Duplicating RTP streams

Packet loss is undesirable for real-time multimedia sessions but can occur due to a variety of reasons including unplanned network outages. In unicast transmissions, recovering from such an outage can be difficult depending on the outage duration, due to the potentially large number of missing packets. In multicast transmissions, recovery is even more challenging as many receivers could be impacted by the outage. For this challenge, one solution that does not incur unbounded delay is to duplicate the packets and send them in separate redundant streams, provided that the underlying network satisfies certain requirements. This document explains how Real-time Transport Protocol (RTP) streams can be duplicated without breaking RTP or RTP Control Protocol (RTCP) rule

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

Detecting and Mitigating Denial-of-Service Attacks on Voice over IP Networks

Voice over IP (VoIP) is more susceptible to Denial of Service attacks than traditional data traffic, due to the former's low tolerance to delay and jitter. We describe the design of our VoIP Vulnerability Assessment Tool (VVAT) with which we demonstrate vulnerabilities to DoS attacks inherent in many of the popular VoIP applications available today. In our threat model we assume an adversary who is not a network administrator, nor has direct control of the channel and key VoIP elements. His aim is to degrade his victim's QoS without giving away his presence by making his attack look like a normal network degradation. Even black-boxed, applications like Skype that use proprietary protocols show poor performance under specially crafted DoS attacks to its media stream. Finally we show how securing Skype relays not only preserves many of its useful features such as seamless traversal of firewalls but also protects its users from DoS attacks such as recording of conversations and disruption of voice quality. We also present our experiences using virtualization to protect VoIP applications from 'insider attacks'. Our contribution is two fold we: 1) Outline a threat model for VoIP, incorporating our attack models in an open-source network simulator/emulator allowing VoIP vendors to check their software for vulnerabilities in a controlled environment before releasing it. 2) We present two promising approaches for protecting the confidentiality, availability and authentication of VoIP Services

Robust P2P Live Streaming

Projecte fet en col.laboració amb la Fundació i2CATThe provisioning of robust real-time communication services (voice, video, etc.) or media contents through the Internet in a distributed manner is an important challenge, which will strongly influence in current and future Internet evolution. Aware of this, we are developing a project named Trilogy leaded by the i2CAT Foundation, which has as main pillar the study, development and evaluation of Peer-to-Peer (P2P) Live streaming architectures for the distribution of high-quality media contents. In this context, this work concretely covers media coding aspects and proposes the use of Multiple Description Coding (MDC) as a flexible solution for providing robust and scalable live streaming over P2P networks. This work describes current state of the art in media coding techniques and P2P streaming architectures, presents the implemented prototype as well as its simulation and validation results

Internet-of-Things Streaming over Realtime Transport Protocol : A reusablility-oriented approach to enable IoT Streaming

The Internet of Things (IoT) as a group of technologies is gaining momentum to become a prominent factor for novel applications. The existence of high computing capability and the vast amount of IoT devices can be observed in the market today. However, transport protocols are also required to bridge these two advantages. This thesis discussed the delivery of IoT through the lens of a few selected streaming protocols, which are Realtime Transport Protocol(RTP) and its cooperatives like RTP Control Protocol(RTCP) and Session Initiation Protocol (SIP). These protocols support multimedia content transfer with a heavy-stream characteristic requirement. The main contribution of this work was the multi-layer reusability schema for IoT streaming over RTP. IoT streaming as a new concept was defined, and its characteristics were introduced to clarify its requirements. After that, the RTP stacks and their commercial implementation-VoLTE(Voice over LTE) were investigated to collect technical insights. Based on this distilled knowledge, the application areas for IoT usage and the adopting methods were described. In addition to the realization, prototypes were made to be a proof of concept for streaming IoT data with RTP functionalities on distanced devices. These prototypes proved the possibility of applying the same duo-plane architect (signaling/data transferring) widely used in RTP implementation for multimedia services. Following a standard IETF, this implementation is a minimal example of adopting an existing standard for IoT streaming applications

MBMS—IP Multicast/Broadcast in 3G Networks

In this article, the Multimedia Broadcast and Multicast Service (MBMS) as standardized in 3GPP is presented. With MBMS, multicast and broadcast capabilities are introduced into cellular networks. After an introduction into MBMS technology, MBMS radio bearer realizations are presented. Different MBMS bearer services like broadcast mode, enhanced broadcast mode and multicast mode are discussed. Streaming and download services over MBMS are presented and supported media codecs are listed. Service layer components as defined in Open Mobile Alliance (OMA) are introduced. For a Mobile TV use case capacity improvements achieved by MBMS are shown. Finally, evolution of MBMS as part of 3GPP standardization is presented

