A New covert channel over RTP

In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP

IPtel - um sistema de IPtel com suporte para vídeo utilizando o protocolo SIP

Estudos recentes apontam que o tráfego de dados em breve excederá o tráfego telefónico, se tal já não tiver acontecido. Estes indicadores, juntamente com mais e melhores acessos à Internet, tornam cada vez maior o interesse em transportar voz e vídeo sobre redes de dados. Neste contexto nasce a Telefonia sobre IP, que oferece através desta infra-estrutura a oportunidade de criar sistemas globais de comunicação multimédia. A redução de custos e a facilidade na implementação de serviços inovadores são argumentos que justificam a forte evolução da IPtel e a tendência eventual de substituir a rede telefónica analógica. O Session Initiation Protocol (SIP), utilizado no desenvolvimento deste serviço, é um protocolo de sinalização e controlo de chamadas entre dois ou mais participantes, que tem ganho uma grande aceitação por parte de empresas ligadas ao mundo das comunicações. Desenvolvido com uma arquitectura normalizada e aberta pela Internet Engineering Task Force (IETF), espera-se que o SIP tenha o mesmo impacto no mundo das comunicações IP que o SMTP teve no e-mail e o HTTP na Web. O SIP anuncia ainda a convergência dos equipamentos e serviços de comunicações, permitindo integrar facilmente serviços de Internet como Web, e-mail, correio de voz, mensagens instantâneas, colaboração multimédia e presença (informação sobre se um utilizador está ou não disponível para comunicar). Nesta dissertação é feito um estudo sobre a evolução das diferentes partes que integram o serviço IPtel. São ainda referidas as vantagens na criação de novos serviços e obstáculos a ultrapassar por esta tecnologia de modo a poderem consolidar-se no mercado das comunicações. São apresentados diversos protocolos tipicamente usados na arquitectura protocolar da IPtel e que serão estudados, para suportar a criação do serviço sIPtel. É feita uma apresentação do serviço de Telefonia sobre IP e é explicada a arquitectura e o funcionamento do protocolo SIP, utilizado para o desenvolvimento da parte de sinalização do sIPtel. É ainda detalhado o desenvolvimento de um serviço que permite a criação, controlo e finalização de sessões de áudio e vídeo entre dois utilizadores através do protocolo SIP e por fim são realizados testes de modo a avaliar a capacidade de interoperabilidade do serviço implementado. Palavras chave: Telefonia sobre IP, Protocolos, Sinalização, Codificadores de áudio, Codificadores de vídeo, Java

VOIP weathermap - a VOIP QOS collection analysis and dissemination system

 Current trends point to VoIP as a cheaper and more effective long term solution than possible future PSTN upgrades. To move towards greater adoption of VoIP the future converged digital network is moving towards a service level management and control regime. To ensure that VoIP services provide an acceptable quality of service (QoS) a measurement solution would be helpful. The research outcome presented in this thesis is a new system for testing, analysing and presenting the call quality of Voice over Internet Protocol (VoIP). The system is called VoIP WeatherMap. Information about the current status of the Internet for VoIP calls is currently limited and a recognised approach to identifying the network status has not been adopted. An important consideration is the difficulty of assessing network conditions across links including network segments belonging to different telecommunication companies and Internet Service Providers. The VoIP WeatherMap includes the use of probes to simulate voice calls by implementing RTP/RTCP stacks. VoIP packets are sent from a probe to a server over the Internet. The important characteristics of VoIP calls such as delay and packet loss rate are collected by the server, analysed, stored in a database and presented through a web based interface. The collected voice call session data is analysed using the E-model algorithm described in ITU-T G.107. The VoIP WeatherMap presentation system includes a geographic display and internet connection links are coloured to represent the Quality of Service rank

Considering Bluetooth's Subband Codec (SBC) for Wideband Speech and Audio on the Internet

The Bluetooth Special Interest Group (SIG) has standardized the subband coding (SBC) audio codec to connect headphones via wireless Bluetooth links. SBC compresses audio at high fidelity while having an ultra-low algorithm delay. To make SBC suitable for the Internet, we extend it by using a time and packet loss concealment (PLC) algorithm that is based on ITU's G.711 Appendix I. The design is novel in the aspect of the interface between codec and speech receiver. We developed a new approach on how to distribute the functionality of a speech receiver between codec and application. Our approach leads to easier implementations of high quality VoIP applications. We conducted subjective and objective listening tests of the audio quality of SBC and PLC in order to determine an optimal coding mode and the trade-off between coding mode and packet loss rate. More precisely, we conducted MUSHRA listening tests for selected sample items. These tests results are then compared with the results of multiple objective assessment algorithms (ITU P.862 PESQ, ITU BS.1387-1 PEAQ, Creusere's algorithm). We found out that a combination of the PEAQ basic and advanced values best matches---after third order linear regression---the subjective MUSHRA results . The linear regression has coefficient of determination of R²=0.907². By comparison, our individual human ratings show a correlation of about R=0.9 compared to our averaged human rating results. Using the combination of both PEAQ algorithms, we calculate hundred thousands of objective audio quality ratings varying audio content and algorithmic parameters of SBC and PLC. The results show which set of parameters value are best suitable for a bandwidth and delay constrained link. The transmission quality of SBC is enhanced significantly by selecting optimal encoding parameters as compared to the default parameter sets given in the standard. Finally, we present preliminary objective tests results on the comparison of the audio codecs SBC, CELT, APT-X and ULD coding speech and audio transmission. They all allow a mono and stereo transmission of music at ultra-low coding delays (<10ms), which is especially useful for distributed ensemble performances over the Internet

User-Centric Quality of Service Provisioning in IP Networks

The Internet has become the preferred transport medium for almost every type of communication, continuing to grow, both in terms of the number of users and delivered services. Efforts have been made to ensure that time sensitive applications receive sufficient resources and subsequently receive an acceptable Quality of Service (QoS). However, typical Internet users no longer use a single service at a given point in time, as they are instead engaged in a multimedia-rich experience, comprising of many different concurrent services. Given the scalability problems raised by the diversity of the users and traffic, in conjunction with their increasing expectations, the task of QoS provisioning can no longer be approached from the perspective of providing priority to specific traffic types over coexisting services; either through explicit resource reservation, or traffic classification using static policies, as is the case with the current approach to QoS provisioning, Differentiated Services (Diffserv). This current use of static resource allocation and traffic shaping methods reveals a distinct lack of synergy between current QoS practices and user activities, thus highlighting a need for a QoS solution reflecting the user services. The aim of this thesis is to investigate and propose a novel QoS architecture, which considers the activities of the user and manages resources from a user-centric perspective. The research begins with a comprehensive examination of existing QoS technologies and mechanisms, arguing that current QoS practises are too static in their configuration and typically give priority to specific individual services rather than considering the user experience. The analysis also reveals the potential threat that unresponsive application traffic presents to coexisting Internet services and QoS efforts, and introduces the requirement for a balance between application QoS and fairness. This thesis proposes a novel architecture, the Congestion Aware Packet Scheduler (CAPS), which manages and controls traffic at the point of service aggregation, in order to optimise the overall QoS of the user experience. The CAPS architecture, in contrast to traditional QoS alternatives, places no predetermined precedence on a specific traffic; instead, it adapts QoS policies to each individual’s Internet traffic profile and dynamically controls the ratio of user services to maintain an optimised QoS experience. The rationale behind this approach was to enable a QoS optimised experience to each Internet user and not just those using preferred services. Furthermore, unresponsive bandwidth intensive applications, such as Peer-to-Peer, are managed fairly while minimising their impact on coexisting services. The CAPS architecture has been validated through extensive simulations with the topologies used replicating the complexity and scale of real-network ISP infrastructures. The results show that for a number of different user-traffic profiles, the proposed approach achieves an improved aggregate QoS for each user when compared with Best effort Internet, Traditional Diffserv and Weighted-RED configurations. Furthermore, the results demonstrate that the proposed architecture not only provides an optimised QoS to the user, irrespective of their traffic profile, but through the avoidance of static resource allocation, can adapt with the Internet user as their use of services change.France Teleco

Opus audiokoodekki matkapuhelinverkoissa

The latest generations in mobile networks have enabled a possibility to include high quality audio coding in data transmission. On the other hand, an on-going effort to move the audio signal processing from dedicated hardware to data centers with generalized hardware introduces a challenge of providing enough computational power needed by the virtualized network elements. This thesis evaluates the usage of a modern hybrid audio codec called Opus in a virtualized network element. It is performed by integrating the codec, testing it for functionality and performance on a general purpose processor, as well as evaluating the performance in comparison to the digital signal processor's performance. Functional testing showed that the codec was integrated successfully and bit compliance with the Opus standard was met. The performance results showed that although the digital signal processor computes the encoder's algorithms with less clock cycles, related to the processor's whole capacity the general purpose processor performs more efficiently due to higher clock frequency. For the decoder this was even clearer, when the generic hardware spends on average less clock cycles for performing the algorithms.Uusimmat sukupolvet matkapuhelinverkoissa mahdollistavat korkealaatuisen audiokoodauksen tiedonsiirrossa. Toisaalta audiosignaalinkäsittelyn siirtäminen sovelluskohtaisesta laitteistosta keskitettyjen palvelinkeskusten yleiskäyttöiseen laitteistoon on käynnissä, mikä aiheuttaa haasteita tarjota riittävästi laskennallista tehoa virtualisoituja verkkoelementtejä varten. Tämä diplomityö arvioi modernin hybridikoodekin, Opuksen, käyttöä virtualisoidussa verkkoelementissä. Se on toteutettu integroimalla koodekki, testaamalla funktionaalisuutta ja suorituskykyä yleiskäyttöisellä prosessorilla sekä arvioimalla suorituskykyä verrattuna digitaalisen signaaliprosessorin suorituskykyyn. Funktionaalinen testaus osoitti että koodekki oli integroitu onnistuneesti ja että bittitason yhdenmukaisuus Opuksen standardin kanssa saavutettiin. Suorituskyvyn testitulokset osoittivat, että vaikka enkoodaus tuotti vähemmän kellojaksoja digitaalisella signaaliprosessorilla, yleiskäyttöinen prosessori suoriutuu tehokkaammin suhteutettuna prosessorin kokonaiskapasiteettiin korkeamman kellotaajuuden ansiosta. Dekooderilla tämä näkyi vielä selkeämmin, sillä yleiskäyttöinen prosessori kulutti keskimäärin vähemmän kellojaksoja algoritmien suorittamiseen

ADAPTIVE UNMANNED VEHICLE AUTOPILOTING USING WEBRTC VIDEO ANALYSIS

Εκμεταλλευόμαστε τις νέες δυνατότητες που παρέχονται από το WebRTC, υπό την έννοια της διαλειτουργικότητας και των τελευταίας γενιάς επικοινωνιών σε πραγματικό χρόνο, προκειμένου να αναπτύξουμε ένα σύστημα για το πιλοτάρισμα μη επανδρωμένων οχημάτων χρησιμοποιώντας ανάλυση βίντεο. Συγκεκριμένα, ορίζουμε μια τοπολογία όπου ένα ROS όχημα μεταδίδει βίντεο μέσω WebRTC προς έναν ενδιάμεσο εξυπηρετητή, ο οποίος με τη σειρά του το μεταβιβάζει σε έναν πελάτη. Ο εξυπηρετητής εκμεταλλεύεται τη βιβλιοθήκη OpenCV και εφαρμόζει ανάλυση βίντεο, με τέτοιο τρόπο ώστε να εξυπηρετήσει ένα επιλεγμένο από τον πελάτη σενάριο. Οι αντίστοιχες εντολές μεταδίδονται στο όχημα, με αποτέλεσμα να έχουμε ένα αυτόματα οδηγούμενο όχημα. Ο πελάτης παρακολουθεί την πορεία του οχήματος και μπορεί να αλλάξει δυναμικά το επιλεγμένο σενάριο – αυτό σημαίνει είτε να αλλάξει ελαφρώς τη λειτουργία του (π.χ. από παρακολούθηση ανθρώπων σε παρακολούθηση παιδιών) είτε να ενεργοποιήσει μια εντελώς διαφορετική φιλοσοφία λειτουργίας – στέλνοντας τα κατάλληλα αιτήματα στον εξυπηρετητή. Μόλις ο εξυπηρετητής λάβει αυτά τα αιτήματα, χρησιμοποιεί τις αντίστοιχες λειτουργίες το OpenCV για να εξυπηρετήσει το νέο σενάριο, και στέλνει τις νέες εντολές οδήγησης στο όχημα, αναγκάζοντας το σύστημα να υιοθετήσει μια νέα λειτουργία αυτόματου πιλότου. Η επικοινωνία μεταξύ του οχήματος, του εξυπηρετητή και του πελάτη εδραιώνεται μέσω των SIP/SDP και ενορχηστρώνεται μέσω ενός Web-Socket εξυπηρετητή που επιτελεί το ρόλο του Signaling Server, ενώ οι εντολές μεταφέρονται μέσω του WebRTC Data Channel πάνω από το SCTP. Περιγράφουμε και αναλύουμε το πώς όλα αυτά τα ετερογενή συστατικά (WebRTC – OpenCV – ROS) συνδυάζονται για τη δημιουργία μιας δικτυακής υποδομής, για το αυτόματο πιλοτάρισμα ROS οχημάτων σύμφωνα με ένα συγκεκριμένο σενάριο χρήσης. Τέλος, τα αποτελέσματα αποδεικνύουν την ιδέα μας, δηλαδή μια οριζόντια υποδομή που (α) αποτελείται από μια ευέλικτη/αρθρωτή αρχιτεκτονική, (β) παρέχει τα απαραίτητα στοιχεία για την μηχανή-σε-μηχανή επικοινωνία, (γ) χρησιμοποιεί τελευταίας γενιάς τεχνολογίες, (δ) επιτρέπει σε έναν προγραμματιστή να εφαρμόσει τη δική του λογική κατακόρυφα σε βάθος και (ε) παρέχει στον τομέα του IoT μια λύση που μπορεί εύκολα να αξιοποιηθεί με πολλούς τρόπους.We exploit the new features provided by WebRTC in terms of interoperability and state-of-the-art real-time communications, in order to develop a system for piloting unmanned vehicles using video analysis. Specifically, we define a topology where a ROS-based vehicle transmits its video using WebRTC to an intermediate server, who in turn relays it to a client. The server takes advantage of the OpenCV library and applies video analysis, with respect to a selected task (i.e. face detection) defined by the client. The corresponding commands are transmitted to the vehicle, resulting in an automatically driven unmanned vehicle. The client monitors the vehicle’s movement and can dynamically change the selected use case; that is, either change slightly its operation (i.e. from human tracking to children tracking) or enable an entirely new core philosophy (i.e. to fire detection) by sending the appropriate requests to the server. Upon reception of these requests, the server utilizes the corresponding OpenCV functionalities to serve the new task, and sends the new piloting commands to the vehicle, forcing the system to adopt a new autopiloting mode. This communication between the vehicle, the server and the client is established using SIP/SDP and orchestrated via a WebSocket server that serves as a Signaling Server, the media are transferred through SRTP/UDP, and the commands are carried via the WebRTC Data Channel over SCTP. We explain and describe how to combine all of these heterogeneous components (WebRTC – OpenCV – ROS), in order to compose a web-based infrastructure for autopiloting ROS-based vehicles upon a specific use case. Finally, the results prove our concept, meaning a horizontal infrastructure that (a) consists of a modular architecture, (b) provides the necessary components for machine-to-machine communication, (c) uses state-of-the-art technologies, (d) allows a developer to implement her own logic vertically, and (e) provides IoT with a solution that can be easily exploited in numerous ways

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

Análisis de la calidad experimentada en aplicaciones de voz sobre IP de libre distribución

En este proyecto el trabajo que se ha realizado ha consistido en determinar los parámetros de la red que determinan la calidad de servicio de las aplicaciones de VoIP y comparar los resultados obtenidos con las encuestas realizadas a un número de personas. Para ello, primeramente se ha realizado un estudio de las características y funcionamiento de los protocolos para VoIP. Seguidamente se eligieron 5 aplicaciones de VoIP y se realizaron capturas de su transferencia de paquetes. Con estos datos se ha podido calcular los parámetros de la red y se ha establecido de este modo la calidad de servicio ofrecida. El análisis de las aplicaciones se realizará mediante capturas con la herramienta wireshark, que serán analizadas posteriormente utilizando awk. Por otro lado para medir la QoE se han realizado encuestas de las mismas aplicaciones a un número elevado de personas. Con estos datos, hemos podido establecer una comparación de los resultados obtenidos en ambos estudios para realizar una posterior conclusión de los mismos.Escuela Técnica Superior de Ingeniería de Telecomunicació