP). Various codecs exist to convert the raw voice data into packets. The coded and packetized speech is transmitted over the Internet. At the receiving end some packets are either lost, damaged or arrive late. This is due to constraints such as network delay (fitter), network congestion and network errors. These constraints degrade the quality of speech. Since voice transmission is in real-time, the receiver can not request the retransmission of lost or damaged packets as this will cause more delay. Instead, concealment methods are applied either at the transmitter side (coder-based) or at the receiver side (decoder-based) to replace these lost or late-arrival packets. This work attempts to implement a novel method for improving the recovery time of concealed speech The method has already been integrated in a wideband speech coder (AMR-WB) and significantly improved the quality of speech in the presence of jitter in the arrival time of speech frames at the decoder. In this work, the same method will be integrated in a narrowband speech coder (ITU-T G.729) that is widely used in VoIP applications. The ITUT G.729 coder defines the standards for coding and decoding speech at 8 kb/s using Conjugate Structure Algebraic Code-Excited Linear Prediction (CS-CELP) Algorithm

ClassBench: A Packet Classification Benchmark

Due to the importance and complexity of the packet classification problem, a myriad of algorithms and re-sulting implementations exist. The performance and capacity of many algorithms and classification devices, including TCAMs, depend upon properties of the filter set and query patterns. Unlike microprocessors in the field of computer architecture, there are no standard performance evaluation tools or techniques avail-able to evaluate packet classification algorithms and products. Network service providers are reluctant to distribute copies of real filter sets for security and confidentiality reasons, hence realistic test vectors are a scarce commodity. The small subset of the research community who obtain real filter sets either limit performance evaluation to the small sample space or employ ad hoc methods of modifying those filter sets. In response to this problem, we present ClassBench, a suite of tools for benchmarking packet classification algorithms and devices. ClassBench includes a Filter Set Generator that produces synthetic filter sets that accurately model the characteristics of real filter sets. Along with varying the size of the filter sets, we provide high-level control over the composition of the filters in the resulting filter set. The tools suite also includes a Trace Generator that produces a sequence of packet headers to exercise the synthetic filter set. Along with specifying the relative size of the trace, we provide a simple mechanism for controlling locality of reference in the trace. While we have already found ClassBench to be very useful in our own research, we seek to initiate a broader discussion and solicit input from the community to guide the refinement of the tools and codification of a formal benchmarking methodology

Platform for quality of experience evaluation in real time applications over LTE networks

Dissertação apresentado à Escola Superior de Tecnologia do Instituto Politécnico de Castelo Branco para cumprimento dos requisitos necessários à obtenção do grau de Mestre em Desenvolvimento de Software e Sistemas InterativosAtualmente existem vário simuladores para várias tecnologias de redes sem fios (LTE, UMTS, Wi-Fi ...). Quase todos eles simulam valores para diferentes utilizadores como por exemplo de taxas de transferência (Mbit/s), a potência recebida, a SNR, entre outros valores, dependendo do tipo de simulação. A maioria dos resultados apresentados pelos simuladores correspondem apenas a números, como valores de taxa de transferência ou BER. Então, é difícil entender o impacto desses valores numa comunicação real. Pretende-se com este projeto dar a utilizador por exemplo 2 Mbits/s de taxa de transferência (uplink/downlink), um valor BER de 1x10−6 ou uma potência recebida em torno 1NW obtidos num cenário de simulação e em tempo real e para um cenário real o utilizador experienciar as condições de comunicação e interatividade com as mesmas aplicações utilizadas na realidade. O desenvolvimento da plataforma proposta neste projeto tem como objetivo verificar e avaliar em tempo real a QoS e a QoE obtida para um utilizador simulado naquele momento e para o cenário simulado. Isso permite que os utilizadores experienciem a interatividade com aplicações para diferentes cenários de simulação. Esta plataforma tem como objetivo converter os valores numéricos obtidos apenas por ferramentas de simulação, para uma experiência em tempo real para um determinado cenário simulado. Inicialmente, a rede pretendida a simular é LTE, mas outros protocolos e tipos de rede poderão ser utilizados e testados nesta plataforma, desde que sejam baseados no protocolo IP, tal como o LTE.Abstract: There are many simulators for various wireless technologies (LTE, UMTS, WI-FI …). Almost all of them have different values for users as bitrate, received power, SNR, among other values depending on the simulation type. Most of the simulators results are just numbers like bitrate or BER values. So is difficult to understand the impact of those values in a real communication. It is intended with this project to give a user for instance 2Mbits/s bitrate, a BER value of 1x10−6 or a received power around 1NW in a simulation scenario and he could in real time and real scenario experience the communication conditions and interactivity with applications. The development of the platform proposed in this project aims to verify in real time the QoS and QoE which simulated user experiences in that moment on the simulated scenario. This allows users to experience the interactivity with applications for different simulation scenarios. This platform aims to convert the values merely numerical, obtained by simulation tools, to a real-time experience for the scenario simulated. Initially the target network is LTE, but other network protocols will be allowed to use ant test, since that they are IP based protocols like LTE