Design and Implement a Hybrid WebRTC SignallingMechanism for Unidirectional & Bi-directional VideoConferencing

WebRTC (Web Real-Time Communication) is a technology that enables browser-to-browser communication. Therefore, a signalling mechanism must be negotiated to create a connection between peers. The main aim of this paper is to create and implement a WebRTC hybrid signalling mechanism named (WebNSM) for video conferencing based on the Socket.io (API) mechanism and Firefox. WebNSM was designed over a combination of different topologies, such as simplex, star and mesh. Therefore it offers several communications at the same time as one-to-one (unidirectional/bidirectional), one-to-many (unidirectional) and many-to-many (bi-directional) without any downloading or installation. In this paper, WebRTC video conferencing was accomplished via LAN and WAN networks, including the evaluation of resources in WebRTC like bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE) and maximum links and RTPs calculation. This paper presents a novel signalling mechanism among different users, devices and networks to offer multi-party video conferencing using various topologies at the same time, as well as other typical features such as using the same server, determining room initiator, keeping the communication active even if the initiator or another peer leaves, etc. This scenario highlights the limitations of resources and the use of different topologies for WebRTC video conferencing

Content-Aware Multimedia Communications

The demands for fast, economic and reliable dissemination of multimedia information are steadily growing within our society. While people and economy increasingly rely on communication technologies, engineers still struggle with their growing complexity. Complexity in multimedia communication originates from several sources. The most prominent is the unreliability of packet networks like the Internet. Recent advances in scheduling and error control mechanisms for streaming protocols have shown that the quality and robustness of multimedia delivery can be improved significantly when protocols are aware of the content they deliver. However, the proposed mechanisms require close cooperation between transport systems and application layers which increases the overall system complexity. Current approaches also require expensive metrics and focus on special encoding formats only. A general and efficient model is missing so far. This thesis presents efficient and format-independent solutions to support cross-layer coordination in system architectures. In particular, the first contribution of this work is a generic dependency model that enables transport layers to access content-specific properties of media streams, such as dependencies between data units and their importance. The second contribution is the design of a programming model for streaming communication and its implementation as a middleware architecture. The programming model hides the complexity of protocol stacks behind simple programming abstractions, but exposes cross-layer control and monitoring options to application programmers. For example, our interfaces allow programmers to choose appropriate failure semantics at design time while they can refine error protection and visibility of low-level errors at run-time. Based on some examples we show how our middleware simplifies the integration of stream-based communication into large-scale application architectures. An important result of this work is that despite cross-layer cooperation, neither application nor transport protocol designers experience an increase in complexity. Application programmers can even reuse existing streaming protocols which effectively increases system robustness.Der Bedarf unsere Gesellschaft nach kostengünstiger und zuverlässiger Kommunikation wächst stetig. Während wir uns selbst immer mehr von modernen Kommunikationstechnologien abhängig machen, müssen die Ingenieure dieser Technologien sowohl den Bedarf nach schneller Einführung neuer Produkte befriedigen als auch die wachsende Komplexität der Systeme beherrschen. Gerade die Übertragung multimedialer Inhalte wie Video und Audiodaten ist nicht trivial. Einer der prominentesten Gründe dafür ist die Unzuverlässigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste und schwankende Laufzeiten können die Darstellungsqualität massiv beeinträchtigen. Wie jüngste Entwicklungen im Bereich der Streaming-Protokolle zeigen, sind jedoch Qualität und Robustheit der Übertragung effizient kontrollierbar, wenn Streamingprotokolle Informationen über den Inhalt der transportierten Daten ausnutzen. Existierende Ansätze, die den Inhalt von Multimediadatenströmen beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert ihren praktischen Nutzen deutlich. Außerdem erfordert der Informationsaustausch eine enge Kooperation zwischen Applikationen und Transportschichten. Da allerdings die Schnittstellen aktueller Systemarchitekturen nicht darauf vorbereitet sind, müssen entweder die Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen werden. Die Gefahr beider Varianten ist jedoch, dass sich die Komplexität eines Systems dadurch weiter erhöhen kann. Das zentrale Ziel dieser Dissertation ist es deshalb, schichtenübergreifende Koordination bei gleichzeitiger Reduzierung der Komplexität zu erreichen. Hier leistet die Arbeit zwei Beträge zum aktuellen Stand der Forschung. Erstens definiert sie ein universelles Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und Abhängigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens beschreibt die Arbeit das Noja Programmiermodell für multimediale Middleware. Noja definiert Abstraktionen zur Übertragung und Kontrolle multimedialer Ströme, die die Koordination von Streamingprotokollen mit Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete Fehlersemantiken und Kommunikationstopologien auswählen und den konkreten Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere